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Versions: (RFC 2326) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 34 35 36 37 38 39 40 RFC 7826

Internet Engineering Task Force                                MMUSIC WG
Internet Draft                                            H. Schulzrinne
draft-ietf-mmusic-rfc2326bis-08.txt                          Columbia U.
October 25, 2004                                                  A. Rao
Expires: April, 2005                                               Cisco
                                                             R. Lanphier
                                                            RealNetworks
                                                           M. Westerlund
                                                                Ericsson
                                                           A. Narasimhan
                                                               Princeton



                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes?
   aware will be disclosed, in accordance with( Section 6 of) RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

Abstract

   This memorandum is a revision of RFC 2326, which is currently a
   Proposed Standard.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time
   properties. RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video. Sources of data can include both live data feeds and stored
   clips. This protocol is intended to control multiple data delivery



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   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).
















































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                           Table of Contents



   1          Introduction ........................................    9
   1.1        RTSP Specification Update ...........................    9
   1.2        Purpose .............................................   10
   1.3        Notational Conventions ..............................   11
   1.4        Terminology .........................................   12
   1.5        Protocol Properties .................................   15
   1.6        Extending RTSP ......................................   17
   1.7        Overall Operation ...................................   18
   1.8        RTSP States .........................................   19
   1.9        Relationship with Other Protocols ...................   19
   2          RTSP Use Cases ......................................   20
   2.1        On-demand Playback of Stored Content ................   20
   2.2        Unicast distribution of Live Content ................   22
   2.3        On-demand Playback using Multicast ..................   22
   2.4        Inviting a RTSP server into a conference ............   22
   2.5        Live Content using Multicast ........................   23
   3          Protocol Parameters .................................   24
   3.1        RTSP Version ........................................   24
   3.2        RTSP URI ............................................   24
   3.3        Session Identifiers .................................   26
   3.4        SMPTE Relative Timestamps ...........................   26
   3.5        Normal Play Time ....................................   27
   3.6        Absolute Time .......................................   27
   3.7        Feature-tags ........................................   28
   3.8        Entity Tags .........................................   28
   4          RTSP Message ........................................   28
   4.1        Message Types .......................................   29
   4.2        Message Headers .....................................   29
   4.3        Message Body ........................................   29
   4.4        Message Length ......................................   29
   5          General Header Fields ...............................   30
   6          Request .............................................   30
   6.1        Request Line ........................................   30
   6.2        Request Header Fields ...............................   32
   7          Response ............................................   32
   7.1        Status-Line .........................................   33
   7.1.1      Status Code and Reason Phrase .......................   33
   7.1.2      Response Header Fields ..............................   34
   8          Entity ..............................................   34
   8.1        Entity Header Fields ................................   35
   8.2        Entity Body .........................................   35



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   9          Connections .........................................   35
   9.1        Pipelining ..........................................   37
   9.2        Reliability and Acknowledgements ....................   37
   9.3        The usage of connections ............................   38
   9.4        Timing Out RTSP messages ............................   39
   9.5        Use of IPv6 .........................................   40
   10         Capability Handling .................................   40
   11         Method Definitions ..................................   42
   11.1       OPTIONS .............................................   43
   11.2       DESCRIBE ............................................   44
   11.3       SETUP ...............................................   46
   11.4       PLAY ................................................   49
   11.5       PAUSE ...............................................   53
   11.6       TEARDOWN ............................................   57
   11.7       GET_PARAMETER .......................................   57
   11.8       SET_PARAMETER .......................................   58
   11.9       REDIRECT ............................................   60
   11.10      PING ................................................   62
   12         Embedded (Interleaved) Binary Data ..................   63
   13         Status Code Definitions .............................   64
   13.1       Success 1xx .........................................   64
   13.1.1     100 Continue ........................................   64
   13.2       Success 2xx .........................................   64
   13.3       Redirection 3xx .....................................   65
   13.3.1     300 Multiple Choices ................................   65
   13.3.2     301 Moved Permanently ...............................   65
   13.3.3     302 Found ...........................................   65
   13.3.4     303 See Other .......................................   66
   13.3.5     304 Not Modified ....................................   66
   13.3.6     305 Use Proxy .......................................   66
   13.4       Client Error 4xx ....................................   66
   13.4.1     400 Bad Request .....................................   66
   13.4.2     405 Method Not Allowed ..............................   66
   13.4.3     451 Parameter Not Understood ........................   67
   13.4.4     452 reserved ........................................   67
   13.4.5     453 Not Enough Bandwidth ............................   67
   13.4.6     454 Session Not Found ...............................   67
   13.4.7     455 Method Not Valid in This State ..................   67
   13.4.8     456 Header Field Not Valid for Resource .............   67
   13.4.9     457 Invalid Range ...................................   67
   13.4.10    458 Parameter Is Read-Only ..........................   68
   13.4.11    459 Aggregate Operation Not Allowed .................   68
   13.4.12    460 Only Aggregate Operation Allowed ................   68
   13.4.13    461 Unsupported Transport ...........................   68
   13.4.14    462 Destination Unreachable .........................   68
   13.4.15    470 Connection Authorization Required ...............   68
   13.4.16    471 Connection Credentials not accepted .............   68
   13.5       Server Error 5xx ....................................   68



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   13.5.1     551 Option not supported ............................   68
   14         Header Field Definitions ............................   69
   14.1       Accept ..............................................   71
   14.2       Accept-Credentials ..................................   71
   14.3       Accept-Encoding .....................................   75
   14.4       Accept-Language .....................................   75
   14.5       Accept-Ranges .......................................   75
   14.6       Allow ...............................................   76
   14.7       Authorization .......................................   76
   14.8       Bandwidth ...........................................   76
   14.9       Blocksize ...........................................   77
   14.10      Cache-Control .......................................   77
   14.11      Connection ..........................................   79
   14.12      Connection-Credentials ..............................   79
   14.13      Content-Base ........................................   80
   14.14      Content-Encoding ....................................   80
   14.15      Content-Language ....................................   80
   14.16      Content-Length ......................................   80
   14.17      Content-Location ....................................   80
   14.18      Content-Type ........................................   81
   14.19      CSeq ................................................   81
   14.20      Date ................................................   81
   14.21      ETag ................................................   81
   14.22      Expires .............................................   82
   14.23      From ................................................   83
   14.24      Host ................................................   83
   14.25      If-Match ............................................   83
   14.26      If-Modified-Since ...................................   83
   14.27      If-None-Match .......................................   83
   14.28      Last-Modified .......................................   84
   14.29      Location ............................................   84
   14.30      Proxy-Authenticate ..................................   84
   14.31      Proxy-Require .......................................   84
   14.32      Proxy-Supported .....................................   84
   14.33      Public ..............................................   85
   14.34      Range ...............................................   86
   14.35      Referer .............................................   88
   14.36      Retry-After .........................................   88
   14.37      Require .............................................   88
   14.38      RTP-Info ............................................   89
   14.39      Scale ...............................................   91
   14.40      Speed ...............................................   92
   14.41      Server ..............................................   92
   14.42      Session .............................................   92
   14.43      Supported ...........................................   95
   14.44      Timestamp ...........................................   95
   14.45      Transport ...........................................   95
   14.46      Unsupported .........................................  102



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   14.47      User-Agent ..........................................  102
   14.48      Vary ................................................  102
   14.49      Via .................................................  102
   14.50      WWW-Authenticate ....................................  102
   15         Caching .............................................  102
   16         Examples ............................................  103
   16.1       Media on Demand (Unicast) ...........................  103
   16.2       Streaming of a Container file .......................  106
   16.3       Single Stream Container Files .......................  109
   16.4       Live Media Presentation Using Multicast .............  111
   16.5       Capability Negotiation ..............................  112
   17         Security Framework ..................................  113
   17.1       RTSP and HTTP Authentication ........................  114
   17.2       RTSP over TLS .......................................  114
   17.3       Security and Proxies ................................  115
   17.3.1     Accept-Credentials ..................................  116
   17.3.2     User approved TLS procedure .........................  117
   18         Syntax ..............................................  118
   18.1       Base Syntax .........................................  118
   18.2       RTSP Protocol Definition ............................  119
   18.2.1     Generic Protocol elements ...........................  119
   18.2.2     Message Syntax ......................................  120
   18.2.3     Header Syntax .......................................  124
   19         Security Considerations .............................  127
   20         IANA Considerations .................................  129
   20.1       Feature-tags ........................................  130
   20.1.1     Description .........................................  130
   20.1.2     Registering New Feature-tags with IANA ..............  130
   20.1.3     Registered entries ..................................  130
   20.2       RTSP Methods ........................................  130
   20.2.1     Description .........................................  130
   20.2.2     Registering New Methods with IANA ...................  131
   20.2.3     Registered Entries ..................................  131
   20.3       RTSP Status Codes ...................................  131
   20.3.1     Description .........................................  131
   20.3.2     Registering New Status Codes with IANA ..............  131
   20.3.3     Registered Entries ..................................  132
   20.4       RTSP Headers ........................................  132
   20.4.1     Description .........................................  132
   20.4.2     Registering New Headers with IANA ...................  132
   20.4.3     Registered entries ..................................  132
   20.5       Transport Header registries .........................  133
   20.5.1     Transport Protocols .................................  133
   20.5.2     Profile .............................................  133
   20.5.3     Lower Transport .....................................  134
   20.5.4     Transport modes .....................................  134
   20.6       Cache Directive Extensions ..........................  135
   20.7       Accept-Credentials policies .........................  135



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   20.8       URI Schemes .........................................  136
   20.9       SDP attributes ......................................  136
   A          RTSP Protocol State Machine .........................  137
   A.1        States ..............................................  137
   A.2        State variables .....................................  138
   A.3        Abbreviations .......................................  138
   A.4        State Tables ........................................  138
   B          Media Transport Alternatives ........................  141
   B.1        RTP .................................................  142
   B.1.1      AVP .................................................  142
   B.1.2      AVP/UDP .............................................  142
   B.1.3      AVP/TCP .............................................  144
   B.1.4      Handling NPT Jumps in the RTP Media Layer ...........  144
   B.1.5      Handling RTP Timestamps after PAUSE .................  147
   B.1.6      RTSP / RTP Integration ..............................  149
   B.1.7      Scaling with RTP ....................................  149
   B.1.8      Maintaining NPT synchronization with RTP
   timestamps .....................................................  150
   B.1.9      Continuous Audio ....................................  150
   B.1.10     Multiple Sources in an RTP Session ..................  150
   B.1.11     Usage of SSRCs and the RTCP BYE Message During an
   RTSP Session ...................................................  150
   B.2        Future Additions ....................................  151
   C          Use of SDP for RTSP Session Descriptions ............  151
   C.1        Definitions .........................................  151
   C.1.1      Control URI .........................................  152
   C.1.2      Media Streams .......................................  153
   C.1.3      Payload Type(s) .....................................  153
   C.1.4      Format-Specific Parameters ..........................  154
   C.1.5      Range of Presentation ...............................  154
   C.1.6      Time of Availability ................................  154
   C.1.7      Connection Information ..............................  155
   C.1.8      Entity Tag ..........................................  155
   C.2        Aggregate Control Not Available .....................  156
   C.3        Aggregate Control Available .........................  156
   C.4        RTSP external SDP delivery ..........................  157
   D          Minimal RTSP implementation .........................  158
   D.1        Client ..............................................  158
   D.1.1      Basic Playback ......................................  159
   D.1.2      Authentication-enabled ..............................  159
   D.2        Server ..............................................  159
   D.2.1      Basic Playback ......................................  160
   D.2.2      Authentication-enabled ..............................  161
   E          Requirements for Unreliable Transport of RTSP
   messages .......................................................  161
   F          Backwards Compatibility Considerations ..............  162
   F.1        Requirement on Pause before Play in Play mode .......  162
   F.2        Usage of persistent connections .....................  163



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   G          Open Issues .........................................  163
   H          Changes .............................................  164
   H.1        Issues Addressed ....................................  164
   H.2        Changes made to the protocol and specification ......  165
   I          Author Addresses ....................................  170
   J          Contributors ........................................  171
   K          Acknowledgements ....................................  171
   L          Normative References ................................  172
   M          Informative References ..............................  173









































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1 Introduction

1.1 RTSP Specification Update

   This document is a draft to an update of RTSP, a proposed standard
   defined in RFC 2326  [1]. The goal the update is to progress RTSP to
   draft standard status. Many flaws have been identified in RTSP since
   its publication. While this draft tries to address these flaws, not
   all known issues have been resolved.  Appendix  H catalogs the issues
   that have already been addressed. Known open issues are listed in
   appendix  G.

   The possibility of progressing RTSP to draft standard without
   republishing RTSP as a proposed standard depends on the changes
   necessary to make the protocol work.

   A list of bugs against the specification is available at
   "http://rtspspec.sourceforge.net". These bugs should be taken into
   account when reading this specification. Input on the unresolved bugs
   and other issues can be sent via e-mail to the MMUSIC WG's mailing
   list mmusic@ietf.org and the authors.

   Not all of the contents of RFC 2326 are part of this draft. In an
   attempt to prevent bloat, the specification has been reduced and
   split. The content of this draft is the core specification of the
   protocol. It contains the general idea behind RTSP and the basic
   functionality necessary to establish an on-demand play-back session.
   It also contains the mechanisms for extending the protocol. Any other
   functionality will be published as extension documents. The Working
   group is currently working on:

        o NAT and FW traversal mechanisms for RTSP are described in a
          document called "How to make Real-Time Streaming Protocol
          (RTSP) traverse Network Address Translators (NAT) and interact
          with Firewalls." [23].

   There have also been discussion or proposals about the following
   extensions to RTSP:

        o Mute and Unmute Extension [24].

        o RTSP Stream Switching [25].

        o Live Streaming Relays [26].

        o Unreliable transport of RTSP messages (rtspu).

        o The Record functionality.



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        o A text body type with suitable syntax for basic parameters to
          be used in SET_PARAMETER, and GET_PARAMETER. Including IANA
          registry within the defined name space.

        o An RTSP MIB.

1.2 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   single or several time-synchronized streams of continuous media such
   as audio and video. Put simply, RTSP acts as a "network remote
   control" for multimedia servers.

   There is no notion of an RTSP connection in the protocol. Instead, an
   RTSP server maintains a session labelled by an identifier to
   associate groups of media streams and their states. An RTSP session
   is not tied to a transport-level connection such as a TCP connection.
   During a session, a client may open and close many reliable transport
   connections to the server to issue RTSP requests for that session.

   This memorandum describes the use of RTSP over a reliable connection
   based transport level protocol such as TCP. RTSP may be implemented
   over an unreliable connectionless transport protocol such as UDP.
   While nothing in RTSP precludes this, additional definition of this
   problem area needs to be handled as an extension to the core
   specification.


        The mechanisms of RTSP's operation over UDP were left out
        of this spec. because they were poorly defined in RFC 2326
        [1] and the tradeoff in size and complexity of this spec.
        for a small gain in a targeted problem space was not deemed
        justifiable.

   The set of streams to be controlled in an RTSP session is defined by
   a presentation description. This memorandum does not define a format
   for the presentation description. However appendix C defines how SDP
   [2] is used for this purpose. The streams controlled by RTSP may use
   RTP [3] for their data transport, but the operation of RTSP does not
   depend on the transport mechanism used to carry continuous media.
   RTSP is intentionally similar in syntax and operation to HTTP/1.1 [4]
   so that extension mechanisms to HTTP can in most cases also be added
   to RTSP.  However, RTSP differs in a number of important aspects from
   HTTP:

        o RTSP introduces a number of new methods and has a different
          protocol identifier.




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        o RTSP has the notion of a session built into the protocol.

        o An RTSP server needs to maintain state by default in almost
          all cases, as opposed to the stateless nature of HTTP.

        o Both an RTSP server and client can issue requests.

        o Data is usually carried out-of-band by a different protocol.
          Session descriptions returned in a DESCRIBE response (see
          Section 11.2) and interleaving of RTP with RTSP over TCP are
          exceptions to this rule (see Section 12).

        o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
          8859-1, consistent with HTML internationalization efforts
          [27].

        o The Request-URI always contains the absolute URI. Because of
          backward compatibility with a historical blunder, HTTP/1.1 [4]
          carries only the absolute path in the request and puts the
          host name in a separate header field.


             This makes "virtual hosting" easier, where a single
             host with one IP address hosts several document trees.

   The protocol supports the following operations:

        Retrieval of media from media server: The client can either
             request a presentation description via RTSP DESCRIBE, HTTP
             or some other method. If the presentation is being
             multicast, the presentation description contains the
             multicast addresses and ports to be used for the continuous
             media. If the presentation is to be sent only to the client
             via unicast, the client provides the destination for
             security reasons.


        Invitation of a media server to a conference: A media server can
             be "invited" to join an existing conference to play back
             media into the presentation. This mode is useful for
             example distributed teaching applications. Several parties
             in the conference may take turns "pushing the remote
             control buttons".

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [4].

1.3 Notational Conventions



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   Since many of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [4]).

   All the mechanisms specified in this document are described in both
   prose and the augmented Backus-Naur form (BNF) described in detail in
   RFC 2234 [5].

   Indented and smaller-type paragraphs are used to provide informative
   background and motivation. This is intended to give readers who were
   not involved with the formulation of the specification an
   understanding of why things are the way they are in RTSP.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [6].

   The word, "unspecified" is used to indicate functionality or features
   that are not defined in this specification. Such functionality cannot
   be used in a standardized manner without further definition and
   review in an extension specification to RTSP.

1.4 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [4]. Terms not
   listed here are defined as in HTTP/1.1.

        Aggregate control: The concept of controlling multiple streams
             using a single timeline, generally maintained by the
             server. A client, for example, uses aggregate control when
             it issues a single play or pause message to simultaneously
             control both the audio and video in a movie.

        Aggregate control URI: The URI used in an RTSP request to refer
             to and control an aggregated session. It normally, but not
             always, corresponds to the presentation URI specified in
             the session description. See Section  11.3 for more
             information.

        Conference: a multiparty, multimedia presentation, where "multi"
             implies greater than or equal to one.

        Client: The client requests media service from the media server.

        Connection: A transport layer virtual circuit established
             between two programs for the purpose of communication.




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        Container file: A file which may contain multiple media streams
             which often constitutes a presentation when played
             together. The concept of a container file is not embedded
             in the protocol.  However, RTSP servers may offer aggregate
             control on the media streams within these files.


        Continuous media: Data where there is a timing relationship
             between source and sink; that is, the sink needs to
             reproduce the timing relationship that existed at the
             source. The most common examples of continuous media are
             audio and motion video.  Continuous media can be real-time
             (interactive or conversational), where there is a "tight"
             timing relationship between source and sink, or streaming
             (playback), where the relationship is less strict.

        Entity: The information transferred as the payload of a request
             or response. An entity consists of meta-information in the
             form of entity-header fields and content in the form of an
             entity-body, as described in Section 8.

        Feature-tag: A tag representing a certain set of functionality,
             i.e. a feature.

        Live: Normally used to describe a presentation or session with
             media coming from ongoing event. This generally results in
             that the session has a unbound or only loosely defined
             duration, and that no seek operations are possible.

        Media initialization: Datatype/codec specific initialization.
             This includes such things as clock rates, color tables,
             etc. Any transport-independent information which is
             required by a client for playback of a media stream occurs
             in the media initialization phase of stream setup.

        Media parameter: Parameter specific to a media type that may be
             changed before or during stream playback.

        Media server: The server providing playback services for one or
             more media streams. Different media streams within a
             presentation may originate from different media servers.  A
             media server may reside on the same host or on a different
             host from which the presentation is invoked.

        Media server indirection: Redirection of a media client to a
             different media server.





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        (Media) stream: A single media instance, e.g., an audio stream
             or a video stream as well as a single whiteboard or shared
             application group. When using RTP, a stream consists of all
             RTP and RTCP packets created by a source within an RTP
             session.

        Message: The basic unit of RTSP communication, consisting of a
             structured sequence of octets matching the syntax defined
             in Section 18 and transmitted over a connection or a
             connectionless transport.

        Non-Aggregated Control: Control of a single media stream.  Only
             possible in RTSP sessions with a single media.

        Participant: Member of a conference. A participant may be a
             machine, e.g., a playback server.


        Presentation: A set of one or more streams presented to the
             client as a complete media feed and described by a
             presentation description as defined below. Presentations
             with more than one media stream is often handled in RTSP
             under aggregate control.

        Presentation description: A presentation description contains
             information about one or more media streams within a
             presentation, such as the set of encodings, network
             addresses and information about the content. Other IETF
             protocols such as SDP (RFC 2327 [2]) use the term "session"
             for a presentation. The presentation description may take
             several different formats, including but not limited to the
             session description protocol format, SDP.

        Response: An RTSP response. If an HTTP response is meant, that
             is indicated explicitly.

        Request: An RTSP request. If an HTTP request is meant, that is
             indicated explicitly.

        Request-URI: The URI used in a request to indicate the resource
             on which the request is to be performed.

        RTSP agent: Refers to either an RTSP client, an RTSP server, or
             an RTSP Proxy. In this specification, there are many
             capabilities that are common to these three entities such
             as the capability to send requests or receive responses.
             This term will be used when describing functionality that
             is applicable to all three of these entities.



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        RTSP session: A stateful abstraction upon which the main control
             methods of RTSP operate. An RTSP session is a server
             entity; it is created, maintained and destroyed by the
             server. It is established by an RTSP server upon the
             completion of a successful SETUP request (when 200 OK
             response is sent) and is labelled by a session identifier
             at that time. The session exists until timed out by the
             server or explicitly removed by a TEARDOWN request. An RTSP
             session is a stateful entity; an RTSP server maintains an
             explicit session state machine (see Appendix A) where most
             state transitions are triggered by client requests. The
             existence of a session implies the existence of state about
             the session's media streams and their respective transport
             mechanisms. A given session can have zero or more media
             streams associated with it. An RTSP server uses the session
             to aggregate control over multiple media streams.

        Transport initialization: The negotiation of transport
             information (e.g., port numbers, transport protocols)
             between the client and the server.

        URI: Universal Resource Identifier, see RFC 2396 [13]. In RTSP
             the used URIs are as general rule in fact URI's as they
             gives an location for the resource. Therefore although RTSP
             URIs are a subset of URIs, they will be refered as URIs.

        URI: Universal Resource Locator, is an URI which identifies the
             resource through its primary access mechanism, rather than
             identifying the resource by name or by some other
             attribute(s) of that resource.

1.5 Protocol Properties

   RTSP has the following properties:

        Extendable: New methods and parameters can be easily added to
             RTSP.

        Easy to parse: RTSP can be parsed by standard HTTP or MIME
             parsers.

        Secure: RTSP re-uses web security mechanisms, either at the
             transport level (TLS, RFC 2246 [7]) or within the protocol
             itself. All HTTP authentication mechanisms such as basic
             (RFC 2616 [4]) and digest authentication (RFC 2617 [8]) are
             directly applicable.





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        Transport-independent: RTSP does not preclude the use of an
             unreliable datagram protocol (UDP) (RFC 768 [9]) as it
             would be possible to implement application-level
             reliability.  The use of a connectionless datagram protocol
             such as UDP requires additional definition that may be
             provided as extensions to the core RTSP specification. The
             usage of the reliable stream protocol TCP (RFC 793 [10])
             and secured reliable stream protocol TLS over TCP [7] is
             what is currently defined as transport protocol of RTSP
             messages.

        Multi-server capable: Each media stream within a presentation
             can reside on a different server. The client automatically
             establishes several concurrent control sessions with the
             different media servers.  Media synchronization is
             performed at the transport level.

        Separation of stream control and conference initiation:  Stream
             control is divorced from inviting a media server to a
             conference. In particular, SIP [28] or H.323 [29] may be
             used to invite a server to a conference.

        Suitable for professional applications: RTSP supports frame-
             level accuracy through SMPTE time stamps to allow remote
             digital editing.

        Presentation description neutral: The protocol does not impose a
             particular presentation description or metafile format and
             can convey the type of format to be used. However, the
             presentation description is required to contain at least
             one RTSP URI.

        Proxy and firewall friendly: The protocol should be readily
             handled by both application and transport-layer (SOCKS
             [30]) firewalls. A firewall may need to understand the
             SETUP method to open a "hole" for the media stream.

        HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
             that the existing infrastructure can be reused. This
             infrastructure includes PICS (Platform for Internet Content
             Selection [31,32]) for associating labels with content.
             However, RTSP does not just add methods to HTTP since the
             controlling continuous media requires server state in most
             cases.

        Appropriate server control: If a client can start a stream, it
             needs to be able to stop a stream. Servers should not start
             streaming to clients in such a way that clients cannot stop



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             the stream.

        Transport negotiation: The client can negotiate the transport
             method prior to actually needing to process a continuous
             media stream.

1.6 Extending RTSP

   Since not all media servers have the same functionality, media
   servers by necessity will support different sets of requests. For
   example:

        o A server may not be capable of seeking (absolute positioning)
          if it is to support live events only.

        o Some servers may not support setting stream parameters and
          thus not support GET_PARAMETER and SET_PARAMETER.


        o Some server may support an RTSP extension, for example the
          currently proposed "end of stream" indication.

   A server SHOULD implement all header fields described in Section 14.

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server. This situation is similar in HTTP/1.1 [4],
   where the methods described in [H19.5] are not likely to be supported
   across all servers.

   RTSP can be extended in three ways, listed here in order of the
   magnitude of changes supported:

        o Existing methods can be extended with new parameters, e.g.
          headers, as long as these parameters can be safely ignored by
          the recipient. If the client needs negative acknowledgement
          when a method extension is not supported, a tag corresponding
          to the extension may be added in the Require: field (see
          Section 14.37).

        o New methods can be added. If the recipient of the message does
          not understand the request, it responds with error code 501
          (Not Implemented) and the sender should not attempt to use
          this method again. A client may also use the OPTIONS method to
          inquire about methods supported by the server. The server MUST
          list the methods it supports using the Public response header.

        o A new version of the protocol can be defined, allowing almost
          all aspects (except the position of the protocol version



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          number) to change.

   The basic capability discovery mechanism can be used to both discover
   support for a certain feature and to ensure that a feature is
   available when performing a request. For detailed explanation of this
   see section  10.

1.7 Overall Operation

   Each presentation and media stream is identified by an RTSP URI.  The
   overall presentation and the properties of the media the presentation
   is made up of are defined by a presentation description file, the
   format of which is outside the scope of this specification. The
   presentation description file may be obtained by the client using
   HTTP or other means such as email and may not necessarily be stored
   on the media server.

   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which
   maintains a common time axis. For simplicity of exposition and
   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation. A presentation
   may contain several media streams.

   The presentation description file contains a description of the media
   streams making up the presentation, including their encodings,
   language, and other parameters that enable the client to choose the
   most appropriate combination of media. In this presentation
   description, each media stream that is individually controllable by
   RTSP is identified by an RTSP URI, which points to the media server
   handling that particular media stream and names the stream stored on
   that server. Several media streams can be located on different
   servers; for example, audio and video streams can be split across
   servers for load sharing. The description also enumerates which
   transport methods the server is capable of.

   Besides the media parameters, the network destination address and
   port need to be determined. Several modes of operation can be
   distinguished:

        Unicast: The media is transmitted to the source of the RTSP
             request, with the port number chosen by the client.
             Alternatively, the media is transmitted on the same
             reliable stream as RTSP.

        Multicast, server chooses address: The media server picks the
             multicast address and port. This is the typical case for a
             live or near-media-on-demand transmission.



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        Multicast, client chooses address: If the server is to
             participate in an existing multicast conference, the
             multicast address, port and encryption key are given by the
             conference description, established by means outside the
             scope of this specification, for example by a SIP created
             conference.

1.8 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may be
   transported on a TCP connection while the media data is conveyed via
   UDP. Thus, data delivery continues even if no RTSP requests are
   received by the media server. Also, during its lifetime, a single
   media stream may be controlled by RTSP requests issued sequentially
   on different TCP connections. Therefore, the server needs to maintain
   "session state" to be able to correlate RTSP requests with a stream.
   The state transitions are described in Appendix  A.

   Many methods in RTSP do not contribute to state. However, the
   following play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING
   and TEARDOWN.

        SETUP: Causes the server to allocate resources for a stream and
             create an RTSP session.

        PLAY: Starts data transmission on a stream allocated via SETUP.

        PAUSE: Temporarily halts a stream without freeing server
             resources.

        REDIRECT: Indicates that the session should be moved to new
             server / location

        PING: Prevents the identified session from being timed out.

        TEARDOWN: Frees resources associated with the stream.  The RTSP
             session ceases to exist on the server.

   RTSP methods that contribute to state use the Session header field
   (Section 14.42) to identify the RTSP session whose state is being
   manipulated. The server generates session identifiers in response to
   SETUP requests (Section 11.3).

1.9 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may



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   interact with HTTP in that the initial contact with streaming content
   is often to be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces round trips in web-browser-based scenarios, yet also allows
   for stand alone RTSP servers and clients which do not rely on HTTP at
   all. However, RTSP differs fundamentally from HTTP in that most data
   delivery takes place out-of-band in a different protocol. HTTP is an
   asymmetric protocol where the client issues requests and the server
   responds. In RTSP, both the media client and media server can issue
   requests. RTSP requests are also stateful; they may set parameters
   and continue to control a media stream long after the request has
   been acknowledged.


        Re-using HTTP functionality has advantages in at least two
        areas, namely security and proxies. The requirements are
        very similar, so having the ability to adopt HTTP work on
        caches, proxies and authentication is valuable.

   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams. Session Description Protocol (SDP)
   [2] is generally the format of choice; however, RTSP is not bound to
   it. For data delivery, most real-time media will use RTP as a
   transport protocol. While RTSP works well with RTP, it is not tied to
   RTP.

2 RTSP Use Cases

   This section describes some of the use cases for RTSP. They are
   listed in descending order of importance in regards to ensuring that
   all necessary functionality is present. This specification does only
   fully support usage of the two first. Also in these first two cases
   are there special cases that will not be supported without
   extensions, e.g. the redirection of media to another address than the
   controlling entity.

2.1 On-demand Playback of Stored Content

   An RTSP capable server stores content suitable for being streamed to
   a client. A client desiring playback of any of the stored content
   uses RTSP to set up the media transport required for the desired
   content. Then RTSP is used to initiate, halt and manipulate the
   transmission of the content. There are also requirement on being able
   to use RTSP to carry necessary description and synchronization
   information for the content. The above high level description can be



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   broken down into a number of functionalities that RTSP needs to be
   capable of.

        Presentation Description: The possibility to carry
             initialization information about the presentation
             (content), for example, which media codec(s) that are
             needed for the content. Other information that are
             important; how many media stream that the presentation
             contains; what transport protocols to use for the media
             streams; and identifiers for these media streams. This
             information is required before setup of the content is
             possible. The information is also needed by the client to
             determine if it is capable at all to support the content.
             This information is not required to be sent using RTSP,
             instead other external protocols can be utilized to
             transport presentation descriptions. Two good examples are
             the use of HTTP  [4] or email to fetch or receive
             presentation descriptions like SDP  [2]. .XP Setup:
             Performing setup of some or all of the media streams in a
             presentation. The setup itself consist of determining which
             protocols for media transport to use; the necessary
             parameters for the protocol, like addresses and ports. .XP
             Control of Transmission: After the necessary media streams
             has been established the client can request the server to
             start transmitting the content. There is need to allow the
             client to arbitrary times start or stop the transmission of
             the content. There are also exist need to be able to start
             the transmission at an any point in the timeline of the
             presentation. .XP Synchronization: For media transport
             protocols like RTP  [17] it might be beneficial to carry
             synchronization information within RTSP. Either due to the
             lack of inter media synchronization within the protocol
             itself, or the potential delay before the synchronization
             is established (which is the case for RTP when using RTCP).
             .XP Termination There is also need to be able to terminate
             the established contexts.
        For this use cases there is a number of assumption about how it
        works. These are listed below:

        On-Demand content: The content available is stored at the server
             and can be accessed at any time during a time period when
             it is intended to be available. .XP Independent sessions: A
             server is capable of serving a number of clients
             simultaneously, including from the same piece of content at
             different points in that presentations time-line. .XP
             Unicast Transport: Content for each individual client is
             transmitted to them using unicast traffic.
        It is also possible to redirect the media traffic to another



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        destination than where the entity controlling traffic uses.
        However allowing this without appropriate mechanisms for
        checking that the destination approves of this is a denial of
        service threat.

2.2 Unicast distribution of Live Content

   This use cases is not that different from the above on-demand content
   case (see section  2.1. The difference is really the restriction the
   content itself establish. Live content is continuously distributed as
   it becomes available from a source, i.e. the main difference to on-
   demand is that one starts distributing content before the end of it
   has become available to the server. In many cases the consumer of
   live content is only interested in consuming what is actually happens
   "now", i.e. very similar to broadcast TV. However in this case it is
   assumed that there exist no broadcast or multicast channel to the
   users, and instead the server functions as a distribution node,
   sending the same content to multiple receivers, using unicast traffic
   between server and client. This unicast traffic and the transport
   parameters are individually negotiated for each receiving client.
   Another aspect of live content is that it has often very limited time
   of availability, as it is only is available for the duration of the
   event the content covers. A example of such a live content could for
   example be a music concert, which lasts 2 hour and starts at a
   predetermined time. Thus there is need to announce when and for how
   long the live content is available.

2.3 On-demand Playback using Multicast

   It is possible to use RTSP to request that media is delivered to a
   multicast group. The entity setting up the session (the controller)
   will then control when and what media that is delivered to the group.
   Also this use case has some potential for denial of service attacks,
   in this case flooding any multicast group. Therefore there is need
   for a mechanism indicating that the group actually accepts the
   traffic from the RTSP server. An open issue in this use case is how
   one ensures that all receivers listening to the multicast or
   broadcast receives the session presentation configuring the
   receivers.

2.4 Inviting a RTSP server into a conference

   If one has an established conference or group session, it is possible
   to have a RTSP server distribute media to the whole group. The
   transmission to the group is simplest controlled by a single
   participant or leader of the conference. Shared control might be
   possible, but would require further investigation and possibly
   extensions. There are some protocol mechanisms missing for this



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   scenario. For reasonable complexity in the media transmission stage,
   this use case assumes that there exist either multicast or a
   conference focus that redistribute media to all participants. In some
   more detail, this use case is intended to be able to handle the
   following scenario: A conference leader or participant (from here
   called the controller) has some pre-stored content on a RTSP server
   that he likes to share with the group. The controller sets up a RTSP
   session at the streaming server for the content the controller likes
   to share. The session description for the content is retrieved to the
   controller. The media destination for the media content is set to the
   shared multicast group or conference focus. When desired by the
   controller, he/she can start and stop the transmission of the media
   to the conference group. There are several issues with this use case
   that is not solved by this core specification for RTSP:

        o Denial of service threat, to avoid a RTSP server from being a
          unknowing participant of a denial of service attack the server
          needs to be able to verify the destinations acceptance for the
          media. Such a mechanism does not yet exist that can be used to
          verify the approval to received media, instead only policies
          can be used, which can be made to work in controlled
          environments. .IP o 2 The problem of distributing the
          presentation description to all participants in the group. To
          enable a media receiver to decode the content correctly the
          media configuration information will need to be distributed
          reliable to all participants. This will most likely require
          support from an external protocol. .IP o 2 Passing the
          control. If it is desired to be able to pass the control of
          the RTSP session between the participants some support will be
          required by an external protocol for the necessary exchange of
          state information and possibly floor control of who is
          controlling the RTSP session.

   So if there interest in this use case further work on the necessary
   extensions has to be performed.

2.5 Live Content using Multicast

   This use case does in its simplest form do not require any use of
   RTSP at all. This is what multicast conferences being announce with
   SAP and SDP are intended to handle. However in use cases where more
   advance features like access control to the multicast session is
   desired, RTSP could be used for session establishment. A client
   desiring to join a live multicasted media session with cryptographic
   (encryption) access control could use RTSP in the following way. The
   source of the session, announces the session and gives all interested
   to join, a RTSP URI. The client connects to the server and requests
   the presentation description allowing for configuration the



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   reception. In this step it is possible to use secured transport for
   the client, and also desired levels of authentication, for example
   for charging purposes or simply access control. An RTSP link also
   allows for load balancing between multiple servers. However if this
   the only thing that occurs it can probably be solved as simple using
   HTTP. However for session where the sender likes to keep track of
   each individual receiver during the session, and possibly use this
   side channel for pushing out key-updates or other side information
   that is desirable to be done on a per receiver basis, and the
   receivers are not know prior to the session start, the state
   establishment that RTSP provides can be beneficial. In this case a
   client would establish a RTSP session to the multicast group. The
   RTSP server will not transmit any media, instead it will simply point
   to the multicast group. However the client and server will be able to
   keep the session alive for as long as the receiver participates in
   the session. Thus enabling for example server to client pushes of
   updates. This use cases will most likely not be able to actually
   implement some extensions in relation to the server to client push
   mechanism. Here a method like ANNOUNCE might be suitable, however it
   will require a RTSP extension to revive the method.

3 Protocol Parameters

3.1 RTSP Version

   HTTP Specification Section [H3.1] applies, with HTTP replaced by
   RTSP. This specification defines version 1.0 of RTSP.

3.2 RTSP URI

   The "rtsp", "rtsps" schemes are used to refer to network resources
   via the RTSP protocol. This section defines the scheme-specific
   syntax and semantics for RTSP URIs. The RTSP URI is case sensitive.
   An URI scheme "rtspu" was defined in RFC 2326 for transport of RTSP
   messages over unreliable transport (UDP) and is currently deprecated
   and reserved, and MUST NOT be used . See Appendix  E for further
   information.

   Informative RTSP URI syntax:

   rtsp[u|s]://host[:port]/abspath[?query]#fragment


   See section  18.2.1 for the formal definition of the RTSP URI syntax.

   The fragment identifier is used as defined in section 4.1 of [13],
   i.e. the fragment is to be stripped from the URI by the requestor and
   not included in the request. The user agent also needs to interpret



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   the value of the fragment based on the media type the request relates
   to, i.e. the media type indicated in Content-Type header in the
   response to DESCRIBE.

   The syntax of any URI query string is unspecified and responder
   (usually the server) specific. As it is from the requestor an opaque
   string, it needs to be handled as such.

   The URI scheme rtsp requires that commands are issued via a reliable
   protocol (within the Internet, TCP), while the scheme rtsps
   identifies a reliable transport using secure transport (TLS [7]).

   If the no port number is provided in the URI, port number 554 SHALL
   be used. The semantics are that the identified resource can be
   controlled by RTSP at the server listening for TCP (scheme "rtsp")
   connections on that port of host, and the Request-URI for the
   resource is rtsp_URI.  For the scheme rtsps the TCP and UDP port 322
   is registered and SHALL be assumed.

   The use of IP addresses in URIs SHOULD be avoided whenever possible
   (see RFC 1924 [11]). Note: Using qualified domain names in any URI is
   one requirement for making it possible for RFC 2326 implementations
   of RTSP to use IPv6. This specification is updated to allow for
   literal IPv6 addresses in RTSP URIs using the host specification in
   RFC 2732 [12].

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions [H3.2] of
   URIs (RFC 2396 [13]). URIs may refer to a stream or an aggregate of
   streams, i.e., a presentation. Accordingly, requests described in
   Section 11 can apply to either the whole presentation or an
   individual stream within the presentation. Note that some request
   methods can only be applied to streams, not presentations and vice
   versa.

   For example, the RTSP URI:

     rtsp://media.example.com:554/twister/audiotrack


   identifies the audio stream within the presentation "twister", which
   can be controlled via RTSP requests issued over a TCP connection to
   port 554 of host media.example.com

   Also, the RTSP URI:

     rtsp://media.example.com:554/twister




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   identifies the presentation "twister", which may be composed of audio
   and video streams.


        This does not imply a standard way to reference streams in
        URIs. The presentation description defines the hierarchical
        relationships in the presentation and the URIs for the
        individual streams. A presentation description may name a
        stream "a.mov" and the whole presentation "b.mov".

   The path components of the RTSP URI are opaque to the client and do
   not imply any particular file system structure for the server.


        This decoupling also allows presentation descriptions to be
        used with non-RTSP media control protocols simply by
        replacing the scheme in the URI.

3.3 Session Identifiers

   Session identifiers are strings of any arbitrary length. A session
   identifier MUST be chosen randomly and MUST be at least eight
   characters long to make guessing it more difficult. (See Section 19.)

3.4 SMPTE Relative Timestamps

   A SMPTE relative timestamp expresses time relative to the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
                  hours:minutes:seconds:frames.subframes,
   with the origin at the start of the clip. The default smpte format
   is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
   Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
   use of alternative use of "smpte time". For the "frames" field in the
   time value can assume the values 0 through 29. The difference between
   30 and 29.97 frames per second is handled by dropping the first two
   frame indices (values 00 and 01) of every minute, except every tenth
   minute. If the frame value is zero, it may be omitted. Subframes are
   measured in one-hundredth of a frame.

   Examples:

     smpte=10:12:33:20-
     smpte=10:07:33-
     smpte=10:07:00-10:07:33:05.01
     smpte-25=10:07:00-10:07:33:05.01





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3.5 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation, not to be confused
   with the Network Time Protocol (NTP) [33]. The timestamp consists of
   a decimal fraction. The part left of the decimal may be expressed in
   either seconds or hours, minutes, and seconds. The part right of the
   decimal point measures fractions of a second.

   The beginning of a presentation corresponds to 0.0 seconds.  Negative
   values are not defined. The special constant now is defined as the
   current instant of a live type event. It MAY only be used for live
   type events, and SHALL NOT be used for on-demand content.

   NPT is defined as in DSM-CC [34]:  "Intuitively, NPT is the clock the
   viewer associates with a program. It is often digitally displayed on
   a VCR. NPT advances normally when in normal play mode (scale = 1),
   advances at a faster rate when in fast scan forward (high positive
   scale ratio), decrements when in scan reverse (high negative scale
   ratio) and is fixed in pause mode. NPT is (logically) equivalent to
   SMPTE time codes."

   Examples:

     npt=123.45-125
     npt=12:05:35.3-
     npt=now-




        The syntax conforms to ISO 8601 [35]. The npt-sec notation
        is optimized for automatic generation, the ntp-hhmmss
        notation for consumption by human readers. The "now"
        constant allows clients to request to receive the live feed
        rather than the stored or time-delayed version. This is
        needed since neither absolute time nor zero time are
        appropriate for this case.

3.6 Absolute Time

   Absolute time is expressed as ISO 8601 [35] timestamps, using UTC
   (GMT). Fractions of a second may be indicated.

   Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
   UTC:

     19961108T143720.25Z



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3.7 Feature-tags

   Feature-tags are unique identifiers used to designate features in
   RTSP. These tags are used in Require (Section 14.37), Proxy-Require
   (Section 14.31), Proxy-Supported (Section  14.32), Unsupported
   (Section  14.46), and Supported (Section 14.43) header fields.

   Feature tag needs to indicate if they apply to servers only, proxies
   only, or both server and proxies.

   The creator of a new RTSP feature-tag should either prefix the
   feature-tag with a reverse domain name (e.g.,
   "com.example.mynewfeature" is an apt name for a feature whose
   inventor can be reached at "example.com"), or register the new
   feature-tag with the Internet Assigned Numbers Authority (IANA), see
   IANA Section  20.

   The usage of feature tags are further described in section 10 that
   deals with capability handling.

3.8 Entity Tags

   Entity tags are opaque strings that are used to compare two entities
   from the same resource, for example in caches or to optimize setup
   after a redirect. Further explanation is present in [H3.11]. For
   explanation on how to compare Entity tags see [H13.3]. Entity tags
   can be carried in the ETag header (see section 14.21) or in SDP (see
   section C.1.8).

   Entity tags are used in RTSP to make some methods conditional. The
   methods are made conditional through the inclusion of headers, see
   14.25 and 14.27.

4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [14]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by
   themselves as line terminators.


        Text-based protocols make it easier to add optional
        parameters in a self-describing manner. Since the number of
        parameters and the frequency of commands is low, processing
        efficiency is not a concern. Text-based protocols, if done
        carefully, also allow easy implementation of research
        prototypes in scripting languages such as Tcl, Visual Basic
        and Perl.



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   The 10646 character set avoids tricky character set switching, but is
   invisible to the application as long as US-ASCII is being used.  This
   is also the encoding used for RTCP. ISO 8859-1 translates directly
   into Unicode with a high-order octet of zero. ISO 8859-1 characters
   with the most-significant bit set are represented as 1100001x
   10xxxxxx. (See RFC 2279 [14])

   Requests contain methods, the object the method is operating upon and
   parameters to further describe the method. Methods are idempotent,
   unless otherwise noted. Methods are also designed to require little
   or no state maintenance at the media server.

4.1 Message Types

   See [H4.1].

4.2 Message Headers

   See [H4.2].

4.3 Message Body

   See [H4.3]

4.4 Message Length

   When a message body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

        1.   Any response message which MUST NOT include a message body
             (such as the 1xx, 204, and 304 responses) is always
             terminated by the first empty line after the header fields,
             regardless of the entity-header fields present in the
             message. (Note: An empty line consists of only CRLF.)

        2.   If a Content-Length header field (section 14.16) is
             present, its value in bytes represents the length of the
             message-body. If this header field is not present, a value
             of zero is assumed.

   Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
   transfer coding(see [H3.6.1]) and requires the presence of the
   Content-Length header field.


        Given the moderate length of presentation descriptions
        returned, the server should always be able to determine its
        length, even if it is generated dynamically, making the



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        chunked transfer encoding unnecessary.

5 General Header Fields

   See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
   and Warning headers are not defined. RTSP further defines the CSeq,
   and Timestamp. The general headers are listed in table 1:


                     Header Name    Comment
                     _________________________________
                     Cache-Control  See section 14.10
                     Connection     See section 14.11
                     CSeq           See section 14.19
                     Date           See section 14.20
                     Supported      See section 14.43
                     Timestamp      See section 14.44
                     Via            See section 14.49


   Table 1: The General headers used in RTSP.


6 Request

   A request messages uses the format outlined below, regardless of the
   direction of a request, client to server or server to client:

        o Request line, containing the method to be applied to the
          resource, the identifier of the resource, and the protocol
          version in use;

        o zero or more Header lines, that can be of the following types:
          general (Section 5), request (Section 6.2), or entity (Section
          8.1);

        o One empty line (CR/LF) to indicate the end of the header
          section;

        o Optionally a message body (entity), consisting of one or more
          lines. the length of the message body in number of bytes is
          indicated by the Content-Length entity header.


6.1 Request Line

   The request line provides the key information about the request:
   What method, on what resources and using which RTSP version. The
   methods that is defined by this specification can be seen in Table


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   6.1. The resource is identified through an absolute RTSP URI (see
   section  3.2.


   <Method> SP <Request-URI> SP <RTSP-Version> CRLF


   Please note: The request line's syntax can't be freely changed in
   future versions of RTSP, as this line indicates the version of the
   messages and need to be parsable also by older versions.

   Note that in contrast to HTTP/1.1 [4], RTSP requests always contain
   the absolute URI (that is, including the scheme, host and port)
   rather than just the absolute path.


        HTTP/1.1 requires servers to understand the absolute URI,
        but clients are supposed to use the Host request header.
        This is purely needed for backward-compatibility with
        HTTP/1.0 servers, a consideration that does not apply to
        RTSP.


                     Method         Defined In Section
                     _________________________________
                     DESCRIBE       Section 11.2
                     GET_PARAMETER  Section 11.7
                     OPTIONS        Section 11.1
                     PAUSE          Section 11.5
                     PLAY           Section 11.4
                     PING           Section 11.10
                     REDIRECT       Section 11.9
                     SETUP          Section 11.3
                     SET_PARAMETER  Section 11.8
                     TEARDOWN       Section 11.6


   Table 2: The RTSP Methods


   The asterisk "*" in the Request-URI means that the request does not
   apply to a particular resource, but to the server or proxy itself,
   and is only allowed when the method used does not necessarily apply
   to a resource.

   One example would be as follows:


     OPTIONS * RTSP/1.0


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   An OPTIONS in this form will determine the capabilities of the server
   or the proxy that first receives the request. If one needs to address
   the server explicitly, then one should use an absolute URI with the
   server's address.


     OPTIONS rtsp://example.com RTSP/1.0



6.2 Request Header Fields

   The RTSP headers in Table 3 can be included in a request with the
   purpose to give further define how the request should be fulfilled. A
   request header MAY also be response header, see section  7.1.2.


                   Header             Defined in Section
                   _____________________________________
                   Accept             Section 14.1
                   Accept-Encoding    Section 14.3
                   Accept-Language    Section 14.4
                   Authorization      Section 14.7
                   Bandwidth          Section 14.8
                   Blocksize          Section 14.9
                   From               Section 14.23
                   If-Match           Section 14.25
                   If-Modified-Since  Section 14.26
                   If-None-Match      Section 14.27
                   Proxy-Require      Section 14.31
                   Range              Section 14.34
                   Referer            Section 14.35
                   Require            Section 14.37
                   Scale              Section 14.39
                   Session            Section 14.42
                   Speed              Section 14.40
                   Supported          Section 14.43
                   Transport          Section 14.45
                   User-Agent         Section 14.47


   Table 3: The RTSP request headers


7 Response

   [H6] applies except that HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   HTTP codes. The valid response codes and the methods they can be used


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   with are defined in Table 4.

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.

7.1 Status-Line

   The first line of a Response message is the Status-Line, consisting
   of the protocol version followed by a numeric status code, and the
   textual phrase associated with the status code, with each element
   separated by SP characters. No CR or LF is allowed except in the
   final CRLF sequence.


   <RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF


7.1.1 Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in Section 13. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The client is not required to examine or display the Reason-
   Phrase.

   The first digit of the Status-Code defines the class of response. The
   last two digits do not have any categorization role.  There are 5
   values for the first digit:

        o 1xx: Informational - Request received, continuing process

        o 2xx: Success - The action was successfully received,
          understood, and accepted

        o 3rr: Redirection - Further action needs to be taken in order
          to complete the request

        o 4xx: Client Error - The request contains bad syntax or cannot
          be fulfilled

        o 5xx: Server Error - The server failed to fulfill an apparently
          valid request

   The individual values of the numeric status codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrases, are
   presented in table  4. The reason phrases listed here are only



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   recommended; they may be replaced by local equivalents without
   affecting the protocol. Note that RTSP adopts most HTTP/1.1 [4]
   status codes and adds RTSP-specific status codes starting at x50 to
   avoid conflicts with newly defined HTTP status codes.

   RTSP status codes are extensible. RTSP applications are not required
   to understand the meaning of all registered status codes, though such
   understanding is obviously desirable. However, applications MUST
   understand the class of any status code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 status code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if an
   unrecognized status code of 431 is received by the client, it can
   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code. In such
   cases, user agents SHOULD present to the user the entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.


7.1.2 Response Header Fields

   The response-header fields allow the request recipient to pass
   additional information about the response which cannot be placed in
   the Status-Line. These header fields give information about the
   server and about further access to the resource identified by the
   Request-URI. All headers currently being classified as response
   headers are listed in table  5.


   Response-header field names can be extended reliably only in
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of response-
   header fields if all parties in the communication recognize them to
   be response-header fields. Unrecognized header fields are treated as
   entity-header fields.

8 Entity

   Request and Response messages MAY transfer an entity if not otherwise
   restricted by the request method or response status code. An entity
   consists of entity-header fields and an entity-body, although some
   responses will only include the entity-headers.

   The SET_PARAMETER, and GET_PARAMETER request and response, and
   DESCRIBE response MAY have an entity. All 4xx and 5xx responses MAY
   also have an entity.




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   In this section, both sender and recipient refer to either the client
   or the server, depending on who sends and who receives the entity.

8.1 Entity Header Fields

   Entity-header fields define optional meta-information about the
   entity-body or, if no body is present, about the resource identified
   by the request. The entity header fields are listed in table  8.1.


                   Header            Defined in Section
                   ____________________________________
                   Allow             Section  14.6
                   Content-Base      Section  14.13
                   Content-Encoding  Section  14.14
                   Content-Language  Section  14.15
                   Content-Length    Section  14.16
                   Content-Location  Section  14.17
                   Content-Type      Section  14.18
                   Expires           Section  14.22
                   Last-Modified     Section  14.28


   Table 6: The RTSP entity headers


   The extension-header mechanism allows additional entity-header fields
   to be defined without changing the protocol, but these fields cannot
   be assumed to be recognizable by the recipient. Unrecognized header
   fields SHOULD be ignored by the recipient and forwarded by proxies.

8.2 Entity Body

   See [H7.2] with the addition that an RTSP message with an entity body
   MUST include the Content-Type and Content-Length headers.

9 Connections

   RTSP requests can be transmitted in several different ways:

        o persistent transport connections used for several request-
          response transactions;

        o one connection per request/response transaction;

        o connectionless mode.

   The type of transport is defined by the RTSP URI (Section 3.2). For
   the scheme "rtsp", a connection is assumed, while the scheme "rtsps"


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        Code  Reason                               Method
        __________________________________________________________
        100   Continue                             all

__________________________________________________________
        200   OK                                   all
        201   Created                              RECORD
        250   Low on Storage Space                 RECORD
        __________________________________________________________
        300   Multiple Choices                     all
        301   Moved Permanently                    all
        302   Found                                all
        303   See Other                            all
        305   Use Proxy                            all

__________________________________________________________
        400   Bad Request                          all
        401   Unauthorized                         all
        402   Payment Required                     all
        403   Forbidden                            all
        404   Not Found                            all
        405   Method Not Allowed                   all
        406   Not Acceptable                       all
        407   Proxy Authentication Required        all
        408   Request Timeout                      all
        410   Gone                                 all
        411   Length Required                      all
        412   Precondition Failed                  DESCRIBE, SETUP
        413   Request Entity Too Large             all
        414   Request-URI Too Long                 all
        415   Unsupported Media Type               all
        451   Parameter Not Understood             SET_PARAMETER
        452   reserved                             n/a
        453   Not Enough Bandwidth                 SETUP
        454   Session Not Found                    all
        455   Method Not Valid In This State       all
        456   Header Field Not Valid               all
        457   Invalid Range                        PLAY, PAUSE
        458   Parameter Is Read-Only               SET_PARAMETER
        459   Aggregate Operation Not Allowed      all
        460   Only Aggregate Operation Allowed     all
        461   Unsupported Transport                all
        462   Destination Unreachable              all
        470   Connection Authorization Required    all
        471   Connection Credentials not accepted  all
        __________________________________________________________
        500   Internal Server Error                all
        501   Not Implemented                      all
        502   Bad Gateway                          all
        503   Service Unavailable                  all
        504   Gateway Timeout                      all
        505   RTSP Version Not Supported           all


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   Table 4: Status codes and their usage with RTSP methods


                Header                  Defined in Section
                __________________________________________
                Accept-Ranges           Section  14.5
                Connection-Credentials  Section  14.12
                ETag                    Section  14.21
                Location                Section  14.29
                Proxy-Authenticate      Section  14.30
                Public                  Section  14.33
                Range                   Section  14.34
                Retry-After             Section  14.36
                RTP-Info                Section  14.38
                Scale                   Section  14.39
                Session                 Section  14.42
                Server                  Section  14.41
                Speed                   Section  14.40
                Transport               Section  14.45
                Unsupported             Section  14.46
                Vary                    Section  14.48
                WWW-Authenticate        Section  14.50


   Table 5: The RTSP response headers

   calls for RTSP requests to be sent using a secure protocol.

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent
   connections, as the media server otherwise has no reliable way of
   reaching the client.  Also, this is the only way that requests from
   media server to client are likely to traverse firewalls.

9.1 Pipelining

   A client that supports persistent connections or connectionless mode
   MAY "pipeline" its requests (i.e., send multiple requests without
   waiting for each response). A server MUST send its responses to those
   requests in the same order that the requests were received.

9.2 Reliability and Acknowledgements

   The transmission of RTSP over UDP was optionally to implement and
   specified in RFC 2326. However that definition was not sufficient for
   interoperable implementations. Due to lack of interest, this
   specification does not specify how RTSP over UDP is implemented.
   However to maintain backwards compatibility in the message format
   certain RTSP headers are maintained.


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   Any RTSP request according to this specification SHALL NOT be sent to
   a multicast address. Any RTSP request SHALL be acknowledged. If a
   reliable transport protocol is used to carry RTSP, requests MUST NOT
   be retransmitted; the RTSP application MUST instead rely on the
   underlying transport to provide reliability.


        If both the underlying reliable transport such as TCP and
        the RTSP application retransmit requests, it is possible
        that each packet loss results in two retransmissions. The
        receiver cannot typically take advantage of the
        application-layer retransmission since the transport stack
        will not deliver the application-layer retransmission
        before the first attempt has reached the receiver. If the
        packet loss is caused by congestion, multiple
        retransmissions at different layers will exacerbate the
        congestion.

   Each request carries a sequence number in the CSeq header (Section
   14.19), which MUST be incremented by one for each distinct request
   transmitted to the destination end-point.  The initial sequence
   number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0.
   If a request is repeated because of lack of acknowledgement, the
   request MUST carry the original sequence number (i.e., the sequence
   number is not incremented).

9.3 The usage of connections

   Systems implementing RTSP MUST support carrying RTSP over TCP.  The
   default port for the RTSP server is 554 for TCP. A number of RTSP
   packets destined for the same control end-point may be encapsulated
   into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP
   packets, see section  12. Unlike HTTP, an RTSP message MUST contain a
   Content-Length header field whenever that message contains a payload
   (entity).  Otherwise, an RTSP packet is terminated with an empty line
   immediately following the last message header.

   TCP can be used for both persistent connections and for one message
   exchange per connection, as presented above. This section gives
   further rules and recommendations on how to handle these connections
   so maximum interoperability and flexibility can be achieved.

   A server SHALL handle both persistent connections and one
   request/response transaction per connection. A persistent connection
   MAY be used for all transactions between the server and client,
   including messages to multiple RTSP sessions. However the persistent
   connection MAY also be closed after a few message exchanges, e.g. the
   initial setup and play command in a session. Later when the client



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   wishes to send a new request, e.g.  pause, to the session a new
   connection is opened. This connection may either be for a single
   message exchange or can be kept open for several messages, i.e.
   persistent.

   A major motivation for allowing non-persistent connections are that
   they ensure fault tolerance. A second one is to allow for application
   layer mobility. A server and client supporting non-persistent
   connection can survive a loss of a TCP connection, e.g. due to a NAT
   timeout. When the client has discovered that the TCP connection has
   been lost, it can set up a new one when there is need to communicate.

   The client MAY close the connection at any time when no outstanding
   request/response transactions exist. The server SHOULD NOT close the
   connection unless at least one RTSP session timeout period has passed
   without data traffic. A server SHOULD NOT initiate a close of a
   connection directly after responding to a TEARDOWN request for the
   whole session. A server SHOULD NOT close the connection as a result
   of responding to a request with an error code. Doing this would
   prevent or result in extra overhead for the client when testing
   advanced or special types of requests.

   The client SHOULD NOT have more than one connection to the server at
   any given point. If a client or proxy handles multiple RTSP sessions
   on the same server, it is RECOMMENDED to use only a single
   connection.

   A Client is strongly RECOMMENDED to use persistent connections as it
   allows the server to send request to the client. In cases where no
   connection exist between the server and the client, this may cause
   the server to be forced to drop the RTSP session without notifying
   the client why, due to the lack of signalling channel.  An example of
   such a case is when the server desires to send a REDIRECT request for
   an RTSP session to the client.

   A server implemented according to this specification MUST respond
   that it supports the "play.basic" feature-tag above. A client MAY
   send a request including the Supported header in a request to
   determine support of non-persistent connections. A server supporting
   non-persistent connections will return the "play.basic" feature-tag
   in its response. If the client receives the feature-tag in the
   response, it can be certain that the server handles non-persistent
   connections.

9.4 Timing Out RTSP messages

   Receivers of a request (responder) SHOULD respond to requests in a
   timely manner even when a reliable transport such as TCP is used.



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   Similarly, the sender of a request (requestor) SHOULD wait for a
   sufficient time for a response before concluding that the responder
   will not be acting upon its request.

   A responder SHOULD respond to all requests within 5 seconds. If the
   responder recognizes that processing of a request will take longer
   than 5 seconds, it SHOULD send a 100 response as soon as possible. It
   SHOULD continue sending a 100 response every 5 seconds thereafter
   until it is ready to send the final response to the requestor. After
   sending a 100 response, the receiver MUST send a final response
   indicating the success or failure of the request.

   A requestor SHOULD wait at least 10 seconds for a response before
   concluding that the responder will not be responding to its request.
   After receiving a 100 response, the requestor SHOULD continue waiting
   for further responses. If more than 10 seconds elapses without
   receiving any response, the requestor MAY assume the responder is
   unresponsive and abort the connection.

   A requestor SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder. The requestor is capable of determining the RTT using the
   Timestamp header (section 14.44) in any RTSP request.

9.5 Use of IPv6

   This specification has been updated so that it supports IPv6.
   However this support was not present in RFC 2326 therefore some
   interoperability issues exist. A RFC 2326 implementation can support
   IPv6 as long as no explicit IPv6 addresses are used within RTSP
   messages. This require that any RTSP URI pointing at a IPv6 host
   needs to use fully qualified domain name and not an IPv6 address.
   Further the Transport header can not use the parameters source and
   destination.

   Implementations according to this specification MUST understand IPv6
   addresses in URIs, and headers. By this requirement the feature-tag
   "play.basic" can be used to determine that a server or client is
   capable of handling IPv6 within RTSP.

10 Capability Handling

   This section describes the capability handling mechanism available in
   RTSP which allows RTSP to be extended. Extensions to this version of
   the protocol are basically done in two ways. First, new headers can
   be added. Secondly, new methods can be added. The capability handling
   mechanism is designed to handle both cases.




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   When a method is added, the involved parties can use the OPTIONS
   method to discover wether it is supported. This is done by issuing a
   OPTIONS request to the other party. Depending on the URI it will
   either apply in regards to a certain media resource, the whole server
   in general, or simply the next hop. The OPTIONS response will contain
   a Public header which declares all methods supported for the
   indicated resource.

   It is not necessary to use OPTIONS to discover support of a method,
   the client could simply try the method. If the receiver of the
   request does not support the method it will respond with an error
   code indicating the the method is either not implemented (501) or
   does not apply for the resource (405). The choice between the two
   discovery methods depends on the requirements of the service.

   Feature-Tags are defined to handle functionality additions that are
   not new methods. Each feature-tag represents a certain block of
   functionality. The amount of functionality that a feature-tag
   represents can vary significantly. A feature-tag can for example
   represent the functionality a single RTSP header provides. Another
   feature-tag can represent much more functionality, such as the
   "play.basic" feature tag which represents the minimal playback
   implementation according to the updated specification.

   Feature-tags are used to determine wether the client, server or proxy
   supports the functionality that is necessary to achieve the desired
   service. To determine support of a feature-tag, several different
   headers can be used, each explained below:


        Supported: The supported header is used to determine the
             complete set of functionality that both client and server
             have. The intended usage is to determine before one needs
             to use a functionality that it is supported. It can be used
             in any method, however OPTIONS is the most suitable one as
             it at the same time determines all methods that are
             implemented. When sending a request the requestor declares
             all its capabilities by including all supported feature-
             tags. This results in that the receiver learns the
             requestors feature support. The receiver then includes its
             set of features in the response.

        Proxy-Supported: The Proxy-Supported header is used similar to
             the Supported header, but instead of giving the supported
             functionality of the client or server it provides both the
             requestor and the responder a view of what functionality
             the proxy chain between the two supports.  Proxies are
             required to add this header whenever the Supported header



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             is present, but proxies may independently of the requestor
             add it.

        Require: The Require header can be included in any request where
             the end-point, i.e. the client or server, is required to
             understand the feature to correctly perform the request.
             This can, for example, be a SETUP request where the server
             is required to understand a certain parameter to be able to
             set up the media delivery correctly. Ignoring this
             parameter would not have the desired effect and is not
             acceptable. Therefore the end-point receiving a request
             containing a Require MUST negatively acknowledge any
             feature that it does not understand and not perform the
             request. The response in cases where features are not
             supported are 551 (Option Not Supported).  Also the
             features that are not supported are given in the
             Unsupported header in the response.

        Proxy-Require: This method has the same purpose and workings as
             Require except that it only applies to proxies and not the
             end-point. Features that needs to be supported by both
             proxies and end-point needs to be included in both the
             Require and Proxy-Require header.

        Unsupported: This header is used in a 551 error response, to
             indicate which feature(s) that was not supported.  Such a
             response is only the result of the usage of the Require
             and/or Proxy-Require header where one or more feature where
             not supported. This information allows the requestor to
             make the best of situations as it knows which features are
             not supported.

11 Method Definitions

   The method indicates what is to be performed on the resource
   identified by the Request-URI. The method name is case-sensitive.
   New methods may be defined in the future. Method names SHALL NOT
   start with a $ character (decimal 24) and MUST be a token as defined
   by the ABNF [5]. Methods are summarized in Table 7.



        Note on Table 7: PAUSE is recommended, but not required.
        For example, a fully functional server can be built to
        deliver live feeds, which do not support this method.

   If an RTSP agent does not support a particular method, it MUST return
   501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD



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    method         direction       object  Server req.    Client req.
    ___________________________________________________________________
    DESCRIBE       C -> S          P,S     recommended    recommended
    GET_PARAMETER  C -> S, S -> C  P,S     optional       optional
    OPTIONS        C -> S, S -> C  P,S     R=Req, Sd=Opt  Sd=Req, R=Opt
    PAUSE          C -> S          P,S     recommended    recommended
    PING           C -> S, S -> C  P,S     recommended    optional
    PLAY           C -> S          P,S     required       required
    REDIRECT       S -> C          P,S     optional       optional
    SETUP          C -> S          S       required       required
    SET_PARAMETER  C -> S, S -> C  P,S     optional       optional
    TEARDOWN       C -> S          P,S     required       required


   Table 7: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation,  S:  stream)  they operate on. Legend:  R=Respond,
   Sd=Send, Opt: Optional, Req: Required, Rec:  Recommended

   NOT try this method again for the given agent / resource combination.

11.1 OPTIONS

   The semantics of the RTSP OPTIONS method is equivalent to that of the
   HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is
   bi-directional, in that a client can request it to a server and vice
   versa. A client MUST implement the capability to send an OPTIONS
   request and a server or a proxy MUST implement the capability to
   respond to an OPTIONS request. The client, server or proxy MAY also
   implement the converse of their required capability.

   An OPTIONS request may be issued at any time. Such a request does not
   modify the session state. However, it may prolong the session
   lifespan (see below). The URI in an OPTIONS request determines the
   scope of the request and the corresponding response. If the Request-
   URI refers to a specific media resource on a given host, the scope is
   limited to the set of methods supported for that media resource by
   the indicated RTSP agent. A Request-URI with only the host address
   limits the scope to the specified RTSP agent's general capabilities
   without regard to any specific media. If the Request-URI is an
   asterisk ("*"), the scope is limited to the general capabilities of
   the next hop (i.e.  the RTSP agent in direct communication with the
   request sender).

   Regardless of scope of the request, the Public header MUST always be
   included in the OPTIONS response listing the methods that are
   supported by the responding RTSP agent. In addition, if the scope of
   the request is limited to a media resource, the Allow header MAY be
   included in the response to enumerate the set of methods that are


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   allowed for that resource. If the given resource is not available,
   the RTSP agent SHOULD return an appropriate response code such as 3rr
   or 4xx.  The Supported header can be included in the request to query
   the set of features that are supported by the responding RTSP agent.

   The OPTIONS method can be used to keep an RTSP session alive.
   However, it is not the preferred means of session keep-alive
   signalling, see section  14.42. An OPTIONS request intended for
   keeping alive an RTSP session MUST include the Session header with
   the associated session ID. Such a request SHOULD also use the media
   or the aggregated control URI as the Request-URI.

   Example:


     C->S:  OPTIONS * RTSP/1.0
            CSeq: 1
            User-Agent: PhonyClient/1.2
            Require:
            Proxy-Require: gzipped-messages
            Supported: play.basic

     S->C:  RTSP/1.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
            Supported: play.basic, implicit-play, gzipped-messages
            Server: PhonyServer/1.0



   Note that some of the feature-tags in Require and Proxy-Require are
   necessarily fictional features (one would hope that we would not
   purposefully overlook a truly useful feature just so that we could
   have a strong example in this section).

11.2 DESCRIBE

   The DESCRIBE method is used to retrieve the description of a
   presentation or media object from a server. The Request-URI of the
   DESCRIBE request identifies the media resource of interest.  The
   client MAY include the Accept header in the request to list the
   description formats that it understands. The server SHALL respond
   with a description of the requested resource and return the
   description in the entity of the response. The DESCRIBE reply-
   response pair constitutes the media initialization phase of RTSP.

   Example:




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     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
           CSeq: 312
           User-Agent: PhonyClient 1.2
           Accept: application/sdp, application/rtsl, application/mheg

     S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Content-Type: application/sdp
           Content-Length: 376

           v=0
           o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
           e=mjh@isi.edu (Mark Handley)
           c=IN IP4 224.2.17.12/127
           t=2873397496 2873404696
           a=recvonly
           m=audio 3456 RTP/AVP 0
           m=video 2232 RTP/AVP 31
           m=application 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization
   information for the resource(s) that it describes. Servers SHOULD NOT
   use the DESCRIBE response as a means of media indirection.


        By forcing a DESCRIBE response to contain all media
        initialization for the set of streams that it describes,
        and discouraging the use of DESCRIBE for media indirection,
        any looping problems can be avoided that might have
        resulted from other approaches.

   Media initialization is a requirement for any RTSP-based system, but
   the RTSP specification does not dictate that this is required to be
   done via the DESCRIBE method. There are three ways that an RTSP
   client may receive initialization information:

        o via an RTSP DESCRIBE method

        o via some other protocol (HTTP, email attachment, etc.)




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        o via some form of a user interface

   If a client obtains a valid description from an alternate source, the
   client MAY use this description for initialization purposes without
   issuing a DESCRIBE request for the same media.

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as "helper applications" that accept a media initialization file
   from a user interface, and/or other means that are appropriate to the
   operating environment of the clients.

11.3 SETUP

   The SETUP request for an URI specifies the transport mechanism to be
   used for the streamed media. The SETUP method may be used in three
   different cases; Create an RTSP session, add a media to a session,
   and change the transport parameters of already set up media stream.
   When in PLAY state, using SETUP to create or add media to a session
   when in PLAY state is unspecified. Otherwise SETUP can be used in all
   three states; INIT, and READY, for both purposes and in PLAY to
   change the transport parameters.

   The Transport header, see section  14.45, specifies the transport
   parameters acceptable to the client for data transmission; the
   response will contain the transport parameters selected by the
   server. This allows the client to enumerate in priority order the
   transport mechanisms and parameters acceptable to it, while the
   server can select the most appropriate. It is expected that the
   session description format used will enable the client to select a
   limited number possible configurations that are offered to the server
   to choose from. All transport parameters SHOULD be included in the
   Transport header, the use of other headers for this purpose is
   discouraged due to middle boxes such as firewalls, or NATs.

   For the benefit of any intervening firewalls, a client SHOULD
   indicate the transport parameters even if it has no influence over
   these parameters, for example, where the server advertises a fixed
   multicast address.


        Since SETUP includes all transport initialization
        information, firewalls and other intermediate network
        devices (which need this information) are spared the more
        arduous task of parsing the DESCRIBE response, which has
        been reserved for media initialization.

   In a SETUP response the server SHOULD include the Accept-Ranges



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   header (see section 14.5 to indicate which time formats that are
   acceptable to use for this media resource.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;unicast;client_port=4588-4589,
                      RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Session: 47112344;timeout=60
           Transport: RTP/AVP;unicast;client_port=4588-4589;
                      server_port=6256-6257;ssrc=2A3F93ED
           Accept-Ranges: NPT



   In the above example the client wants to create an RTSP session
   containing the media resource "rtsp://example.com/foo/bar/baz.rm".
   The transport parameters acceptable to the client is either
   RTP/AVP/UDP (UDP per default) to be received on client port 4588 and
   4589 or RTP/AVP interleaved on the RTSP control channel. The server
   selects the RTP/AVP/UDP transport and adds the ports it will send and
   received RTP and RTCP from, and the RTP SSRC that will be used by the
   server.

   The server MUST generate a session identifier in response to a
   successful SETUP request, unless a SETUP request to a server includes
   a session identifier, in which case the server MUST bundle this setup
   request into the existing session (aggregated session) or return
   error 459 (Aggregate Operation Not Allowed) (see Section  13.4.11).
   An Aggregate control URI MUST be used to control an aggregated
   session. This URI MUST be different from the stream control URIs of
   the individual media streams included in the aggregate. The Aggregate
   control URI is to be specified by the session description if the
   server supports aggregated control and aggregated control is desired
   for the session. However even if aggregated control is offered the
   client MAY chose to not set up the session in aggregated control. If
   an Aggregate control URI is not specified in the session description,
   it is normally an indication that non-aggregated control should be
   used. The SETUP of media streams in an aggregate which has not been
   given an aggregated control URI is unspecified.


        While the session ID sometimes has enough information for



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        aggregate control of a session, the Aggregate control URI
        is still important for some methods such as SET_PARAMETER
        where the control URI enables the resource in question to
        be easily identified. The Aggregate control URI is also
        useful for proxies, enabling them to route the request to
        the appropriate server, and for logging, where it is useful
        to note the actual resource that a request was operating
        on. Finally, presence of the Aggregate control URI allows
        for backwards compatibility with RFC 2326 [1].

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server. The server MAY remove a session that
   has not demonstrated liveness signs from the client(s) within a
   certain timeout period. The default timeout value is 60 seconds; the
   server MAY set this to a different value and indicate so in the
   timeout field of the Session header in the SETUP response. For
   further discussion see section  14.42. Signs of liveness for an RTSP
   session are:

        o Any RTSP request from a client(s) which includes a Session
          header with that session's ID.

        o If RTP is used as a transport for the underlying media
          streams, an RTCP sender or receiver report from the client(s)
          for any of the media streams in that RTSP session. RTCP Sender
          Reports may for example be received in sessions where the
          server is invited into a conference session and is as valid
          for keep-alive.


   If a SETUP request on a session fails for any reason, the session
   state, as well as transport and other parameters for associated
   streams SHALL remain unchanged from their values as if the SETUP
   request had never been received by the server.

   A client MAY issue a SETUP request for a stream that is already set
   up or playing in the session to change transport parameters, which a
   server MAY allow. If it does not allow this, it MUST respond with
   error 455 (Method Not Valid In This State). Reasons to support
   changing transport parameters, is to allow for application layer
   mobility and flexibility to utilize the best available transport as
   it becomes available.

   In a SETUP response for a request to change the transport parameters
   while in Play state, the server SHOULD include the Range to indicate
   from what point the new transport parameters are used. Further, if
   RTP is used for delivery, the server SHOULD also include the RTP-Info
   header to indicate from what timestamp and RTP sequence number the



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   change has taken place. If both RTP-Info and Range is included in the
   response the "rtp_time" parameter and range MUST be for the
   corresponding time, i.e. be used in the same way as for PLAY to
   ensure the correct synchronization information is available.

   If the transport parameters change while in PLAY state results in a
   change of synchronization related information, for example changing
   RTP SSRC, the server MUST provide in the SETUP response the necessary
   synchronization information. However the server is RECOMMENDED to
   avoid changing the synchronization information if possible.

11.4 PLAY

   The PLAY method tells the server to start sending data via the
   mechanism specified in SETUP. A client MUST NOT issue a PLAY request
   until any outstanding SETUP requests have been acknowledged as
   successful. PLAY requests are valid when the session is in READY
   state; the use of PLAY requests when the session is in PLAY state is
   deprecated. A PLAY request MUST include a Session header to indicate
   which session the request applies to.

   In an aggregated session the PLAY request MUST contain an aggregated
   control URI. A server SHALL responde with error 460 (Only Aggregate
   Operation Allowed) if the client PLAY Request-URI is for one of the
   media. The media in an aggregate SHALL be played in sync. If a client
   want individual control of the media it needs to use separate RTSP
   sessions for each media.

   The PLAY request SHALL position the normal play time to the beginning
   of the range specified by the Range header and delivers stream data
   until the end of the range if given, else to the end of the media is
   reached. To allow for precise composition multiple ranges MAY be
   specified in one PLAY Request. The range values are valid if all
   given ranges are part of any media within the aggregate. If a given
   range value points outside of the media, the response SHALL be the
   457 (Invalid Range) error code.

   The below example will first play seconds 10 through 15, then,
   immediately following, seconds 20 to 25, and finally seconds 30
   through the end.


     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=10-15, npt=20-25, npt=30-





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   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal. It SHALL start
   playing a stream from the beginning (npt=0-) unless the stream has
   been paused. If a stream has been paused via PAUSE, stream delivery
   resumes at the pause point. The stream SHALL play until the end of
   the media.

   The Range header MUST NOT contain a time parameter. The usage of time
   in PLAY method has been deprecated. If a request with time parameter
   is received the server SHOULD respond with a 457 (Invalid Range) to
   indicate that the time parameter is not supported.

   Server MUST include a "Range" header in any PLAY response. The
   response MUST use the same format as the request's range header
   contained. If no Range header was in the request, the NPT time format
   SHOULD be used unless the client showed support for an other format
   more appropriate. Also for a session with live media streams the
   Range header MUST indicate a valid time. It is RECOMMENDED that
   normal play time is used, either the "now" indicator, for example
   "npt=now-", or the time since session start as an open interval, e.g.
   "npt=96.23-". An absolute time value (clock) for the corresponding
   time MAY be given, i.e.  "clock=20030213T143205Z-". The UTC clock
   format SHOULD only be used if client has shown support for it.

   A media server only supporting playback MUST support the npt format
   and MAY support the clock and smpte formats.

   For an on-demand stream, the server MUST reply with the actual range
   that will be played back, i.e. for which duration any media (having
   content at this time) is delivered. This may differ from the
   requested range if alignment of the requested range to valid frame
   boundaries is required for the media source. Note that some media
   streams in an aggregate may need to be delivered from even earlier
   points. Also, some media format have a very long duration per
   individual data unit, therefore it might be necessary for the client
   to parse the data unit, and select where to start.


   Example: Single audio stream (MIDI)

   C->S: PLAY rtsp://example.com/audio RTSP/1.0
         CSeq: 836
         Session: 12345678
         Range: npt=7.05-

   S->C: RTSP/1.0 200 OK
         CSeq: 836



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         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: npt=3.52-
         RTP-Info:url=rtsp://example.com/audio;
            seq=14783;rtptime=2345962545

   S->C: RTP Packet TS=2345962545 => NPT=3.52
         Duration: 4.15 seconds



   In this example the client receives the first media packet that
   stretches all the way up and past the requested playtime. Thus, it is
   the client's decision if to render to the user the time between 3.52
   and 7.05, or to skip it. In most cases it is probably most suitable
   to not render that time period.

   For live media sources it might be impossible to specify from which
   point in time all media streams carrying active content can actually
   be delivered. Therefore a server MAY specify a start time (or now-)
   in the range header, for which not all media will be available from.

   If no range is specified in the request, the start position SHALL
   still be returned in the reply. If the medias that are part of an
   aggregate has different lengths, the PLAY request SHALL be performed
   as long as the given range is valid for any media, for example the
   longest media. Media will be sent whenever it is available for the
   given play-out point.

   A PLAY response MAY include a header(s) carrying synchronization
   information. As the information necessary is dependent on the media
   transport format, further rules specifying the header and its usage
   is needed. For RTP the RTP-Info header is specified, see section
   14.38.

   After playing the desired range, the presentation is NOT
   automatically paused, media delivery simply stops. A PAUSE request
   MUST be issued before another PLAY request can be issued.


        Note: The above is a change resulting in a non-operability
        with RFC 2326 implementations. See Appendix F.1

   A client desiring to play the media from the beginning MUST send a
   PLAY request with a Range header pointing at the beginning, e.g.
   npt=0-. If a PLAY request is received without a Range header when
   media delivery has stopped at the end, the server SHOULD respond with
   a 457 "Invalid Range" error response. In that response the current



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   pause point in a Range header SHALL be included.

   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip. Note: The RTP-Info
   headers has been broken into several lines to fit the page.


   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-

   S->C: RTSP/1.0 200 OK
         CSeq: 833
         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: smpte=0:10:22-0:15:45
         RTP-Info:url=rtsp://example.com/twister.en;
            seq=14783;rtptime=2345962545



   For playing back a recording of a live presentation, it may be
   desirable to use clock units:


     C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: clock=19961108T142300Z-19961108T143520Z

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:06 GMT
           Server:PhonyServer 1.0
           Range: clock=19961108T142300Z-19961108T143520Z
           RTP-Info:url=rtsp://example.com/meeting.en;
              seq=53745;rtptime=484589019




   All range specifiers in this specification allow for ranges with
   unspecified begin times (e.g. "npt=-30"). When used in a PLAY
   request, the server treats this as a request to start/resume playback
   from the current pause point, ending at the end time specified in the
   Range header. If the pause point is located later than the given end
   value, a 457 (Invalid Range) response SHALL be given.



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   The queued play functionality described in RFC 2326 [1] is removed
   and multiple ranges can be used to achieve a similar functionality.
   If a server receives a PLAY request while in the PLAY state, the
   server SHALL respond using the error code 455 (Method Not Valid In
   This State). This will signal the client that queued play are not
   supported.

   The use of PLAY for keep-alive signaling, i.e. PLAY request without a
   range header in PLAY state, has also been deprecated. Instead a
   client can use, PING, SET_PARAMETER or OPTIONS for keep alive. A
   server receiving a PLAY keep alive SHALL respond with the 455 error
   code.

11.5 PAUSE

   The PAUSE request causes the stream delivery to be interrupted
   (halted) temporarily. A PAUSE request MUST be done with the
   aggregated control URI for aggregated sessions, resulting in all
   media being halted, or the media URI for non-aggregated sessions.
   Any attempt to do muting of a single media with an PAUSE request in
   an aggregated session SHALL be responded with error 460 (Only
   Aggregate Operation Allowed). After resuming playback,
   synchronization of the tracks MUST be maintained. Any server
   resources are kept, though servers MAY close the session and free
   resources after being paused for the duration specified with the
   timeout parameter of the Session header in the SETUP message.

   Example:


     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-



   The PAUSE request MAY contain a Range header specifying when the
   stream or presentation is to be halted. This point is referred to as
   the "pause point". The time parameter in the Range MUST NOT be used.
   The Range header MUST contain a single value, expressed as the
   beginning value an open range. For example, the following clip will
   be played from 10 seconds through 21 seconds of the clip's normal
   play time, under the assumption that the PAUSE request reaches the



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   server within 11 seconds of the PLAY request. Note that some lines
   has been broken in an non-correct way to fit the page:


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-30

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Range: npt=10-30
           RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
                   seq=5712;rtptime=934207921,
                   url=rtsp://example.com/fizzle/videotrack;
                   seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=21-

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=21-
           Session: 12345678



   The pause request becomes effective the first time the server is
   encountering the time point specified in any of the multiple ranges.
   If the Range header specifies a time outside any range from the PLAY
   request, the error 457 (Invalid Range) SHALL be returned. If a media
   unit (such as an audio or video frame) starts presentation at exactly
   the pause point, it is not played. If the Range header is missing,
   stream delivery is interrupted immediately on receipt of the message
   and the pause point is set to the current normal play time. However,
   the pause point in the media stream MUST be maintained. A subsequent
   PLAY request without Range header SHALL resume from the pause point
   and play until media end.

   If the server has already sent data beyond the time specified in the
   PAUSE request's Range header, a PLAY without range SHALL resume at



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   the point in time specified by the PAUSE request's Range header, as
   it is assumed that the client has discarded data after that point.
   This ensures continuous pause/play cycling without gaps.

   The pause point after any PAUSE request SHALL be returned to the
   client by adding a Range header with what remains unplayed of the
   PLAY request's ranges, i.e. including all the remaining ranges part
   of multiple range specification. If one desires to resume playing a
   ranged request, one simply includes the Range header from the PAUSE
   response. Note that this server behavior was not mandated previously
   and servers implementing according to RFC 2326 will probably not
   return the range header.

   For example, if the server have a play request for ranges 10 to 15
   and 20 to 29 pending and then receives a pause request for NPT 21, it
   would start playing the second range and stop at NPT 21. If the pause
   request is for NPT 12 and the server is playing at NPT 13 serving the
   first play request, the server stops immediately. If the pause
   request is for NPT 16, the server returns a 457 error message. To
   prevent that the second range is played and the server stops after
   completing the first range, a PAUSE request for NPT 20 needs to be
   issued.

   As another example, if a server has received requests to play ranges
   10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
   request for NPT=14 would take effect while the server plays the first
   range, with the second range effectively being ignored, assuming the
   PAUSE request arrives before the server has started playing the
   second, overlapping range. Regardless of when the PAUSE request
   arrives, it sets the pause point to 14. The below example messages is
   for the above case when the PAUSE request arrives before the first
   occurrence of NPT=14.


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-15, npt=13-20

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Range: npt=10-15, npt=13-20
           RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
                   seq=5712;rtptime=934207921,
                   url=rtsp://example.com/fizzle/videotrack;
                   seq=57654;rtptime=2792482193



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           Session: 12345678

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=14-

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=14-15, npt=13-20
           Session: 12345678



   If a client issues a PAUSE request and the server acknowledges and
   enters the READY state, the proper server response, if the player
   issues another PAUSE, is still 200 OK. The 200 OK response MUST
   include the Range header with the current pause point, even if the
   PAUSE request is asking for some other pause point. See examples
   below:

   Examples:

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Session: 12345678
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-98.36

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: 86-

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Session: 12345678
           Date: 23 Jan 1997 15:35:07 GMT
           Range: npt=45.76-98.36






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11.6 TEARDOWN

   The TEARDOWN client to server request stops the stream delivery for
   the given URI, freeing the resources associated with it. A TEARDOWN
   request MAY be performed on either an aggregated or a media control
   URI. However some restrictions apply depending on the current state.
   The TEARDOWN request SHALL contain a Session header indicating what
   session the request applies to.

   A TEARDOWN using the aggregated control URI or the media URI in a
   session under non-aggregated control MAY be done in any state (Ready,
   and Play). A successful request SHALL result in that media delivery
   is immediately halted and the session state is destroyed. This SHALL
   be indicated through the lack of a Session header in the response.

   A TEARDOWN using a media URI in an aggregated session MAY only be
   done in Ready state. Such a request only removes the indicated media
   stream and associated resources from the session.  This may result in
   that a session returns to non-aggregated control, due to that it only
   contains a single media after the requests completion. A session that
   will exist after the processing of the TEARDOWN request SHALL in the
   response to that TEARDOWN request contain a Session header. Thus the
   presence of the Session indicates to the receiver of the response if
   the session is still existing or has been removed.

   Note, the indication with the session header if sessions state remain
   may not be done correctly by a RFC 2326 client, but will be for any
   server signalling the "play.basic" tag.

   Example:


     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 892
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 892
           Server: PhonyServer 1.0



11.7 GET_PARAMETER

   The GET_PARAMETER request retrieves the value of a parameter or
   parameters for a presentation or stream specified in the URI. If the
   Session header is present in a request, the value of a parameter MUST
   be retrieved in the specified session context.  The content of the



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   reply and response is left to the implementation.

   The method MAY also be used without a body (entity). If the this
   request is successful, i.e. a 200 OK response is received, then the
   keep-alive timer has been updated. Any non-required header present in
   such a request may or may not been processed. To allow a client to
   determine if any such header has been processed, it is necessary to
   use a feature tag and the Require header. Due to this reason it is
   RECOMMENDED that any parameters to be retrieved are sent in the body,
   rather than using any header.

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 431
           Content-Type: text/parameters
           Session: 12345678
           Content-Length: 15

           packets_received
           jitter

     C->S: RTSP/1.0 200 OK
           CSeq: 431
           Content-Length: 46
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838




        The "text/parameters" section is only an example type for
        parameter body.

11.8 SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URI. The
   method MAY also be used without a body (entity). If this request is
   successful, i.e. a 200 OK response is received, then the keep-alive
   timer has been updated. Any non-required header present in such a
   request may or may not been processed. To allow a client to determine
   if any such header has been processed, it is necessary to use a
   feature tag and the Require header.  Due to this reason it is
   RECOMMENDED that any parameters are sent in the body, rather than



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   using any header.

   A request is RECOMMENDED to only contain a single parameter to allow
   the client to determine why a particular request failed. If the
   request contains several parameters, the server MUST only act on the
   request if all of the parameters can be set successfully. A server
   MUST allow a parameter to be set repeatedly to the same value, but it
   MAY disallow changing parameter values.  If the receiver of the
   request does not understand or cannot locate a parameter, error 451
   (Parameter Not Understood) SHALL be used. In the case a parameter is
   not allowed to change, the error code is 458 (Parameter Is Read-
   Only). The response body SHOULD contain only the parameters that have
   errors. Otherwise no body SHALL be returned.

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

        Restricting setting transport parameters to SETUP is for
        the benefit of firewalls.


        The parameters are split in a fine-grained fashion so that
        there can be more meaningful error indications. However, it
        may make sense to allow the setting of several parameters
        if an atomic setting is desirable. Imagine device control
        where the client does not want the camera to pan unless it
        can also tilt to the right angle at the same time.

   Example:


     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 421
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/1.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10
           Content-type: text/parameters

           barparam







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        The "text/parameters" section is only an example type for
        parameter. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined after further experimentation.

11.9 REDIRECT

   The REDIRECT method is issued by a server to inform a client that it
   required to connect to another server location to access the resource
   indicated by the Request-URI. The presence of the Session header in a
   REDIRECT request indicates the scope of the request, and determines
   the specific semantics of the request.

   A REDIRECT request with a Session header has end-to-end (i.e. server
   to client) scope and applies only to the given session. Any
   intervening proxies SHOULD NOT disconnect the control channel while
   there are other remaining end-to-end sessions. The OPTIONAL Location
   header, if included in such a request, SHALL contain a complete
   absolute URI pointing to the resource to which the client SHOULD
   reconnect.  Specifically, the Location SHALL NOT contain just the
   host and port. A client may receive a REDIRECT request with a Session
   header, if and only if, an end-to-end session has been established.

   A client may receive a REDIRECT request without a Session header at
   any time when it has communication or a connection established with a
   server. The scope of such a request is limited to the next-hop (i.e.
   the RTSP agent in direct communication with the server) and applies,
   as well, to the control connection between the next-hop RTSP agent
   and the server.  A REDIRECT request without a Session header
   indicates that all sessions and pending requests being managed via
   the control connection MUST be redirected. The OPTIONAL Location
   header, if included in such a request, SHOULD contain an absolute URI
   with only the host address and the OPTIONAL port number of the server
   to which the RTSP agent SHOULD reconnect. Any intervening proxies
   SHOULD do all of the following in the order listed:

        1.   respond to the REDIRECT request

        2.   disconnect the control channel from the requesting server

        3.   connect to the server at the given host address

        4.   pass the REDIRECT request to each applicable client
             (typically those clients with an active session or an
             unanswered request)

   Note: The proxy is responsible for accepting REDIRECT responses from
   its clients; these responses MUST NOT be passed on to either the



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   original server or the redirected server.

   The lack of a Location header in any REDIRECT request is indicative
   of the server no longer being able to fulfill the current request and
   having no alternatives for the client to continue with its normal
   operation. It is akin to a server initiated TEARDOWN that applies
   both to sessions as well as the general connection associated with
   that client.

   When the Range header is not included in a REDIRECT request, the
   client SHOULD perform the redirection immediately and return a
   response to the server. The server can consider the session as
   terminated and can free any associated state after it receives the
   successful (2xx) response. The server MAY close the signalling
   connection upon receiving the response and the client SHOULD close
   the signalling connection after sending the 2xx response. The
   exception to this is when the client has several sessions on the
   server being managed by the given signalling connection. In this
   case, the client SHOULD close the connection when it has received and
   responded to REDIRECT requests for all the sessions managed by the
   signalling connection.

   If the OPTIONAL Range header is included in a REDIRECT request, it
   indicates when the redirection takes effect. The range value MUST be
   an open ended single value, e.g. npt=59-, indicating the play out
   time when redirection SHALL occur. Alternatively, a range with a
   time= parameter indicates the wall clock time by when the redirection
   MUST take place. When the time= parameter is present in the range,
   any range value MUST be ignored even though it MUST be syntactically
   correct. When the indicated redirect point is reached, a client MUST
   issue a TEARDOWN request and SHOULD close the signalling connection
   after receiving a 2xx response. The normal connection considerations
   apply for the server.


        The differentiation of REDIRECT requests with and without
        range headers is to allow for clear and explicit state
        handling. As the state in the server needs to be kept until
        the point of redirection, the handling becomes more clear
        if the client is required to TEARDOWN the session at the
        redirect point.

   After a REDIRECT request has been processed, a client that wants to
   continue to send or receive media for the resource identified by the
   Request-URI will have to establish a new session with the designated
   host. If the URI given in the Location header is a valid resource
   URI, a client SHOULD issue a DESCRIBE request for the URI.




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        Note: The media resource indicated by the Location header
        can be identical, slightly different or totally different.
        This is the reason why a new DESCRIBE request SHOULD be
        issued.

   If the Location header contains only a host address, the client MAY
   assume that the media on the new server is identical to the media on
   the old server, i.e. all media configuration information from the old
   session is still valid except for the host address.

   This example request redirects traffic for this session to the new
   server at the given absolute time:


     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 732
           Location: rtsp://s2.example.com:8001
           Range: npt=0- ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M



11.10 PING

   This method is a bi-directional mechanism for server or client
   liveness checking. It has no side effects. The issuer of the request
   MUST include a session header with the session ID of the session that
   is being checked for liveness.

   Prior to using this method, an OPTIONS method is RECOMMENDED to be
   issued in the direction which the PING method would be used. This
   method MUST NOT be used if support is not indicated by the Public
   header. Note: That an 501 (Not Implemented) response means that the
   keep-alive timer has not been updated.

   When a proxy is in use, PING with a * indicates a single-hop liveness
   check, whereas PING with an URI including an host address indicates
   an end-to-end liveness check.

   Example:

     C->S: PING * RTSP/1.0
           CSeq: 123
           Session:12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 123
           Session:12345678



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12 Embedded (Interleaved) Binary Data

   In order to fulfill certain requirements on the network side, e.g.
   in conjunction with firewalls that block RTP traffic, it may be
   necessary to interleave RTSP messages and media stream data. This
   interleaving should generally be avoided unless necessary since it
   complicates client and server operation and imposes additional
   overhead. Also head of line blocking may cause problems.  Interleaved
   binary data SHOULD only be used if RTSP is carried over TCP.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order. The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers. Each $ block SHALL contain exactly one upper-layer
   protocol data unit, e.g., one RTP packet.




       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     "$" = 24     Channel ID   Length in bytes
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      : Length number of bytes of binary data                         :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





   The channel identifier is defined in the Transport header with the
   interleaved parameter(Section 14.45).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection. The usage of RTCP messages is
   indicated by including a range containing a second channel in the
   interleaved parameter of the Transport header, see section 14.45. If
   RTCP is used, packets SHALL be sent on the first available channel
   higher than the RTP channel. The channels are bi-directional and
   therefore RTCP traffic are sent on the second channel in both
   directions.


        RTCP is needed for synchronization when two or more streams
        are interleaved in such a fashion. Also, this provides a



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        convenient way to tunnel RTP/RTCP packets through the TCP
        control connection when required by the network
        configuration and transfer them onto UDP when possible.



     C->S: SETUP rtsp://example.com/bar.file RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Date: 05 Jun 1997 18:57:18 GMT
           Transport: RTP/AVP/TCP;unicast;interleaved=5-6
           Session: 12345678

     C->S: PLAY rtsp://example.com/bar.file RTSP/1.0
           CSeq: 3
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: 05 Jun 1997 18:59:15 GMT
           RTP-Info: url=rtsp://example.com/bar.file;
             seq=232433;rtptime=972948234

     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $006{2 byte length}{"length" bytes  RTCP packet}



13 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 4 for a
   listing of which status codes may be returned by which requests. All
   error messages, 4xx and 5xx MAY return a body containing further
   information about the error.

13.1 Success 1xx

13.1.1 100 Continue

   See, [H10.1.1].

13.2 Success 2xx



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13.3 Redirection 3xx

   The notation "3rr" indicates response codes from 300 to 399 inclusive
   which are meant for redirection. The response code 304 is excluded
   from this set, as it is not used for redirection.

   See [H10.3] for definition of status code 300 to 305. However
   comments are given for some to how they apply to RTSP.

   Within RTSP, redirection may be used for load balancing or
   redirecting stream requests to a server topologically closer to the
   client.  Mechanisms to determine topological proximity are beyond the
   scope of this specification.

   A 3rr code MAY be used to respond to any request. It is RECOMMENDED
   that they are used if necessary before a session is established, i.e.
   in response to DESCRIBE or SETUP. However in cases where a server is
   not able to send a REDIRECT request to the client, the server MAY
   need to resort to using 3rr responses to inform a client with a
   established session about the need for redirecting the session. If an
   3rr response is received for an request in relation to a established
   session, the client SHOULD send a TEARDOWN request for the session,
   and MAY reestablish the session using the resource indicated by the
   Location.

   If the the Location header is used in a response it SHALL contain an
   absolute URI pointing out the media resource the client is redirected
   to, the URI SHALL NOT only contain the host name.

13.3.1 300 Multiple Choices

   See [H10.3.1] [TBW]

13.3.2 301 Moved Permanently

   The request resource are moved permanently and resides now at the URI
   given by the location header. The user client SHOULD redirect
   automatically to the given URI. This response MUST NOT contain a
   message-body.

13.3.3 302 Found

   The requested resource reside temporarily at the URI given by the
   Location header. The Location header MUST be included in the
   response. Is intended to be used for many types of temporary
   redirects, e.g. load balancing. It is RECOMMENDED that one set the
   reason phrase to something more meaningful than "Found" in these
   cases. The user client SHOULD redirect automatically to the given



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   URI. This response MUST NOT contain a message-body.

13.3.4 303 See Other

   This status code SHALL NOT be used in RTSP. However as it was allowed
   to use in RFC 2326 it is possible that such response may be received,
   in which case the behavior is undefined.

13.3.5 304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   14.26) and the requested resource has not been modified, the server
   SHOULD send a 304 response. This response MUST NOT contain a
   message-body.

   The response MUST include the following header fields:

        o Date

        o ETag and/or Content-Location, if the header would have been
          sent in a 200 response to the same request.

        o Expires, Cache-Control, and/or Vary, if the field-value might
          differ from that sent in any previous response for the same
          variant.

   This response is independent for the DESCRIBE and SETUP requests.
   That is, a 304 response to DESCRIBE does NOT imply that the resource
   content is unchanged and a 304 response to SETUP does NOT imply that
   the resource description is unchanged. The ETag and If-Match headers
   may be used to link the DESCRIBE and SETUP in this manner.

13.3.6 305 Use Proxy

   See [H10.3.6].

13.4 Client Error 4xx

13.4.1 400 Bad Request

   The request could not be understood by the server due to malformed
   syntax. The client SHOULD NOT repeat the request without
   modifications [H10.4.1]. If the request does not have a CSeq header,
   the server MUST NOT include a CSeq in the response.

13.4.2 405 Method Not Allowed

   The method specified in the request is not allowed for the resource



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   identified by the Request-URI. The response MUST include an Allow
   header containing a list of valid methods for the requested resource.
   This status code is also to be used if a request attempts to use a
   method not indicated during SETUP, e.g., if a RECORD request is
   issued even though the mode parameter in the Transport header only
   specified PLAY.

13.4.3 451 Parameter Not Understood

   The recipient of the request does not support one or more parameters
   contained in the request. When returning this error message the
   sender SHOULD return a entity body containing the offending
   parameter(s).

13.4.4 452 reserved

   This error code was removed from RFC 2326 [1] and is obsolete.

13.4.5 453 Not Enough Bandwidth

   The request was refused because there was insufficient bandwidth.
   This may, for example, be the result of a resource reservation
   failure.

13.4.6 454 Session Not Found

   The RTSP session identifier in the Session header is missing,
   invalid, or has timed out.

13.4.7 455 Method Not Valid in This State

   The client or server cannot process this request in its current
   state.  The response SHOULD contain an Allow header to make error
   recovery easier.

13.4.8 456 Header Field Not Valid for Resource

   The server could not act on a required request header. For example,
   if PLAY contains the Range header field but the stream does not allow
   seeking. This error message may also be used for specifying when the
   time format in Range is impossible for the resource. In that case the
   Accept-Ranges header SHOULD be returned to inform the client of which
   format(s) that are allowed.

13.4.9 457 Invalid Range

   The Range value given is out of bounds, e.g., beyond the end of the
   presentation.



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13.4.10 458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not
   modified. When returning this error message the sender SHOULD return
   a entity body containing the offending parameter(s).

13.4.11 459 Aggregate Operation Not Allowed

   The requested method may not be applied on the URI in question since
   it is an aggregate (presentation) URI. The method may be applied on a
   media URI.

13.4.12 460 Only Aggregate Operation Allowed

   The requested method may not be applied on the URI in question since
   it is not an aggregate control (presentation) URI. The method may be
   applied on the aggregate control URI.

13.4.13 461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

13.4.14 462 Destination Unreachable

   The data transmission channel could not be established because the
   client address could not be reached. This error will most likely be
   the result of a client attempt to place an invalid Destination
   parameter in the Transport field.

13.4.15 470 Connection Authorization Required

   The secured connection attempt need user or client authorization
   before proceeding. The next hops certificate is included in this
   response in the Accept-Credentials header.

13.4.16 471 Connection Credentials not accepted

   When performing a secure connection over multiple connections, a
   intermediary has refused to connect to the next hop and carry out the
   request due to unacceptable credentials for the used policy.

13.5 Server Error 5xx

13.5.1 551 Option not supported

   A feature-tag given in the Require or the Proxy-Require fields was
   not supported. The Unsupported header SHOULD be returned stating the



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   feature for which there is no support.

14 Header Field Definitions


             method        direction      object acronym Body
             _________________________________________________
             DESCRIBE      C -> S         P,S    DES     r
             GET_PARAMETER C -> S, S -> C P,S    GPR     R,r
             OPTIONS       C -> S         P,S    OPT
                           S -> C
             PAUSE         C -> S         P,S    PSE
             PING          C -> S, S -> C P,S    PNG
             PLAY          C -> S         P,S    PLY
             REDIRECT      S -> C         P,S    RDR
             SETUP         C -> S         S      STP
             SET_PARAMETER C -> S, S -> C P,S    SPR     R,r
             TEARDOWN      C -> S         P,S    TRD


   Table 8: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation, S: stream) they operate on. Body notes if a method
   is allowed to carry  body  and  in  which  direction,  R  =  Request,
   r=response. Note: It is allowed for all error messages 4xx and 5xx to
   have a body


   The general syntax for header fields is covered in Section 4.2 This
   section lists the full set of header fields along with notes on
   meaning, and usage. The syntax definition for headers are present in
   section 18.2.3. Throughout this section, we use [HX.Y] to refer to
   Section X.Y of the current HTTP/1.1 specification RFC 2616 [4].
   Examples of each header field are given.

   Information about header fields in relation to methods and proxy
   processing is summarized in Tables  9, 10,  11, and 12.

   The "where" column describes the request and response types in which
   the header field can be used. Values in this column are:

        R: header field may only appear in requests;

        r: header field may only appear in responses;

        2xx, 4xx, etc.: A numerical value or range indicates response
             codes with which the header field can be used;

        c: header field is copied from the request to the response.



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   An empty entry in the "where" column indicates that the header field
   may be present in all requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field. An empty proxy column indicates that the proxy SHALL
   NOT do any changes to that header, all allowed operations are
   explicitly stated:

        a: A proxy can add or concatenate the header field if not
             present.

        m: A proxy can modify an existing header field value.

        d: A proxy can delete a header field value.

        r: A proxy needs to be able to read the header field, and thus
             this header field cannot be encrypted.

   The rest of the columns relate to the presence of a header field in a
   method. The method names when abbreviated, are according to table 8:

        c: Conditional; requirements on the header field depend on the
             context of the message.

        m: The header field is mandatory.

        m*: The header field SHOULD be sent, but clients/servers need to
             be prepared to receive messages without that header field.

        o: The header field is optional.

        *: The header field is SHALL be present if the message body is
             not empty. See sections 14.16, 14.18 and 4.3 for details.

        -: The header field is not applicable.

   "Optional" means that a Client/Server MAY include the header field in
   a request or response. The Client/Server behavior when receiving such
   headers varies, for some it may ignore the header field, in other
   case it is request to process the header. This is regulated by the
   method and header descriptions. Example of such headers that require
   processing are the Require and Proxy-Require header fields discussed
   in 14.37 and 14.31. A "mandatory" header field MUST be present in a
   request, and MUST be understood by the Client/Server receiving the
   request. A mandatory response header field MUST be present in the
   response, and the header field MUST be understood by the
   Client/Server processing the response. "Not applicable" means that
   the header field MUST NOT be present in a request. If one is placed



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   in a request by mistake, it MUST be ignored by the Client/Server
   receiving the request. Similarly, a header field labeled "not
   applicable" for a response means that the Client/Server MUST NOT
   place the header field in the response, and the Client/Server MUST
   ignore the header field in the response.

   A Client/Server SHOULD ignore extension header parameters that are
   not understood.

   The From, Location, and RTP-Info header fields contain an URI. If the
   URI contains a comma, or semicolon, the URI MUST be enclosed in
   double quotas ("). Any URI parameters are contained within these
   quotas. If the URI is not enclosed in double quotas, any semicolon-
   delimited parameters are header-parameters, not URI parameters.





14.1 Accept

   The Accept request-header field can be used to specify certain
   presentation description content types which are acceptable for the
   response.

        The "level" parameter for presentation descriptions is
        properly defined as part of the MIME type registration, not
        here.

   See [H14.1] for syntax.

   Example of use:


     Accept: application/rtsl q=1.0, application/sdp



14.2 Accept-Credentials

   The Accept-Credentials header is a request header used to indicate to
   any trusted intermediary how to handle further secured connections to
   proxies or servers. See section 17 for the usage of this header. It
   SHALL only be included in client to server requests.

   In a request the header SHALL contain the method (User, Proxy, or
   Any) for approving credentials selected by the requestor. The method
   SHALL NOT be changed by any proxy. If the method is "User" the header



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   Header                  Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   _________________________________________________________________
   Accept                    R           o   -    -    -     -   -
   Accept-Credentials        R      r    o   o    o    o     o   o
   Accept-Encoding           R      r    o   -    -    -     -   -
   Accept-Language           R      r    o   -    -    -     -   -
   Accept-Ranges             r      r    -   -    o    -     -   -
   Accept-Ranges            456     r    -   -    -    o     o   -
   Allow                     r           -   o    -    -     -   -
   Allow                    405          m   m    m    m     m   m
   Authorization             R           o   o    o    o     o   o
   Bandwidth                 R           o   o    o    o     -   -
   Blocksize                 R           o   -    o    o     -   -
   Cache-Control                    r    -   -    o    -     -   -
   Connection                            o   o    o    o     o   o
   Connection-Credentials 470,407  ar    o   o    o    o     o   o
   Content-Base              r           o   -    -    -     -   -
   Content-Base             4xx          o   o    o    o     o   o
   Content-Encoding          R      r    -   -    -    -     -   -
   Content-Encoding          r      r    o   -    -    -     -   -
   Content-Encoding         4xx     r    o   o    o    o     o   o
   Content-Language          R      r    -   -    -    -     -   -
   Content-Language          r      r    o   -    -    -     -   -
   Content-Language         4xx     r    o   o    o    o     o   o
   Content-Length            r      r    *   -    -    -     -   -
   Content-Length           4xx     r    *   *    *    *     *   *
   Content-Location          r           o   -    -    -     -   -
   Content-Location         4xx          o   o    o    o     o   o
   Content-Type              r           *   -    -    -     -   -
   Content-Type             4xx          *   *    *    *     *   *
   CSeq                     Rc           m   m    m    m     m   m
   Date                            am    o   o    o    o     o   o
   ETag                      r      r    o   -    o    -     -   -
   Expires                   r      r    o   -    -    -     -   -
   From                      R      r    o   o    o    o     o   o
   Host                                  -   -    -    -     -   -
   If-Match                  R      r    -   -    o    -     -   -
   If-Modified-Since         R      r    o   -    o    -     -   -
   If-None-Match             R      r    o   -    -    -     -   -
   Last-Modified             r      r    o   -    -    -     -   -
   Location                 3rr          o   o    o    o     o   o


   Table 9: Overview of RTSP header  fields  (A-L)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.




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   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   _____________________________________________________________
   Proxy-Authenticate   407    amr   m   m    m    m     m    m
   Proxy-Require         R     ar    o   o    o    o     o    o
   Proxy-Supported       R     amr  oc  oc   oc    oc   oc   oc
   Proxy-Supported       r           c   c    c    c     c    c
   Public                r    admr   -  m*    -    -     -    -
   Public               501   admr  m*  m*   m*    m*   m*   m*
   Range                 R           -   -    -    o     o    -
   Range                 r           -   -    c    m*   m*    -
   Referer               R           o   o    o    o     o    o
   Require               R           o   o    o    o     o    o
   Retry-After        3rr,503        o   o    o    -     -    -
   RTP-Info              r           -   -    o    c     -    -
   Scale                             -   -    -    o     -    -
   Session               R           -   o    o    m     m    m
   Session               r           -   c    m    m     m    o
   Server                R           -   o    -    -     -    -
   Server                r           o   o    o    o     o    o
   Speed                             -   -    -    o     -    -
   Supported             R           o   o    o    o     o    o
   Supported             r           c   c    c    c     c    c
   Timestamp             R           o   o    o    o     o    o
   Timestamp             c           m   m    m    m     m    m
   Transport                         -   -    m    -     -    -
   Unsupported           r           c   c    c    c     c    c
   User-Agent            R          m*  m*   m*    m*   m*   m*
   Vary                  r           c   c    c    c     c    c
   Via                   R     amr   o   o    o    o     o    o
   Via                   c     dr    m   m    m    m     m    m
   WWW-Authenticate     401          m   m    m    m     m    m

_____________________________________________________________
   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD



   Table 10: Overview of RTSP header fields  (P-W)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.


   contains zero or more of credentials that the client accept. Each
   credential SHALL consist of one URI identifying the proxy or server,
   and the SHA-1  [15] hash computed over that entity's DER encoded
   certificate [16] in Base64 [36].


   Example:



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   Header                  Where  Proxy GPR SPR RDR PNG
   ______________________________________________________
   Accept-Credentials        R      r    o   o   o   o
   Allow                    405          m   m   m   m
   Authorization             R           o   o   o   o
   Bandwidth                 R           -   o   -   -
   Blocksize                 R           -   o   -   -
   Connection                            o   o   o   -
   Connection-Credentials 470,407  ar    o   o   o   o
   Content-Base              R           o   o   -   -
   Content-Base              r           o   o   -   -
   Content-Base             4xx          o   o   o   -
   Content-Encoding          R      r    o   o   -   -
   Content-Encoding          r      r    o   o   -   -
   Content-Encoding         4xx     r    o   o   o   -
   Content-Language          R      r    o   o   -   -
   Content-Language          r      r    o   o   -   -
   Content-Language         4xx     r    o   o   o   -
   Content-Length            R      r    *   *   -   -
   Content-Length            r      r    *   *   -   -
   Content-Length           4xx     r    *   *   *   -
   Content-Location          R           o   o   -   -
   Content-Location          r           o   o   -   -
   Content-Location         4xx          o   o   o   -
   Content-Type              R           *   *   -   -
   Content-Type              r           *   *   -   -
   Content-Type             4xx          *   *   *   -
   CSeq                     Rc           m   m   m   m
   Date                            am    o   o   o   o
   From                      R      r    o   o   o   o
   Host                                  -   -   -   -
   Last-Modified             R      r    -   -   -   -
   Last-Modified             r      r    o   -   -   -
   Location                 3rr          o   o   o   o
   Location                  R           -   -   m   -
   Proxy-Authenticate       407    amr   m   m   m   m
   Proxy-Require             R     ar    o   o   o   o
   Proxy-Supported           R     amr  oc  oc  oc  oc
   Proxy-Supported           r           c   c   c   c
   Public                   501   admr  m*  m*  m*  m*

______________________________________________________
   Header                  Where  Proxy GPR SPR RDR PNG


   Table 11: Overview of RTSP header fields  (A-P)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, REDIRECT, and PING.




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             Header            Where  Proxy GPR SPR RDR PNG
             ________________________________________________
             Range               R           -   -   o   -
             Referer             R           o   o   o   -
             Require             R           o   o   o   o
             Retry-After      3rr,503        o   o   -   -
             Scale                           -   -   -   -
             Session             R           o   o   o   m
             Session             r           c   c   o   m
             Server              R           o   o   o   o
             Server              r           o   o   -   o
             Supported           R           o   o   o   o
             Supported           r           c   c   c   c
             Timestamp           R           o   o   o   o
             Timestamp           c           m   m   m   m
             Unsupported         r           c   c   c   c
             User-Agent          R          m*  m*   -  m*
             User-Agent          r           -   -  m*   -
             Vary                r           c   c   -   -
             Via                 R     amr   o   o   o   o
             Via                 c     dr    m   m   m   m
             WWW-Authenticate   401          m   m   m   m

________________________________________________
             Header            Where  Proxy GPR SPR RDR PNG


   Table 12: Overview of RTSP header fields  (R-W)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, REDIRECT, and PING.


          "rtsps://server.example.com/";lurbjj5khhB0NhIuOXtt4bBRH1M=




14.3 Accept-Encoding

   See [H14.3]

14.4 Accept-Language

   See [H14.4]. Note that the language specified applies to the
   presentation description and any reason phrases, not the media
   content.

14.5 Accept-Ranges

   The Accept-Ranges response-header field allows the server to indicate



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   its acceptance of range requests and possible formats for a resource:


   Accept-Ranges: NPT, SMPTE



   This header has the same syntax as [H14.5] and the syntax is defined
   in 18.2.3. However, new range-units are defined. Inclusion of any of
   the time formats indicates acceptance by the server for PLAY and
   PAUSE requests with this format. The headers value is valid for the
   resource specified by the URI in the request, this response
   corresponds to. A server SHOULD use this header in SETUP responses to
   indicate to the client which range time formats the media supports.
   The header SHOULD also be included in "456" responses which is a
   result of use of unsupported range formats.

14.6 Allow

   The Allow entity-header field lists the methods supported by the
   resource identified by the Request-URI. The purpose of this field is
   to strictly inform the recipient of valid methods associated with the
   resource. An Allow header field MUST be present in a 405 (Method Not
   Allowed) response. See [H14.7] for syntax definition.

   Example of use:

     Allow: SETUP, PLAY, SET_PARAMETER



14.7 Authorization

   See [H14.8]

14.8 Bandwidth

   The Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second. The bandwidth available to the client may change
   during an RTSP session, e.g., due to mobility.

   Example:

     Bandwidth: 4000






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14.9 Blocksize

   The Blocksize request-header field is sent from the client to the
   media server asking the server for a particular media packet size.
   This packet size does not include lower-layer headers such as IP,
   UDP, or RTP. The server is free to use a blocksize which is lower
   than the one requested. The server MAY truncate this packet size to
   the closest multiple of the minimum, media-specific block size, or
   override it with the media-specific size if necessary. The block size
   MUST be a positive decimal number, measured in octets. The server
   only returns an error (4xx) if the value is syntactically invalid.

14.10 Cache-Control

   The Cache-Control general-header field is used to specify directives
   that MUST be obeyed by all caching mechanisms along the
   request/response chain.

   Cache directives MUST be passed through by a proxy or gateway
   application, regardless of their significance to that application,
   since the directives may be applicable to all recipients along the
   request/response chain. It is not possible to specify a cache-
   directive for a specific cache.

   Cache-Control should only be specified in a SETUP request and its
   response. Note: Cache-Control does not govern the caching of
   responses as for HTTP, instead it applies to the media stream
   identified by the SETUP request.  The caching of RTSP requests are
   generally not cacheable, for further information see 15. Below is the
   description of the cache directives that can be included in the
   Cache-Control header.

        no-cache: Indicates that the media stream MUST NOT be cached
             anywhere. This allows an origin server to prevent caching
             even by caches that have been configured to return stale
             responses to client requests.

        public: Indicates that the media stream is cacheable by any
             cache.

        private: Indicates that the media stream is intended for a
             single user and MUST NOT be cached by a shared cache. A
             private (non-shared) cache may cache the media stream.

        no-transform: An intermediate cache (proxy) may find it useful
             to convert the media type of a certain stream. A proxy
             might, for example, convert between video formats to save
             cache space or to reduce the amount of traffic on a slow



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             link.  Serious operational problems may occur, however,
             when these transformations have been applied to streams
             intended for certain kinds of applications. For example,
             applications for medical imaging, scientific data analysis
             and those using end-to-end authentication all depend on
             receiving a stream that is bit-for-bit identical to the
             original media stream. Therefore, if a response includes
             the no-transform directive, an intermediate cache or proxy
             MUST NOT change the encoding of the stream.  Unlike HTTP,
             RTSP does not provide for partial transformation at this
             point, e.g., allowing translation into a different
             language.

        only-if-cached: In some cases, such as times of extremely poor
             network connectivity, a client may want a cache to return
             only those media streams that it currently has stored, and
             not to receive these from the origin server. To do this,
             the client may include the only-if-cached directive in a
             request. If it receives this directive, a cache SHOULD
             either respond using a cached media stream that is
             consistent with the other constraints of the request, or
             respond with a 504 (Gateway Timeout) status. However, if a
             group of caches is being operated as a unified system with
             good internal connectivity, such a request MAY be forwarded
             within that group of caches.

        max-stale: Indicates that the client is willing to accept a
             media stream that has exceeded its expiration time. If
             max-stale is assigned a value, then the client is willing
             to accept a response that has exceeded its expiration time
             by no more than the specified number of seconds. If no
             value is assigned to max-stale, then the client is willing
             to accept a stale response of any age.

        min-fresh: Indicates that the client is willing to accept a
             media stream whose freshness lifetime is no less than its
             current age plus the specified time in seconds. That is,
             the client wants a response that will still be fresh for at
             least the specified number of seconds.


        must-revalidate: When the must-revalidate directive is present
             in a SETUP response received by a cache, that cache MUST
             NOT use the entry after it becomes stale to respond to a
             subsequent request without first revalidating it with the
             origin server. That is, the cache is required to do an
             end-to-end revalidation every time, if, based solely on the
             origin server's Expires, the cached response is stale.)



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        proxy-revalidate: The proxy-revalidate directive has the same
             meaning as the must-revalidate directive, except that it
             does not apply to non-shared user agent caches. It can be
             used on a response to an authenticated request to permit
             the user's cache to store and later return the response
             without needing to revalidate it (since it has already been
             authenticated once by that user), while still requiring
             proxies that service many users to revalidate each time (in
             order to make sure that each user has been authenticated).
             Note that such authenticated responses also need the public
             cache control directive in order to allow them to be cached
             at all.

        max-age: When an intermediate cache is forced, by means of a
             max-age=0 directive, to revalidate its own cache entry, and
             the client has supplied its own validator in the request,
             the supplied validator might differ from the validator
             currently stored with the cache entry. In this case, the
             cache MAY use either validator in making its own request
             without affecting semantic transparency.

             However, the choice of validator might affect performance.
             The best approach is for the intermediate cache to use its
             own validator when making its request. If the server
             replies with 304 (Not Modified), then the cache can return
             its now validated copy to the client with a 200 (OK)
             response. If the server replies with a new entity and cache
             validator, however, the intermediate cache can compare the
             returned validator with the one provided in the client's
             request, using the strong comparison function. If the
             client's validator is equal to the origin server's, then
             the intermediate cache simply returns 304 (Not Modified).
             Otherwise, it returns the new entity with a 200 (OK)
             response.

14.11 Connection

   See [H14.10]. The use of the connection option "close" in RTSP
   messages SHOULD be limited to error messages when the server is
   unable to recover and therefore see it necessary to close the
   connection. The reason is that the client has the choice of
   continuing using a connection indefinitely, as long as it sends valid
   messages.


14.12 Connection-Credentials

   The Connection-Credentials response header is used to carry the



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   credentials of any next hop that need to be approved by the
   requestor. It SHALL only be used in server to client responses.

   The Connection-Credentials header in an RTSP response SHALL, if
   included, contain the credentials information of the next hop that an
   intermediary needs to securely connect to. The credential MUST
   include the URI of the next proxy or server and the DER encoded
   X.509v3  [16] certificate in base64 [36].


   Example:
     Accept-Credentials:"rtsps://proxy2.example.com/";MIIDNTCCAp6gA...



14.13 Content-Base

   The Content-Base entity-header field may be used to specify the base
   URI for resolving relative URIs within the entity.

   Content-Base: rtsp://media.example.com/movie/twister


   If no Content-Base field is present, the base URI of an entity is
   defined either by its Content-Location (if that Content-Location URI
   is an absolute URI) or the URI used to initiate the request, in that
   order of precedence. Note, however, that the base URI of the contents
   within the entity-body may be redefined within that entity-body.

14.14 Content-Encoding

   See [H14.11]

14.15 Content-Language

   See [H14.12]

14.16 Content-Length

   The Content-Length general-header field contains the length of the
   content of the method (i.e. after the double CRLF following the last
   header). Unlike HTTP, it MUST be included in all messages that carry
   content beyond the header portion of the message. If it is missing, a
   default value of zero is assumed. It is interpreted according to
   [H14.13].

14.17 Content-Location




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   See [H14.14]

14.18 Content-Type

   See [H14.17]. Note that the content types suitable for RTSP are
   likely to be restricted in practice to presentation descriptions and
   parameter-value types.

14.19 CSeq

   The CSeq general-header field specifies the sequence number for an
   RTSP request-response pair. This field MUST be present in all
   requests and responses. For every RTSP request containing the given
   sequence number, the corresponding response will have the same
   number. Any retransmitted request MUST contain the same sequence
   number as the original (i.e. the sequence number is not incremented
   for retransmissions of the same request). For each new RTSP request
   the CSeq value SHALL be incremented by one. The initial sequence
   number MAY be any number, however it is RECOMMENDED to start at 1.
   Each sequence number series is unique between each requester and
   responder, i.e. the client has one series for its request to a server
   and the server has another when sending request to the client. Each
   requester and responder is identified with its network address.

   Example:

   CSeq: 239



14.20 Date

   See [H14.18]. An RTSP message containing a body MUST include a Date
   header if the sending host has a clock. Servers SHOULD include a Date
   header in all other RTSP messages.

14.21 ETag

   The ETag response header MAY be included in DESCRIBE or SETUP
   responses. The entity tag returned in a DESCRIBE response is for the
   included entity, while for SETUP it refers to the media resource just
   set up. This differentiation allows for cache validation of both
   session description and the media resource associated with the
   description. If the ETag is provided both inside the entity, e.g.
   within the "a=etag" attribute in SDP, and in the response message,
   then both tags SHALL be identical. It is RECOMMENDED that the ETag is
   primarily given in the RTSP response message, to ensure that caches
   can use the ETag without requiring content inspection.



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   SETUP and DESCRIBE requests can be made conditional upon the ETag
   using the headers If-Match (Section 14.25) and If-None-Match (Section
   14.27).

14.22 Expires

   The Expires entity-header field gives a date and time after which the
   description or media-stream should be considered stale. The
   interpretation depends on the method:

        DESCRIBE response: The Expires header indicates a date and time
             after which the description SHOULD be considered stale.

        SETUP response: The Expires header indicate a date and time
             after which the media stream SHOULD be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with
   the origin server (or with an intermediate cache that has a fresh
   copy of the entity). See section 15 for further discussion of the
   expiration model.

   The presence of an Expires field does not imply that the original
   resource will change or cease to exist at, before, or after that
   time.

   The format is an absolute date and time as defined by HTTP-date in
   [H3.3]; it MUST be in RFC1123-date format:

   An example of its use is

     Expires: Thu, 01 Dec 1994 16:00:00 GMT



   RTSP/1.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).

   To mark a response as "already expired," an origin server should use
   an Expires date that is equal to the Date header value. To mark a
   response as "never expires," an origin server SHOULD use an Expires
   date approximately one year from the time the response is sent.
   RTSP/1.0 servers SHOULD NOT send Expires dates more than one year in
   the future.

   The presence of an Expires header field with a date value of some
   time in the future on a media stream that otherwise would by default



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   be non-cacheable indicates that the media stream is cacheable, unless
   indicated otherwise by a Cache-Control header field (Section 14.10).

14.23 From

   See [H14.22].

14.24 Host

   The Host HTTP request header field [H14.23] is not needed for RTSP,
   and SHALL NOT be sent. It SHALL be silently ignored if received.

14.25 If-Match

   See [H14.24].

   The If-Match request-header field is especially useful for ensuring
   the integrity of the presentation description, in both the case where
   it is fetched via means external to RTSP (such as HTTP), or in the
   case where the server implementation is guaranteeing the integrity of
   the description between the time of the DESCRIBE message and the
   SETUP message. By including the ETag given in or with the session
   description in a SETUP request, the client ensures that resources set
   up are matching the description. A SETUP request for which the ETag
   validation check fails, SHALL responde using 412 (Precondition
   Failed).

   This validation check is also very useful if a session has been
   redirected from one server to another.

14.26 If-Modified-Since

   The If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional. If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP). Instead, a 304 (Not Modified)
   response SHALL be returned without any message-body.

   An example of the field is:

     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT



14.27 If-None-Match

   See [H14.26].



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   This request header can be used with entity tags to make DESCRIBE
   requests conditional. A new session description is retrieved only if
   another entity than the already available would be included. If the
   entity available for delivery is matching the one the client already
   has, then a 304 (Not Modified) response is given.

14.28 Last-Modified

   The Last-Modified entity-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified. See [H14.29]. For the methods
   DESCRIBE, the header field indicates the last modification date and
   time of the description, for SETUP that of the media stream.

14.29 Location

   See [H14.30].

14.30 Proxy-Authenticate

   See [H14.33].

14.31 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-
   sensitive features that MUST be supported by the proxy. Any Proxy-
   Require header features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client using the
   Unsupported header. The proxy SHALL use the 551 (Option Not
   Supported) status code in the response. Any feature tag included in
   the Proxy-Require does not apply to the end-point (server or client).
   To ensure that a feature is supported by both proxies and servers the
   tag needs to be included in also a Require header.

   See Section 14.37 for more details on the mechanics of this message
   and a usage example.

   Example of use:

      Proxy-Require: play.basic




14.32 Proxy-Supported

   The Proxy-Supported header field enumerates all the extensions
   supported by the proxy using feature tags. The header carries the



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   intersection of extensions supported by the forwarding proxies. The
   Proxy-Supported header MAY be included in any request by a proxy. It
   SHALL be added by any proxy if the Supported header is present in a
   request. When present in a request, the receiver MUST in the response
   copy the received Proxy-Supported header.

   The Proxy-Supported header field contains a list of feature-tags
   applicable to proxies, as described in Section 3.7. The list are the
   intersection of all feature-tags understood by the proxies. To
   achieve an intersection, the proxy adding the Proxy-Supported header
   includes all proxy feature-tags it understands. Any proxy receiving a
   request with the header, checks the list and removes any feature tag
   it do not support. A Proxy-Supported header present in the response
   SHALL NOT be touched by the proxies.

   Example:

     C->P1: OPTIONS rtsp://example.com/ RTSP/1.0
            Supported: foo, bar, blech

    P1->P2: OPTIONS rtsp://example.com/ RTSP/1.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
            Via: 1.0 prox1.example.com

    P2->S:  OPTIONS rtsp://example.com/ RTSP/1.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-blech
            Via: 1.0 prox1.example.com, 1.0 prox2.example.com

     S->C:  RTSP/1.0 200 OK
            Supported: foo, bar, baz
            Proxy-Supported: proxy-foo, proxy-blech
            Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
            Via: 1.0 prox1.example.com, 1.0 prox2.example.com




14.33 Public

   The Public response header field lists the set of methods supported
   by the response sender. This header applies to the general
   capabilities of the sender and its only purpose is to indicate the
   sender's capabilities to the recipient. The methods listed may or may
   not be applicable to the Request-URI; the Allow header field (section
   14.7) MAY be used to indicate methods allowed for a particular URI.




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   Example of use:

      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN



   In the event that there are proxies between the sender and the
   recipient of a response, each intervening proxy MUST modify the
   Public header field to remove any methods that are not supported via
   that proxy. The resulting Public header field will contain an
   intersection of the sender's methods and the methods allowed through
   by the intervening proxies.

        In general proxies should allow all methods to
        transparently pass through from the sending RTSP agent to
        the receiving RTSP agent, but there may be cases where this
        is not desirable for a given proxy. Modification of the
        Public response header field by the intervening proxies
        ensures that the request sender gets an accurate response
        indicating the methods that can be used on the target agent
        via the proxy chain.

14.34 Range

   The Range request and response header field specifies a range of
   time. The header is used in PLAY request to indicate the range of
   time the client desires the server to play back. The Range header in
   a PLAY indicates what range of time that is actually being played. In
   a SETUP response the header MAY be used, to ensure correct
   synchronization information after changes of transport parameters.

   The range can be specified in a number of units. This specification
   defines the smpte (Section 3.4), npt (Section 3.5), and clock
   (Section 3.6) range units. Within RTSP, byte ranges [H14.35.1] are
   normally not meaningful, and the behavior is unspecified, but they
   and other extended units MAY be used. Servers supporting the Range
   header MUST understand the NPT range format and SHOULD understand the
   SMPTE range format. If the Range header is given in a time format
   that is not understood, the recipient should return 456 (Header Field
   Not Valid for Resource) and include a Accept-Ranges header indicating
   which time format that is supported for this resource.

   The header MAY contain a time parameter in UTC, specifying the time
   at which the operation is to be made effective. This functionality
   SHALL only be used with the REDIRECT method.

   Ranges are half-open intervals, including the first point, but
   excluding the second point. In other words, a range of A-B starts



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   exactly at time A, but stops just before B. Only the start time of a
   media unit such as a video or audio frame is relevant. As an example,
   assume that video frames are generated every 40 ms. A range of
   10.0-10.1 would include a video frame starting at 10.0 or later time
   and would include a video frame starting at 10.08, even though it
   lasted beyond the interval. A range of 10.0-10.08, on the other hand,
   would exclude the frame at 10.08.

   Example:

     Range: clock=19960213T143205Z-;time=19970123T143720Z



        The notation is similar to that used for the HTTP/1.1 [4]
        byte-range header. It allows clients to select an excerpt
        from the media object, and to play from a given point to
        the end as well as from the current location to a given
        point. The start of playback can be scheduled for any time
        in the future, although a server may refuse to keep server
        resources for extended idle periods.

   By default, range intervals increase, where the second point is
   larger than the first point.

   Example:

       Range: npt=10-15



   However, range intervals can also decrease if the Scale header (see
   section  14.39) indicates a negative scale value. For example, this
   would be the case when a playback in reverse is desired.

   Example:

       Scale: -1
       Range: npt=15-10



   Decreasing ranges are still half open intervals as described above.
   Thus, For range A-B, A is closed and B is open. In the above example,
   15 is closed and 10 is open. An exception to this rule is the case
   when B=0 in a decreasing range. In this case, the range is closed on
   both ends, as otherwise there would be no way to reach 0 on a reverse
   playback.



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   Example:

       Scale: -1
       Range: npt=15-0



   In this range both 15 and 0 are closed.

   A decreasing range interval without a corresponding negative Scale
   header is not valid.

14.35 Referer

   See [H14.36]. The URI refers to that of the presentation description,
   typically retrieved via HTTP.

14.36 Retry-After

   See [H14.37].

14.37 Require

   The Require request-header field is used by clients or servers to
   ensure that the other end-point supports features that are required
   in respect to this request. It can also be used to query if the other
   end-point supports certain features, however the use of the Supported
   (Section  14.43) is much more effective in this purpose. The server
   MUST respond to this header by using the Unsupported header to
   negatively acknowledge those feature-tags which are NOT supported.
   The response SHALL use the error code 551 (Option Not Supported).
   This header does not apply to proxies, for the same functionality in
   respect to proxies see, header Proxy-Require (Section 14.31).


        This is to make sure that the client-server interaction
        will proceed without delay when all features are understood
        by both sides, and only slow down if features are not
        understood (as in the example below). For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes state ambiguity
        when the client requires features that the server does not
        understand.

   Example:

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0



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           CSeq: 302
           Require: funky-feature
           Funky-Parameter: funkystuff

   S->C:   RTSP/1.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature




   In this example, "funky-feature" is the feature-tag which indicates
   to the client that the fictional Funky-Parameter field is required.
   The relationship between "funky-feature" and Funky-Parameter is not
   communicated via the RTSP exchange, since that relationship is an
   immutable property of "funky-feature" and thus should not be
   transmitted with every exchange.

   Proxies and other intermediary devices SHALL ignore this header. If a
   particular extension requires that intermediate devices support it,
   the extension should be tagged in the Proxy-Require field instead
   (see Section 14.31).

14.38 RTP-Info

   The RTP-Info response-header field is used to set RTP-specific
   parameters in the PLAY response. These parameters correspond to the
   synchronization source identified by the first value of the ssrc
   parameter of the Transport response header in the SETUP response. For
   streams using RTP as transport protocol the RTP-Info header SHOULD be
   part of a 200 response to PLAY.


        The exclusion of the RTP-Info in a PLAY response for RTP
        transported media will result in that a client needs to
        synchronize the media streams using RTCP. This may have
        negative impact as the RTCP can be lost, and does not need
        to be particulary timely in their arrival. Also
        functionality as informing the client from which packet a
        seek has occurred is affected.

   The RTP-Info MAY also be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   11.3.

   The header can carry the following parameters:

        url: Indicates the stream URI which for which the following RTP



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             parameters correspond, this URI MUST be the same used in
             the SETUP request for this media stream. Any relative URI
             SHALL use the Request-URI as base URI.

        seq: Indicates the sequence number of the first packet of the
             stream that is direct result of the request.  This allows
             clients to gracefully deal with packets when seeking.  The
             client uses this value to differentiate packets that
             originated before the seek from packets that originated
             after the seek. Note that a client may not receive the
             packet with the expressed sequence number, and instead
             packets with a higher sequence number, due to packet loss
             or reordering.

        rtptime: Indicates the RTP timestamp corresponding to the time
             value in the Range response header. (Note: For aggregate
             control, a particular stream may not actually generate a
             packet for the Range time value returned or implied. Thus,
             there is no guarantee that the packet with the sequence
             number indicated by seq actually has the timestamp
             indicated by rtptime.) The client uses this value to
             calculate the mapping of RTP time to NPT.


             A mapping from RTP timestamps to NTP timestamps (wall
             clock) is available via RTCP. However, this
             information is not sufficient to generate a mapping
             from RTP timestamps to NPT. Furthermore, in order to
             ensure that this information is available at the
             necessary time (immediately at startup or after a
             seek), and that it is delivered reliably, this mapping
             is placed in the RTSP control channel.

             In order to compensate for drift for long, uninterrupted
             presentations, RTSP clients should additionally map NPT to
             NTP, using initial RTCP sender reports to do the mapping,
             and later reports to check drift against the mapping.

   Additionally, the RTP-Info header parameter fields only apply to a
   single SSRC within a stream (the first SSRC reported in the transport
   response header; see section  14.45). If there are multiple
   synchronization sources (SSRCs) present within a RTP session
   transmitting media, RTCP needs to be used to map RTP and NTP
   timestamps for those sources, for both synchronization and drift-
   checking. Due to backwards compatibility reasons these shortcomings
   can't be fixed without defining a new header, which is for future
   work if needed.




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   Additional constraint: The syntax element "safe-url" (see section
   18.2.3) MUST NOT contain the semicolon (";") or comma (",")
   characters. The quoted-url form SHOULD only be used when an URI does
   not meet the safe-url constraint, in order to ensure compatibility
   with implementations conformant to RFC 2326 [1].

   Example:

   RTP-Info: url=rtsp://example.com/bar.avi/streamid=0;seq=45102,
             url=rtsp://example.com/bar.avi/streamid=1;seq=30211



14.39 Scale

   A scale value of 1 indicates normal play at the normal forward
   viewing rate. If not 1, the value corresponds to the rate with
   respect to normal viewing rate. For example, a ratio of 2 indicates
   twice the normal viewing rate ("fast forward") and a ratio of 0.5
   indicates half the normal viewing rate. In other words, a ratio of 2
   has normal play time increase at twice the wallclock rate. For every
   second of elapsed (wallclock) time, 2 seconds of content will be
   delivered. A negative value indicates reverse direction. For certain
   media transports this may require certain considerations to work
   consistent, see section B.1 for description on how RTP handles this.

   Unless requested otherwise by the Speed parameter, the data rate
   SHOULD not be changed. Implementation of scale changes depends on the
   server and media type. For video, a server may, for example, deliver
   only key frames or selected key frames. For audio, it may time-scale
   the audio while preserving pitch or, less desirably, deliver
   fragments of audio.

   The server should try to approximate the viewing rate, but may
   restrict the range of scale values that it supports. The response
   MUST contain the actual scale value chosen by the server.

   If the server does not implement the possibility to scale, it will
   not return a Scale header. A server supporting Scale operations for
   PLAY SHALL indicate this with the use of the "play.scale" feature-
   tags.

   When indicating a negative scale for a reverse playback, the Range
   header MUST indicate a decreasing range as described in section
   14.34.

   Example of playing in reverse at 3.5 times normal rate:




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     Scale: -3.5
     Range: npt=15-10



14.40 Speed

   The Speed request-header field requests the server to deliver data to
   the client at a particular speed, contingent on the server's ability
   and desire to serve the media stream at the given speed.
   Implementation by the server is OPTIONAL. The default is the bit rate
   of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of
   2.0 indicates that data is to be delivered twice as fast as normal. A
   speed of zero is invalid. All speeds may not be possible to support.
   Therefore the actual used speed MUST be included in the response. The
   lack of a response header is indication of lack of support from the
   server of this functionality. Support of the speed functionality are
   indicated by the "play.speed" featuretag.

   Example:

     Speed: 2.5



   Use of this field changes the bandwidth used for data delivery. It is
   meant for use in specific circumstances where preview of the
   presentation at a higher or lower rate is necessary. Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary. When data is delivered over UDP, it is highly
   recommended that means such as RTCP be used to track packet loss
   rates. If the data transport is performed over public best-effort
   networks the sender SHOULD perform congestion control of the
   stream(s). This can result in that the communicated speed is
   impossible to maintain.

14.41 Server

   See [H14.38], however the header syntax is corrected in section
   18.2.3.

14.42 Session

   The Session request-header and response-header field identifies an
   RTSP session. An RTSP session is created by the server as a result of



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   a successful SETUP request and in the response the session identifier
   is given to the client. The RTSP session exist until destroyed by a
   TEARDOWN or timed out by the server.

   The session identifier is chosen by the server (see Section 3.3) and
   MUST be returned in the SETUP response. Once a client receives a
   session identifier, it SHALL be included in any request related to
   that session.  This means that the Session header MUST be included in
   a request using the following methods: PLAY, PAUSE, PING, and
   TEARDOWN, and MAY be included in SETUP, OPTIONS, SET_PARAMETER,
   GET_PARAMETER, and REDIRECT, and SHALL NOT be included in DESCRIBE.
   In an RTSP response the session header SHALL be included in methods,
   SETUP, PING, PLAY, and PAUSE, and MAY be included in methods,
   TEARDOWN, and REDIRECT, and if included in the request of the
   following methods it SHALL also be included in the response, OPTIONS,
   GET_PARAMETER, and SET_PARAMETER, and SHALL NOT be included in
   DESCRIBE.

   Note that RFC 2326 servers and client may in some cases not include
   or return a Session header when expected according to the above text.
   Any client or server is RECOMMENDED to be forgiving of this error if
   possible (which it is in many cases).

   The timeout parameter MAY be included in a SETUP response, and SHALL
   NOT be included in requests. The server uses it to indicate to the
   client how long the server is prepared to wait between RTSP commands
   or other signs of life before closing the session due to lack of
   activity (see below and Section A). The timeout is measured in
   seconds, with a default of 60 seconds (1 minute). The length of the
   session timeout SHALL NOT be changed in a established session.

   The mechanisms for showing liveness of the client is, any RTSP
   request with a Session header, if RTP & RTCP is used an RTCP message,
   or through any other used media protocol capable of indicating
   liveness of the RTSP client. It is RECOMMENDED that a client does not
   wait to the last second of the timeout before trying to send a
   liveness message. The RTSP message may be lost or when using reliable
   protocols, such as TCP, the message may take some time to arrive
   safely at the receiver. To show liveness between RTSP request issued
   to accomplish other things, the following mechanisms can be used, in
   descending order of preference:

        RTCP: If RTP is used for media transport RTCP SHOULD be used. If
             RTCP is used to report transport statistics, it SHALL also
             work as keep alive. The server can determine the client by
             used network address and port together with the fact that
             the client is reporting on the servers SSRC(s). A downside
             of using RTCP is that it only gives statistical guarantees



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             to reach the server. However that probability is so low
             that it can be ignored in most cases. For example, a
             session with 60 seconds timeout and enough bitrate assigned
             to RTCP messages to send a message from client to server on
             average every 5 seconds. That client have for a network
             with 5 % packet loss, the probability to fail showing
             liveness sign in that session within the timeout interval
             of 2.4*E-16. In sessions with shorter timeout times, or
             much higher packet loss, or small RTCP bandwidths SHOULD
             also use any of the mechanisms below.

        PING: The use of the PING method is the best of the RTSP based
             methods. It has no other effects than updating the timeout
             timer. In that way it will be a minimal message, that also
             does not cause any extra processing for the server. The
             downside is that it may not be implemented. A client SHOULD
             use a OPTIONS request to verify support of the PING at the
             server. It is also possible to detect support by sending a
             PING to the server. If a 200 (OK) message is received the
             server supports it. In case a 501 (Not Implemented) is
             received it does not support PING and there is no meaning
             in continue trying.  Also the reception of a error message
             will also mean that the liveness timer has not been
             updated.

        SET_PARAMETER: When using SET_PARAMETER for keep alive, no body
             SHOULD be included. This method is basically as good as
             PING, however the implementation support of the method is
             today limited. The same considerations as for PING apply
             regarding checking of support in server and proxies.

        OPTIONS: This method does also work. However it causes the
             server to perform unnecessary processing and result in
             bigger responses than necessary for the task. The reason
             for this is that the Public is always included creating
             overhead.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections. RTSP requests for a given session
   can use different URIs (Presentation and media URIs).  Note, that
   there are restrictions depending on the session which URIs that are
   acceptable for a given method. However, multiple "user" sessions for
   the same URI from the same client will require use of different
   session identifiers.

        The session identifier is needed to distinguish several
        delivery requests for the same URI coming from the same
        client.



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   The response 454 (Session Not Found) SHALL be returned if the session
   identifier is invalid.

14.43 Supported

   The Supported header field enumerates all the extensions supported by
   the client or server using feature tags. The header carries the
   extensions supported by the message sending entity.  The Supported
   header MAY be included in any request.  When present in a request,
   the receiver MUST respond with its corresponding Supported header.
   Note, also in 4xx and 5xx responses is the supported header included.

   The Supported header field contains a list of feature-tags, described
   in Section 3.7, that are understood by the client or server.

   Example:

     C->S:  OPTIONS rtsp://example.com/ RTSP/1.0
            Supported: foo, bar, blech

     S->C:  RTSP/1.0 200 OK
            Supported: bar, blech, baz



14.44 Timestamp

   The Timestamp general-header field describes when the client sent the
   request to the server. The value of the timestamp is of significance
   only to the client and may use any timescale. The server MUST echo
   the exact same value and MAY, if it has accurate information about
   this, add a floating point number indicating the number of seconds
   that has elapsed since it has received the request. The timestamp is
   used by the client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions. It also
   resolves retransmission ambiguities for unreliable transport of RTSP.

14.45 Transport

   The Transport request and response header field indicates which
   transport protocol is to be used and configures its parameters such
   as destination address, compression, multicast time-to-live and
   destination port for a single stream. It sets those values not
   already determined by a presentation description.

   Transports are comma separated, listed in order of preference.
   Parameters may be added to each transport, separated by a semicolon.
   The server SHOULD return a Transport response-header field in the



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   response to indicate the values actually chosen. The Transport header
   field MAY also be used to change certain transport parameters. A
   server MAY refuse to change parameters of an existing stream.

   A Transport request header field MAY contain a list of transport
   options acceptable to the client, in the form of multiple
   transportspec entries. In that case, the server MUST return the
   single option (transport-spec) which was actually chosen. The number
   of transportspec entries is expected to be limited as the client will
   get guidance on what configurations that are possible from the
   presentation description.

   A transport-spec transport option may only contain one of any given
   parameter within it. Parameters may be given in any order.
   Additionally, it may only contain the unicast or multicast transport
   type parameter. Unknown transport parameters SHALL be ignored. The
   requester need to ensure that the responder understands the
   parameters through the use of feature tags and the Require header.

   A request or a response including a transport header with any
   parameter not defined in RFC 2326 SHOULD use the Require header and
   the "play.basic" feature tag. This is to ensure consistent behavior
   with the new parameters. Any parameters part of future extensions
   requires clarification if they are safe to ignore in accordance to
   this specification, or is required to be understood. If a parameter
   is required to be understood, then a feature tag MUST be defined and
   used in the Require and/or Proxy-Require headers.


        The Transport header field is restricted to describing a
        single media stream. (RTSP can also control multiple
        streams as a single entity.) Making it part of RTSP rather
        than relying on a multitude of session description formats
        greatly simplifies designs of firewalls.

   The syntax for the transport specifier is

   transport/profile/lower-transport.


   The default value for the "lower-transport" parameters is specific to
   the profile. For RTP/AVP, the default is UDP.

   There is three different methods for how to specify where the media
   should be delivered:

        o The presence of this parameter and its values indicates
          address and port pairs for one or more IP flow necessary for



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          the media transport. This is an improved version of the
          Destination parameter.

        o The presence of this parameter and its value indicates what IP
          address the media SHALL be delivered to. This method is kept
          for backwards compatibility reasons, dest_addr is a better
          choice.

        o The lack of of both of the above parameters indicates that the
          server SHALL send media to same address for which the RTSP
          messages originates.

   The choice of method for indicating where the media is to be
   delivered depends on the use case. In many case the only allowed
   method will be to use no explicit indication and have the server
   deliver media to the source of the RTSP messages.

   An RTSP proxy will also need to take care. If the media is not
   desired to be routed through the proxy, the proxy will need to
   introduce the destination indication.

   Below are the configuration parameters associated with transport:

   General parameters:

        unicast / multicast: This parameter is a mutually exclusive
             indication of whether unicast or multicast delivery will be
             attempted. One of the two values MUST be specified. Clients
             that are capable of handling both unicast and multicast
             transmission MUST indicate such capability by including two
             full transport-specs with separate parameters for each.


        destination: The address of the stream recipient to which a
             stream will be sent. The client originating the RTSP
             request may specify the destination address of the stream
             recipient with the destination parameter. When the
             destination field is specified, the recipient may be a
             different party than the originator of the request. To
             avoid becoming the unwitting perpetrator of a remote-
             controlled denial-of-service attack, a server SHOULD
             authenticate the client originating the request and SHOULD
             log such attempts before allowing the client to direct a
             media stream to a recipient address not chosen by the
             server. Implementations cannot rely on TCP as reliable
             means of client identification.

             The server SHOULD NOT allow the destination field to be set



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             unless a mechanism exists in the system to authorize the
             request originator to direct streams to the recipient. It
             is preferred that this authorization be performed by the
             recipient itself and the credentials passed along to the
             server. However, in certain cases, such as when recipient
             address is a multicast group, or when the recipient is
             unable to communicate with the server in an out-of-band
             manner, this may not be possible. In these cases server may
             chose another method such as a server-resident
             authorization list to ensure that the request originator
             has the proper credentials to request stream delivery to
             the recipient.

             This parameter SHALL NOT be used when src_addr and
             dest_addr is used in a transport declaration. For IPv6
             addresses it is RECOMMENDED that they be given as fully
             qualified domain to make it backwards compatible with RFC
             2326 implementations.

        source: If the source address for the stream is different than
             can be derived from the RTSP end-point address (the server
             in playback), the source address SHOULD be specified. To
             maintain backwards compatibility with RFC 2326, any IPv6
             host's address needs to be given as a fully qualified
             domain name. This parameter SHALL NOT be used when src_addr
             and dest_addr is used in a transport declaration.


             This information may also be available through SDP.
             However, since this is more a feature of transport
             than media initialization, the authoritative source
             for this information should be in the SETUP response.

        layers: The number of multicast layers to be used for this media
             stream. The layers are sent to consecutive addresses
             starting at the destination address.

        dest_addr: A general destination address parameter that can
             contain one or more address and port pair. For each
             combination of Protocol/Profile/Lower Transport the
             interpretation of the address or addresses needs to be
             defined.  The host address part of the tuple MAY be empty,
             for example ":8000", in cases when only destination port is
             desired to be specified.

             The client or server SHALL NOT use this parameter unless
             both client and server has shown support. This parameter
             MUST be supported by client and servers that implements



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             this specification. Support is indicated by the use of the
             feature-tag "play.basic". This parameter SHALL NOT be used
             in the same transport specification as any of the
             parameters "destination", "source", "port", "client_port",
             and "server_port".

             The same security consideration that are given for the
             "Destination" parameter does also applies to this
             parameter. This parameter can be used for redirecting
             traffic to recipient not desiring the media traffic.


        src_addr: A general source address parameter that can contain
             one or more address and port pairs. For each combination of
             Protocol/Profile/Lower Transport the interpretation of the
             address or addresses needs to be defined. The client or
             server SHALL NOT use this parameter unless both client and
             server have shown support. This parameter MUST be supported
             by client and servers that implement this specification.
             Support is indicated by the use the feature-tag
             "play.basic". This parameter SHALL NOT be used in the same
             transport specification as any of the parameters
             "destination", "source", "port", "client_port", and
             "server_port".

             This parameter MUST be specified by the server if it
             transmits media packets from another address than the one
             RTSP messages are sent to. This will allow the client to
             verify source address and give it a destination address for
             its RTCP feedback packets if RTP is used. The address or
             addresses indicated in the src_addr parameter SHOULD be
             used both for sending and receiving of the media streams
             data packets. The main reasons are threefold: First,
             indicating the port and source address(s) lets the receiver
             know where from the packets is expected to originate.
             Secondly, traversal of NATs are greatly simplified when
             traffic is flowing symmetrically over a NAT binding.
             Thirdly, certain NAT traversal mechanisms, needs to know to
             which address and port to send so called "binding packets"
             from the receiver to the sender, thus creating a address
             binding in the NAT that the sender to receiver packet flow
             can use.

        mode: The mode parameter indicates the methods to be supported
             for this session. Valid values are PLAY and RECORD. If not
             provided, the default is PLAY.  The RECORD value was
             defined in RFC 2326 and is deprecated in this
             specification.



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        append: The append parameter was used together with RECORD and
             is now deprecated.

        interleaved: The interleaved parameter implies mixing the media
             stream with the control stream in whatever protocol is
             being used by the control stream, using the mechanism
             defined in Section 12. The argument provides the channel
             number to be used in the $ statement and MUST be present.
             This parameter MAY be specified as a range, e.g.,
             interleaved=4-5 in cases where the transport choice for the
             media stream requires it, e.g. for RTP with RTCP.  The
             channel number given in the request are only a guidance
             from the client to the server on what channel number(s) to
             use. The server MAY set any valid channel number in the
             response. The declared channel(s) are bi-directional, so
             both end-parties MAY send data on the given channel. One
             example of such usage is the second channel used for RTCP,
             where both server and client sends RTCP packets on the same
             channel.


             This allows RTP/RTCP to be handled similarly to the
             way that it is done with UDP, i.e., one channel for
             RTP and the other for RTCP.

   Multicast-specific:

        ttl: multicast time-to-live.

   RTP-specific:

   These parameters are MAY only be used if the media transport protocol
   is RTP.

        port: This parameter provides the RTP/RTCP port pair for a
             multicast session. It is should be specified as a range,
             e.g., port=3456-3457

        client_port: This parameter provides the unicast RTP/RTCP port
             pair on the client where media data and control information
             is to be sent. It is specified as a range, e.g.,
             port=3456-3457. This parameter SHALL NOT be used when
             src_addr and dest_addr is used in a transport declaration.

        server_port: This parameter provides the unicast RTP/RTCP port
             pair on the server where media data and control information
             is to be sent. It is specified as a range, e.g.,
             port=3456-3457. This parameter SHALL NOT be used when



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             src_addr and dest_addr is used in a transport declaration.

        ssrc: The ssrc parameter, if included in a SETUP response,
             indicates the RTP SSRC [17] value(s) that will be used by
             the media server for RTP packets within the stream. It is
             expressed as an eight digit hexadecimal value. Multiple
             values MAY only be specified if the client has indicated
             support for this specification, i.e.  if including multiple
             SSRC values, the request is required to include the
             "Require: play.basic" or "Supported: play.basic" headers.
             If no such support is present only a single value SHALL be
             included.

             If the server does not act as a synchronization source for
             stream data (for instance, server is a translator,
             reflector, etc.), and only a single value can be specified,
             the value will be the "packet sender's SSRC" that would
             have been used in the RTCP Receiver Reports generated by
             the server, regardless of whether the server actually
             generates RTCP RRs.

             The first SSRC value is the one that RTP-Info
             synchronization information relates to, see section  14.38.

             The functionality of specifying the ssrc parameter in a
             SETUP request is deprecated as it is incompatible with the
             specification of RTP in RFC 3550  [17]. If the parameter is
             included in the transport header of a SETUP request, the
             server MAY ignore it, and choose an appropriate SSRC for
             the stream. The server MAY set the ssrc parameter in the
             transport header of the response.

   The combination of transport protocol, profile and lower transport
   needs to be defined. A number of combinations are defined in the
   appendix B.

   Below is a usage example, showing a client advertising the capability
   to handle multicast or unicast, preferring multicast.  Since this is
   a unicast-only stream, the server responds with the proper transport
   parameters for unicast.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

     S->C: RTSP/1.0 200 OK



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           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;client_port=3456-3457;
               server_port=6256-6257;mode="PLAY"



14.46 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. In the case where the feature was specified
   via the Proxy-Require field (Section 14.31), if there is a proxy on
   the path between the client and the server, the proxy MUST send a
   response message with a status code of 551 (Option Not Supported).
   The request SHALL NOT be forwarded.

   See Section 14.37 for a usage example.

14.47 User-Agent

   See [H14.43] for explanation, however the syntax is clarified due to
   an error in RFC 2616. A Client SHOULD include this header in all RTSP
   messages it sends.

14.48 Vary

   See [H14.44]

14.49 Via

   See [H14.45].

14.50 WWW-Authenticate

   See [H14.47].

15 Caching

   In HTTP, response-request pairs are cached. RTSP differs
   significantly in that respect. Responses are not cacheable, with the
   exception of the presentation description returned by DESCRIBE.
   (Since the responses for anything but DESCRIBE and GET_PARAMETER do
   not return any data, caching is not really an issue for these
   requests.) However, it is desirable for the continuous media data,
   typically delivered out-of-band with respect to RTSP, to be cached,
   as well as the session description.




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   On receiving a SETUP or PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description. It can determine whether the copy is up-to-date by
   issuing a SETUP or DESCRIBE request, respectively, and comparing the
   Last-Modified header with that of the cached copy. If the copy is not
   up-to-date, it modifies the SETUP transport parameters as appropriate
   and forwards the request to the origin server. Subsequent control
   commands such as PLAY or PAUSE then pass the proxy unmodified. The
   proxy delivers the continuous media data to the client, while
   possibly making a local copy for later reuse. The exact behavior
   allowed to the cache is given by the cache-response directives
   described in Section 14.10. A cache MUST answer any DESCRIBE requests
   if it is currently serving the stream to the requestor, as it is
   possible that low-level details of the stream description may have
   changed on the origin-server.

   Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
   through" variety. Rather than retrieving the whole resource from the
   origin server, the cache simply copies the streaming data as it
   passes by on its way to the client. Thus, it does not introduce
   additional latency.

   To the client, an RTSP proxy cache appears like a regular media
   server, to the media origin server like a client. Just as an HTTP
   cache has to store the content type, content language, and so on for
   the objects it caches, a media cache has to store the presentation
   description.  Typically, a cache eliminates all transport-references
   (that is, multicast information) from the presentation description,
   since these are independent of the data delivery from the cache to
   the client.  Information on the encodings remains the same. If the
   cache is able to translate the cached media data, it would create a
   new presentation description with all the encoding possibilities it
   can offer.

16 Examples

   This section contains several different examples trying to illustrate
   possible ways of using RTSP. The examples can also help with the
   understanding of how functions of RTSP work. However remember that
   this is examples and the normative and syntax description in the
   other sections takes precedence. Please also note that many of the
   example MAY contain syntax illegal line breaks to accommodate the
   formatting restriction that the RFC series impose.

16.1 Media on Demand (Unicast)

   Client C requests a movie distributed from two different media
   servers A (audio.example.com ) and V (video.example.com ). The media



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   description is stored on a web server W. The media description
   contains descriptions of the presentation and all its streams,
   including the codecs that are available, dynamic RTP payload types,
   the protocol stack, and content information such as language or
   copyright restrictions. It may also give an indication about the
   timeline of the movie.

   In this example, the client is only interested in the last part of
   the movie.


   C->W: GET /twister.sdp HTTP/1.1
         Host: www.example.com
         Accept: application/sdp

   W->C: HTTP/1.0 200 OK
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 255
         Expires: 23 Jan 1998 15:35:06 GMT

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         e=adm@example.com
         a=range:npt=0-1:49:34
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://audio.example.com/twister/audio.en
         m=video 0 RTP/AVP 31
         a=control:rtsp://video.example.com/twister/video

   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057,
                    RTP/AVP/TCP;unicast;interleaved=0-1

   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Expires: 24 Jan 1997 15:35:12 GMT
         Cache-Control: public
         Accept-Ranges: NPT, SMPTE



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   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059,
                    RTP/AVP/TCP;unicast;interleaved=0-1

   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Cache-Control: public
         Expires: 24 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT, SMPTE

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url=rtsp://video.example.com/twister/video;
                   seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:35:13 GMT

   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
                   seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:13 GMT





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   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 12345678

   A->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:52 GMT

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 23456789

   V->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:36:52 GMT




   Even though the audio and video track are on two different servers,
   may start at slightly different times, and may drift with respect to
   each other, the client can synchronize the two using standard RTP
   methods, in particular the time scale contained in the RTCP sender
   reports. Initial synchronization is achieved through the RTP-Info and
   Range headers information in the PLAY response.

16.2 Streaming of a Container file

   For purposes of this example, a container file is a storage entity in
   which multiple continuous media types pertaining to the same end-user
   presentation are present. In effect, the container file represents an
   RTSP presentation, with each of its components being RTSP streams.
   Container files are a widely used means to store such presentations.
   While the components are transported as independent streams, it is
   desirable to maintain a common context for those streams at the
   server end.


        This enables the server to keep a single storage handle
        open easily. It also allows treating all the streams
        equally in case of any prioritization of streams by the
        server.

   It is also possible that the presentation author may wish to prevent



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   selective retrieval of the streams by the client in order to preserve
   the artistic effect of the combined media presentation. Similarly, in
   such a tightly bound presentation, it is desirable to be able to
   control all the streams via a single control message using an
   aggregate URI.

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URIs. In a
   container file it is also desirable to not write any URI parts which
   is not kept, when the container is distributed, like the host and
   most of the path element. Therefore this example also uses the "*"
   and relative URI in the delivered SDP.

   Client C requests a presentation from media server M. The movie is
   stored in a container file. The client has obtained an RTSP URI to
   the container file.


   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 257
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: 24 Jan 1997 15:35:06 GMT

         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         a=control: *
         a=range: npt=0-0:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Require: play.basic



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         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001;
                    src_addr="172.16.2.93:9000"/"172.16.2.93:9001"
                    ssrc=93CB001E
         Session: 12345678
         Expires: 24 Jan 1997 15:35:12 GMT
         Date: 23 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003;
                    src_addr="172.16.2.93:9002"/"172.16.2.93:9003";
                    ssrc=A813FC13
         Session: 12345678
         Expires: 24 Jan 1997 15:35:13 GMT
         Date: 23 Jan 1997 15:35:13 GMT
         Accept-Range: NPT

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-10, npt=30-
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:14 GMT
         Session: 12345678
         Range: npt=0-10, npt=30-623.10
         RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4;
            seq=12345;rtptime=3450012,
           url=rtsp://example.com/twister.3gp/trackID=1;
            seq=54321;rtptime=2876889




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   C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 5
         User-Agent: PhonyClient/1.2
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.0
         CSeq: 6
         User-Agent: PhonyClient/1.2
         Range: npt=34.57-623.10
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 6
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10
         RTP-Info: url=rtsp://example.com/twister.3gp/trackID=4;
            seq=12555;rtptime=6330012,
           url=rtsp://example.com/twister.3gp/trackID=1;
            seq=55021;rtptime=3132889




16.3 Single Stream Container Files

   Some RTSP servers may treat all files as though they are "container
   files", yet other servers may not support such a concept. Because of
   this, clients SHOULD use the rules set forth in the session
   description for Request-URIs, rather than assuming that a consistent
   URI may always be used throughout. Below are an example of how a
   multi-stream server might expect a single-stream file to be served:


   C->S: DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
         Accept: application/x-rtsp-mh, application/sdp
         CSeq: 1
         User-Agent: PhonyClient/1.2

   S->C: RTSP/1.0 200 OK



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         CSeq: 1
         Content-base: rtsp://foo.com/test.wav/
         Content-type: application/sdp
         Content-length: 48
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Expires: 23 Jan 1997 17:00:00 GMT

         v=0
         o=- 872653257 872653257 IN IP4 172.16.2.187
         s=mu-law wave file
         i=audio test
         t=0 0
         a=control: *
         m=audio 0 RTP/AVP 0
         a=control:streamid=0

   C->S: SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
         Transport: RTP/AVP/UDP;unicast;
            client_port=6970-6971;mode="PLAY"
         CSeq: 2
         User-Agent: PhonyClient/1.2

   S->C: RTSP/1.0 200 OK
         Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
                     server_port=6970-6971;mode="PLAY";ssrc=EAB98712
         CSeq: 2
         Session: 2034820394
         Expires: 23 Jan 1997 16:00:00 GMT
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT

   C->S: PLAY rtsp://foo.com/test.wav/ RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 2034820394

   S->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:08 GMT
         Session: 2034820394
         Range: npt=0-600
         RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
            seq=981888;rtptime=3781123






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   Note the different URI in the SETUP command, and then the switch back
   to the aggregate URI in the PLAY command.  This makes complete sense
   when there are multiple streams with aggregate control, but is less
   than intuitive in the special case where the number of streams is
   one. However the server has declared that the aggregated control URI
   in the SDP and therefore this is legal.

   In this case, it is also required that servers accept implementations
   that use the non-aggregated interpretation and use the individual
   media URI, like this:


   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2



16.4 Live Media Presentation Using Multicast

   The media server M chooses the multicast address and port. Here, it
   is assumed that the web server only contains a pointer to the full
   description, while the media server M maintains the full description.

   Editors note: Is this example really valid? In what situations does
   it make sense to do a setup to a multicast distribution channel, and
   also issue PLAY requests?


   C->W: GET /sessions.html HTTP/1.1
         Host: www.example.com

   W->C: HTTP/1.1 200 OK
         Content-Type: text/html

         <html>
           ...
           <href "Stremed Live Music performance"
              src="rtsp://live.example.com/concert/audio">
           ...
         </html>

   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 1
         Supported: play.basic, play.scale

   M->C: RTSP/1.0 200 OK
         CSeq: 1



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         Content-Type: application/sdp
         Content-Length: 181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Supported: play.basic

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         m=audio 3456 RTP/AVP 0
         c=IN IP4 224.2.0.1/16
         a=control: rtsp://live.example.com/concert/audio
         a=range:npt=0-

   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;multicast

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Transport: RTP/AVP;multicast;destination=224.2.0.1;
                    port=3456-3457;ttl=16
         Session: 0456804596
         Accept-Ranges: NPT, UTC

   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT
         Session: 0456804596
         Range:npt=1256-
         RTP-Info: url=rtsp://live.example.com/concert/audio;
                   seq=1473; rtptime=80000



16.5 Capability Negotiation

   This examples illustrate how the client and server determines their
   capability to support a special feature, in this case "play.scale".
   The server, through the clients request and the included Supported
   header, learns that the client is supporting this updated



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   specification, and also supports the playback time scaling feature of
   RTSP. The server's response contains the following feature related
   information to the client; it supports the updated specification
   (play.basic), the extended functionality of time scaling of content
   (play.scale), and one "example.com" proprietary feature
   (example.com.flight). The client also learns the methods supported
   (Public header) by the server for the indicated resource.


   C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/1.0
         CSeq: 1
         Supported: play.basic, play.scale
         User-Agent: PhonyClient/1.2

   S->C: RTSP/1.0 200 OK
         CSeq: 1
         Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
         Server: PhonyServer/2.0
         Supported: play.basic, play.scale, example.com.flight



   When the client sends its SETUP request it tells the server that it
   is requires support of the play.scale feature for this session by
   including the Require header.


   C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057,
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Require: play.scale

   S->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001
         Server: PhonyServer/2.0
         Accept-Ranges: NPT, SMPTE



17 Security Framework

   The RTSP security framework consists of two high level components:
   the pure authentication mechanisms based on HTTP authentication, and



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   the transport protection based on TLS, which is independent of RTSP.
   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security for HTTP is re-used to a large extent.

17.1 RTSP and HTTP Authentication

   RTSP and HTTP share common authentication schemes, and thus follow
   the same usage guidelines as specified in  [8] and also in [H15].
   Servers SHOULD implement both basic and digest [8] authentication.

   It should be stressed that using the HTTP authentication alone does
   not provide full control message security. Therefore, in environments
   requiring tighter security for the control messages, TLS SHOULD be
   used, see Section  17.2.

17.2 RTSP over TLS

   RTSP SHALL follow the same guidelines with regards to TLS [7] usage
   as specified for HTTP, see [18]. RTSP over TLS is separated from
   unsecured RTSP both on URI level and port level. Instead of using the
   "rtsp" scheme identifier in the URI, the "rtsps" scheme identifier
   MUST be used to signal RTSP over TLS. If no port is given in a URI
   with the "rtsps" scheme, port 322 SHALL be used for TLS over TCP/IP.

   When a client tries to setup an insecure channel to the server (using
   the "rtsp" URI), and the policy for the resource requires a secure
   channel, the server SHALL redirect the client to the secure service
   by sending a 301 redirect response code together with the correct
   Location URI (using the "rtsps" scheme).

   It should be noted that TLS allows for mutual authentication (when
   using both server and client certificates). Still, one of the more
   common way TLS is used is to only provide server side authentication
   (often to avoid client certificates). TLS is then used in addition to
   HTTP authentication, providing transport security and server
   authentication, while HTTP Authentication is used to authenticate the
   client.

   RTSP includes the possibility to keep a TCP session up between the
   client and server, throughout the RTSP session lifetime. It may be
   convenient to keep the TCP session, not only to save the extra setup
   time for TCP, but also the extra setup time for TLS (even if TLS uses
   the resume function, there will be almost two extra roundtrips).
   Still, when TLS is used, such behavior introduces extra active state
   in the server, not only for TCP and RTSP, but also for TLS. This may
   increase the vulnerability to DoS attacks.

   In addition to these recommendations, Section 17.3 gives further



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   recommendations of TLS usage with proxies.

17.3 Security and Proxies

   The nature of a proxy is often to act as a "man-in-the-middle", while
   security is often about preventing the existence of a "man-in-the-
   middle". This section provides the clients with the possibility to
   use proxies even when applying secure transports (TLS). The client
   needs to select between using the below specified procedure or using
   a TLS connection directly (by-passing any proxies) to the server. The
   choice may be dependent on policies.

   There are basically two categories of inspecting proxies, the
   transparent proxies (which the client is not aware of) and the non-
   transparent proxies (which the client is aware of). An infrastructure
   based on proxies requires that the trust model is such that both
   client and servers can trust the proxies to handle the RTSP messages
   correctly. To be able to trust a proxy, the client and server also
   needs to be aware of the proxy. Hence, transparent proxies cannot
   generally be seen as trusted and will not work well with security
   (unless they work only at transport layer). In the rest of this
   section any reference to proxy will be to a non-transparent proxy,
   which requires to inspect/manipulate the RTSP messages.

   The HTTP Authentication is built on the assumption of proxies and can
   provide user-proxy authentication and proxy-proxy/server
   authentication in addition to the client-server authentication.

   When TLS is applied and a proxy is used, the client will use the
   proxy's destination URI address when sending messages. This implies
   that for TLS, the client will authenticate the proxy server and not
   the end server. Note that, when the client checks the server
   certificate in TLS, it MUST check the proxy's identity (URI or
   possibly other known identity) against the proxy's identity as
   presented in the proxy's Certificate message.

   The problem is that for proxy accepted by the client, it needs to be
   provided information on which grounds it should accept the next-hop
   certificate. Both the proxy and the user may have rules for this, and
   the user have the possibility to select the desired behavior. To
   handle this case, the Accept-Credentials header (See Section 14.2) is
   used, where the client can force the proxy/proxies to relay back the
   certificates used by any intermediate proxies as well as the server.
   Given the assumption that the proxies are viewed as trusted, it gives
   the user a possibility to enforce policies to each trusted proxy of
   whether it should accept the next entity in the chain.

   A proxy MUST use TLS for the next hop if the RTSP request includes a



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   "rtsps" URI. TLS MAY be applied on intermediate links (e.g. between
   client and proxy, or between proxy and proxy), even if the resource
   and the end server does not require to use it.

17.3.1 Accept-Credentials

   The Accept-Credentials header can be used by the client to distribute
   simple authorization policies to intermediate proxies. The client
   includes the Accept-Credentials header to dictate how the proxy
   treats the server/next proxy certificate. There are currently three
   methods defined:

        Any, which means that the proxy (or proxies) SHALL accept
             whatever certificate presented. This is of course not a
             recommended option to use, but may be useful in certain
             circumstances (such as testing).


        Proxy, which means that the proxy (or proxies) MUST use its own
             policies to validate the certificate and decide whether to
             accept it or not. This is convenient in cases where the
             user has a strong trust relation with the proxy. Reason why
             a strong trust relation may exist are; personal/company
             proxy, proxy has a out-of-band policy configuration
             mechanism.

        User, which means that the proxy (or proxies) MUST send
             credential information about the next hop to the client for
             authorization. The client can then decide whether the proxy
             should accept the certificate or not. See section 17.3.2
             for further details.

   If the Accept-Credentials header is not included in the RTSP request
   from the client, the default method used SHALL be "Proxy". If
   something else than the "Proxy" method is used, the Accept-
   Credentials header SHALL always be included in the RTSP request from
   the client. This is because it cannot be assumed that the proxy
   always keeps the TLS state or the users previously preference between
   different RTSP messages (in particular if the time interval between
   the messages is long).

   The "Any" and "Proxy" methods does not require the proxy to provide
   any specific response, but only apply the policy as defined for
   respectively method. If the policy do not accept the credentials of
   the next hop, the entity SHALL respond with a message using status
   code 471 (Connection Credentials not accepted).

   An RTSP request in the direction server to client MUST NOT include



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   the Accept-Credential header. As for the non-secured communication,
   the possibility for these request depends on the presence of a client
   established connection.  However if the server to client request is
   in relation to a session established over a TLS secured channel, if
   MUST be sent in a TLS secured connection. That secured connection
   MUST also be the one used by the last client to server request. If no
   such transport connection exist at the time when the server desire to
   send the request, it silently fails.

   Further policies MAY be defined and registered, but should be done so
   with caution.

17.3.2 User approved TLS procedure

   For the "User" method each proxy MUST perform the the following
   procedure for each RTSP request:

        o Setup the TLS session to the next hop if not already present
          (i.e. run the TLS handshake, but do not send the RTSP
          request).

        o Extract the peer certificate for the TLS session.

        o Check if a matching identity and hash of the peer certificate
          is present in the Accept-Credentials header.  If present, send
          the message to the next hop, and conclude these procedures. If
          not, go to the next step.

        o The proxy responds to the RTSP request with a 470 or 407
          response code. The 407 response code MAY be used when the
          proxy requires both user and connection authorization from
          user or client. In this message the proxy SHALL include a
          Connection-Credentials header, see section 14.12 with the next
          hop's identity and certificate.

   The client MUST upon receiving a 470 or 407 response with
   Connection-Credentials header take the decision on whether to accept
   the certificate or not (if it cannot do so, the user SHOULD be
   consulted). If the certificate is accepted, the client has to again
   send the RTSP request. In that request the client has to include the
   Accept-Credentials header including the hash over the DER encoded
   certificate for all trusted proxies in the chain.


   Example:
   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP ;unicast ;client_port=4588-4589



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   P->C: RTSP/1.0 470 Connection Authorization Required
         CSeq: 2
         Connection-Credentials: "rtsps://test.example.org";
         MIIDNTCCAp...

   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP ;unicast ;client_port=4588-4589
         Accept-Credentials: User "rtsps://test.example.org" ;
         dPYD 7txp oGTb AqZZ QJ+v aeOk yH4= ...



   One implication of this process is that the connection for secured
   RTSP messages may take significantly more round-trip times for the
   first message. An complete extra message exchange between the proxy
   connecting to the next hop and the client results because of the
   process for approval for each hop. However after the first message
   exchange the remaining message should not be delayed, if each message
   contains the chain of proxies that the requestor accepts. The
   procedure of including the credentials in each request rather than
   building state in each proxy, avoids the need for revocation
   procedures.

18 Syntax

   The RTSP syntax is described in an augmented Backus-Naur Form (BNF)
   as defined in RFC 2234 [5].

18.1 Base Syntax


   OCTET      =  %x00-FF ; any 8-bit sequence of data
   CHAR       =  %x01-7F ; any US-ASCII character (octets 1 - 127)
   UPALPHA    =  %x41-5A ; any US-ASCII uppercase letter "A".."Z"
   LOALPHA    =  %x61-7A ;any US-ASCII lowercase letter "a".."z"
   ALPHA      =  UPALPHA / LOALPHA
   DIGIT      =  %x30-39 ; any US-ASCII digit "0".."9"
   CTL        =  %x00-1F / %x7F ; any US-ASCII control character
                 (octets 0 - 31) and DEL (127)>
   CR         =  %x0D ; US-ASCII CR, carriage return (13)>
   LF         =  %x0A ; US-ASCII LF, linefeed (10)>
   SP         =  %x20 ; US-ASCII SP, space (32)>
   HT         =  %x09 ; US-ASCII HT, horizontal-tab (9)>
   DQUOTE     =  %x22 ; US-ASCII double-quote mark (34)>
   BACKSLASH  =  %x5C ; US-ASCII backslash (92)>
   CRLF       =  CR LF




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   LWS            =  [CRLF] 1*( SP / HT )
   TEXT           =  %x20-7D / %x80-FF ; any OCTET except CTLs>
   tspecials      =  "(" / ")" / "<" / ">" / "@"
                  /  "," / ";" / ":" / BACKSLASH / DQUOTE
                  /  "/" / "[" / "]" / "?" / "="
                  /  "{" / "}" / SP / HT
   token          =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                  /  %x41-5A / %x5E-7A / %x7C / %x7E)
                     ; 1*<any CHAR except CTLs or tspecials>
   quoted-string  =  ( DQUOTE *(qdtext) DQUOTE )
   qdtext         =  %x20-21 / %x23-7D / %x80-FF ; any TEXT except <">
   quoted-pair    =  BACKSLASH CHAR



   safe         =  "$" / "-" / "_" / "." / "+"
   extra        =  "!" / "*" / "'" / "(" / ")" / ","
   hex          =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /
                   "a" / "b" / "c" / "d" / "e" / "f"
   escape       =  "%" hex hex
   reserved     =  ";" / "/" / "?" / ":" / "@" / "&" / "="
   unreserved   =  alpha / digit / safe / extra
   xchar        =  unreserved / reserved / escape
   base64       =  0*base64-unit [base64-pad]
   base64-unit  =  4base64-char
   base64-pad   =  (2base64-char "==") / (3base64-char "=")
   base64-char  =  ALPHA / DIGIT / "+" / "/"


18.2 RTSP Protocol Definition

18.2.1 Generic Protocol elements


   absoluteURL  =  < as defined in RFC 2396  [13] and RFC2732  [12] >
   relativeURL  =  < as defined in RFC 2396  [13] and RFC2732  [12] >
   rtsp-URI     =  rtsp-scheme "//" host [":" port]
                   [abs-path ["?" query]] ["#" fragment]
   rtsp-scheme  = ( "rtsp:" / "rtspu:" / "rtsps:" )
   host         =  As defined by RFC 2732 [12]
   abs-path     =  As defined by RFC 2396 [13]
   port         =  *DIGIT ; Is expected to be 1*5DIGIT
   query        =  As defined by RFC 2396 [13]
   fragment     =  As defined by RFC 2396 [13]



   smpte-range           =  smpte-type "=" smpte-range-spec



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                            ;Section 3.4
   smpte-range-spec      =  ( smpte-time "-" [ smpte-time ] )
                         /  ( "-" smpte-time )
   smpte-type            =  "smpte" / "smpte-30-drop"
                         /  "smpte-25" / smpte-type-extension
                            ; other timecodes may be added
   smpte-type-extension  =  token
   smpte-time            =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                            [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]



   npt-range       =  ["npt" "="] npt-range-spec ; Section 3.5
                      ; implementations SHOULD use npt= prefix,
                      ;but SHOULD be prepared to interoperate with
                      ; RFC 2326 implementations which don't use it.
   npt-range-spec  =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
   npt-time        =  "now" / npt-sec / npt-hhmmss
   npt-sec         =  1*DIGIT [ "." *DIGIT ]
   npt-hhmmss      =  npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
   npt-hh          =  1*DIGIT ; any positive number
   npt-mm          =  1*2DIGIT ; 0-59
   npt-ss          =  1*2DIGIT ; 0-59



   utc-range       =  "clock" "=" utc-range-spec ; Section 3.6
   utc-range-spec  =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
   utc-time        =  utc-date "T" utc-clock "Z"
   utc-date        =  8DIGIT ; < YYYYMMDD >
   utc-clock       =  6DIGIT [ "." fraction ]; < HHMMSS.fraction >
   fraction        =  1*DIGIT



   feature-tag     =  token
   session-id      =  8*( ALPHA / DIGIT / safe )
   message-header  =  field-name ":" [ field-value ] CRLF
   field-name      =  token
   field-value     =  *( field-content / LWS )
   field-content   =  <the OCTETs making up the field-value and
                      consisting of either *TEXT or combinations
                      of token, tspecials, and quoted-string>


18.2.2 Message Syntax





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        RTSP-message  =   Request / Response ; RTSP/1.0 messages
        Request       =   Request-Line        ; Section 6.1
                      *(  general-header      ; Section 5
                      /   request-header      ; Section 6.2
                      /   entity-header )     ; Section 8.1
                          CRLF
                          [ message-body ]    ; Section 4.3
        Response      =   Status-Line         ; Section 7.1
                      *(  general-header      ; Section 5
                      /   response-header     ; Section 7.1.2
                      /   entity-header )     ; Section 8.1
                          CRLF
                          [ message-body ]    ; Section 4.3



   Request-Line  =  Method SP Request-URI SP RTSP-Version CRLF
   Status-Line   =  RTSP-Version SP Status-Code SP Reason-Phrase CRLF



   Method            =  "DESCRIBE"        ; Section 11.2
                     /  "GET_PARAMETER"   ; Section 11.7
                     /  "OPTIONS"         ; Section 11.1
                     /  "PAUSE"           ; Section 11.5
                     /  "PLAY"            ; Section 11.4
                     /  "PING"            ; Section 11.10
                     /  "REDIRECT"        ; Section 11.9
                     /  "SETUP"           ; Section 11.3
                     /  "SET_PARAMETER"   ; Section 11.8
                     /  "TEARDOWN"        ; Section 11.6
                     /  extension-method
   extension-method  =  token



   Request-URI   =  "*" / absolute-URL
   RTSP-Version  =  "RTSP" "/" 1*DIGIT "." 1*DIGIT




        Status-Code  =  "100" ; Continue
                     /  "200" ; OK
                     /  "201" ; Created
                     /  "250" ; Low on Storage Space
                     /  "300" ; Multiple Choices
                     /  "301" ; Moved Permanently



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                     /  "302" ; Moved Temporarily
                     /  "303" ; See Other
                     /  "304" ; Not Modified
                     /  "305" ; Use Proxy
                     /  "400" ; Bad Request
                     /  "401" ; Unauthorized
                     /  "402" ; Payment Required
                     /  "403" ; Forbidden
                     /  "404" ; Not Found
                     /  "405" ; Method Not Allowed
                     /  "406" ; Not Acceptable
                     /  "407" ; Proxy Authentication Required
                     /  "408" ; Request Time-out
                     /  "410" ; Gone
                     /  "411" ; Length Required
                     /  "412" ; Precondition Failed
                     /  "413" ; Request Entity Too Large
                     /  "414" ; Request-URI Too Large
                     /  "415" ; Unsupported Media Type
                     /  "451" ; Parameter Not Understood
                     /  "452" ; reserved
                     /  "453" ; Not Enough Bandwidth
                     /  "454" ; Session Not Found
                     /  "455" ; Method Not Valid in This State
                     /  "456" ; Header Field Not Valid for Resource
                     /  "457" ; Invalid Range
                     /  "458" ; Parameter Is Read-Only
                     /  "459" ; Aggregate operation not allowed
                     /  "460" ; Only aggregate operation allowed
                     /  "461" ; Unsupported transport
                     /  "462" ; Destination unreachable
                     /  "470" ; Connection Authorization Required
                     /  "471" ; Connection Credentials not accepted
                     /  "500" ; Internal Server Error
                     /  "501" ; Not Implemented
                     /  "502" ; Bad Gateway
                     /  "503" ; Service Unavailable
                     /  "504" ; Gateway Time-out
                     /  "505" ; RTSP Version not supported
                     /  "551" ; Option not supported
                     /  extension-code



        extension-code  =  3DIGIT
        Reason-Phrase   =  *<TEXT, excluding CR, LF>





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   general-header  =  Cache-Control     ; Section 14.10
                   /  Connection        ; Section 14.11
                   /  CSeq              ; Section 14.19
                   /  Date              ; Section 14.20
                   /  Proxy-Supported   ; Section 14.32
                   /  Supported         ; Section 14.43
                   /  Timestamp         ; Section 14.44
                   /  Via               ; Section 14.49
                   /  extension-header



   request-header  =  Accept              ; Section 14.1 and [H14.1]
                   /  Accept-Credentials  ; Section 14.2
                   /  Accept-Encoding     ; Section 14.3 and [H14.3]
                   /  Accept-Language     ; Section 14.4 and [H14.4]
                   /  Authorization       ; Section 14.7 and [H14.8]
                   /  Bandwidth           ; Section 14.8
                   /  Blocksize           ; Section 14.9
                   /  From                ; Section 14.23
                   /  If-Match            ; Section 14.25
                   /  If-Modified-Since   ; Section 14.26 and [H14.25]
                   /  If-None-Match       ; Section 14.27
                   /  Proxy-Require       ; Section 14.31
                   /  Range               ; Section 14.34
                   /  Referer             ; Section 14.35
                   /  Require             ; Section 14.37
                   /  Scale               ; Section 14.39
                   /  Session             ; Section 14.42
                   /  Speed               ; Section 14.40
                   /  Supported           ; Section 14.43
                   /  Transport           ; Section 14.45
                   /  User-Agent          ; Section 14.47
                   /  extension-header



   response-header  =  Accept-Credentials  ; Section 14.2
                    /  Accept-Ranges       ; Section 14.5
                    /  Connection-Creds    ; Section 14.12
                    /  ETag                ; Section 14.21
                    /  Location            ; Section 14.29
                    /  Proxy-Authenticate  ; Section 14.30
                    /  Public              ; Section 14.33
                    /  Range               ; Section 14.34
                    /  Retry-After         ; Section 14.36
                    /  RTP-Info            ; Section 14.38
                    /  Scale               ; Section 14.39
                    /  Session             ; Section 14.42


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                    /  Server              ; Section 14.41
                    /  Speed               ; Section 14.40
                    /  Transport           ; Section 14.45
                    /  Unsupported         ; Section 14.46
                    /  Vary                ; Section 14.48
                    /  WWW-Authenticate    ; Section 14.50
                    /  extension-header



   entity-header     =  Allow             ; Section 14.6
                     /  Content-Base      ; Section 14.13
                     /  Content-Encoding  ; Section 14.14
                     /  Content-Language  ; Section 14.15
                     /  Content-Length    ; Section 14.16
                     /  Content-Location  ; Section 14.17
                     /  Content-Type      ; Section 14.18
                     /  Expires           ; Section 14.22 and [H14.21]
                     /  Last-Modified     ; Section 14.28
                     /  extension-header
   extension-header  =  message-header


18.2.3 Header Syntax

   All header syntaxes not defined in this section are defined in
   section 14 of the HTTP 1.1 specification [4].


   accept-credentials   =  "Accept-Credentials" ":" credential-decision
   credential-decision  =  ("User" "," [credential-info])
                        /  "Proxy"
                        /  "Any"
                        /  token ; For future extensions
   credential-info      =  cred-info-data 0*("," cred-info-data)
   cred-info-data       =  DQUOTE rtsp-URI DQUOTE ";" base64
   Accept-Ranges        =  "Accept-Ranges" ":" acceptable-ranges
   acceptable-ranges    =  (range-unit *("," LWS range-unit))
                        /  "none"
   range-unit           =  NPT / SMPTE / UTC / extension-format
   extension-format     =  token
   Bandwidth            =  "Bandwidth" ":" 1*DIGIT
   Blocksize            =  "Blocksize" ":" 1*DIGIT



   Cache-Control             =  "Cache-Control" ":" cache-directive
                                *("," LWS cache-directive)



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   cache-directive           =  cache-request-directive
                             /  cache-response-directive
   cache-request-directive   =  "no-cache"
                             /  "max-stale" ["=" delta-seconds]
                             /  "min-fresh" "=" delta-seconds
                             /  "only-if-cached"
                             /  cache-extension
   cache-response-directive  =  "public"
                             /  "private"
                             /  "no-cache"
                             /  "no-transform"
                             /  "must-revalidate"
                             /  "proxy-revalidate"
                             /  "max-age" "=" delta-seconds
                             /  cache-extension
   cache-extension           =  token ["=" (token / quoted-string)]
   delta-seconds             =  1*DIGIT



   connection-creds  =  "Connection-Credentials" ":" credential-info
   connection        =  "Connection" ":" (connection-token)
                        *("," connection-token)
   connection-token  =  token
   Content-Base      =  "Content-Base" ":" absoluteURL
   CSeq              =  "Cseq" ":" 1*DIGIT
   Proxy-Require     =  "Proxy-Require" ":" feature-tag
                        *("," LWS feature-tag)
   Proxy-Supported   =  "Proxy-Supported" ":" feature-tag
                        *("," LWS feature-tag)
   Public            =  "Public" ":" method *("," LWS method)
   Range             =  "Range" ":" ranges-spec *("," LWS ranges-spec)
                        [ ";" "time" "=" utc-time ]
   ranges-spec       =  npt-range / utc-range / smpte-range
   Require           =  "Require" ":" feature-tag *("," LWS feature-tag)



   RTP-Info        =  "RTP-Info" ":" rtsp-info-spec
                      *("," LWS rtsp-info-spec)
   rtsp-info-spec  =  stream-url 1*ri-parameter
   stream-url      =  quoted-url / unquoted-url
   unquoted-url    =  "url" "=" safe-url
   quoted-url      =  "url" "=" DQUOTE needquote-url DQUOTE
   safe-url        =  url
   needquote-url   =  url //That contains ; or ,
   url             =  ( absoluteURL / relativeURL )
   ri-parameter    =  ";" "seq" "=" 1*DIGIT
                   /  ";" "rtptime" "=" 1*DIGIT


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   Scale      =  "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
   Speed      =  "Speed" ":" 1*DIGIT [ "." *DIGIT ]
   Server     =  "Server" ":" ( product / comment )
                 *(SP (product / comment))
   Session    =  "Session" ":" session-id
                 [ ";" "timeout" "=" delta-seconds ]
   Supported  =  "Supported" ":" [feature-tag *("," LWS feature-tag)]



   Timestamp       =  "Timestamp" ":" *(DIGIT) ["." *(DIGIT)] [delay]
   delay           =  *(DIGIT) [ "." *(DIGIT) ]
   Transport       =  "Transport" ":" transport-spec
                      *("," LWS transport-spec)
   transport-spec  =  transport-id *tr-parameter
   transport-id    =  transport-prot "/" profile ["/" lower-transport]
                      ; no LWS is allowed inside transport-id



   transport-prot    =  "RTP" / token
   profile           =  "AVP" / token
   lower-transport   =  "TCP" / "UDP" / token
   tr-parameter      =  ";" ( "unicast" / "multicast" )
                    /   ";" "source" "=" host
                    /   ";" "destination" [ "=" host ]
                    /   ";" "interleaved" "=" channel [ "-" channel ]
                    /   ";" "append"
                    /   ";" "ttl" "=" ttl
                    /   ";" "layers" "=" 1*DIGIT
                    /   ";" "port" "=" port-spec
                    /   ";" "client_port" "=" port-spec
                    /   ";" "server_port" "=" port-spec
                    /   ";" "ssrc" "=" ssrc *("/" ssrc)
                    /   ";" "client_ssrc" "=" ssrc
                    /   ";" "mode" "=" mode-spec
                    /   ";" "dest_addr" "=" addr-list
                    /   ";" "src_addr" "=" addr-list
                    /   ";" trn-param-ext
   port-spec         =  port [ "-" port ]
   trn-param-ext     =  par-name "=" trn-par-value
   par-name          =  token
   trn-par-value     =  *(unreserved / DQUOTE *TEXT DQUOTE)
   ttl               =  1*3(DIGIT)
   ssrc              =  8*8(HEX)
   channel           =  1*3(DIGIT)
   mode-spec         =  ( DQUOTE mode *("," *SP mode) DQUOTE ) / mode
   mode              =  "PLAY" / "RECORD" / token



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   addr-list         =  quoted-host-port *("/" quoted-host-port)
   quoted-host-port  =  DQUOTE host [":" port] DQUOTE



   Unsupported  =  "Unsupported" ":" feature-tag *("," feature-tag)
   User-Agent   =  "User-Agent" ":" ( product / comment )
                   0*(SP (product / comment)


19 Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

        Abuse of Server Log Information: RTSP and HTTP servers will
             presumably have similar logging mechanisms, and thus should
             be equally guarded in protecting the contents of those
             logs, thus protecting the privacy of the users of the
             servers. See [H15.1.1] for HTTP server recommendations
             regarding server logs.

        Transfer of Sensitive Information: There is no reason to believe
             that information transferred via RTSP may be any less
             sensitive than that normally transmitted via HTTP.
             Therefore, all of the precautions regarding the protection
             of data privacy and user privacy apply to implementors of
             RTSP clients, servers, and proxies. See [H15.1.2] for
             further details.

        Attacks Based On File and Path Names: Though RTSP URIs are
             opaque handles that do not necessarily have file system
             semantics, it is anticipated that many implementations will
             translate portions of the Request-URIs directly to file
             system calls. In such cases, file systems SHOULD follow the
             precautions outlined in [H15.5], such as checking for ".."
             in path components.


        Personal Information: RTSP clients are often privy to the same
             information that HTTP clients are (user name, location,
             etc.)  and thus should be equally sensitive. See [H15.1]
             for further recommendations.

        Privacy Issues Connected to Accept Headers: Since may of the
             same "Accept" headers exist in RTSP as in HTTP, the same
             caveats outlined in [H15.1.4] with regards to their use



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             should be followed.

        DNS Spoofing: Presumably, given the longer connection times
             typically associated to RTSP sessions relative to HTTP
             sessions, RTSP client DNS optimizations should be less
             prevalent.  Nonetheless, the recommendations provided in
             [H15.3] are still relevant to any implementation which
             attempts to rely on a DNS-to-IP mapping to hold beyond a
             single use of the mapping.


        Location Headers and Spoofing: If a single server supports
             multiple organizations that do not trust each another, then
             it needs to check the values of Location and Content-
             Location header fields in responses that are generated
             under control of said organizations to make sure that they
             do not attempt to invalidate resources over which they have
             no authority. ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [4], as of this writing) and also of the previous RFC2068
   [19], future HTTP specifications may provide additional guidance on
   security issues.

   The following are added considerations for RTSP implementations.

        Concentrated denial-of-service attack: The protocol offers the
             opportunity for a remote-controlled denial-of-service
             attack.

             The attacker may initiate traffic flows to one or more IP
             addresses by specifying them as the destination in SETUP
             requests. While the attacker's IP address may be known in
             this case, this is not always useful in prevention of more
             attacks or ascertaining the attackers identity. Thus, an
             RTSP server SHOULD only allow client-specified destinations
             for RTSP-initiated traffic flows if the server has verified
             the client's identity, either against a database of known
             users using RTSP authentication mechanisms (preferably
             digest authentication or stronger), or other secure means.

        Session hijacking: Since there is no or little relation between
             a transport layer connection and an RTSP session, it is
             possible for a malicious client to issue requests with
             random session identifiers which would affect unsuspecting
             clients. The server SHOULD use a large, random and non-
             sequential session identifier to minimize the possibility
             of this kind of attack.



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        Authentication: Servers SHOULD implement both basic and digest
             [8] authentication. In environments requiring tighter
             security for the control messages, the transport layer
             mechanism TLS (RFC 2246 [7]) SHOULD be used.

        Stream issues: RTSP only provides for stream control. Stream
             delivery issues are not covered in this section, nor in the
             rest of this draft. RTSP implementations will most likely
             rely on other protocols such as RTP, IP multicast, RSVP and
             IGMP, and should address security considerations brought up
             in those and other applicable specifications.

        Persistently suspicious behavior: RTSP servers SHOULD return
             error code 403 (Forbidden) upon receiving a single instance
             of behavior which is deemed a security risk. RTSP servers
             SHOULD also be aware of attempts to probe the server for
             weaknesses and entry points and MAY arbitrarily disconnect
             and ignore further requests clients which are deemed to be
             in violation of local security policy.

20 IANA Considerations

   This section set up a number of registers for RTSP that should be
   maintained by IANA. For each registry there is a description on what
   it is required to contain, what specification is needed when adding a
   entry with IANA, and finally the entries that this document needs to
   register. See also the section 1.6 "Extending RTSP". There is also an
   IANA registration of two SDP attributes.

   The sections describing how to register an item uses some of the
   requirements level described in RFC 2434 [20], namely " First Come,
   First Served", "Specification Required", and "Standards Action".

   A registration request to IANA MUST contain the following
   information:

        o A name of the item to register according to the rules
          specified by the intended registry.

        o Indication of who has change control over the feature (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium, a particular company or group of
          companies, or an individual);

        o A reference to a further description, if available, for
          example (in order of preference) an RFC, a published standard,
          a published paper, a patent filing, a technical report,
          documented source code or a computer manual;



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        o For proprietary features, contact information (postal and
          email address);

20.1 Feature-tags

20.1.1 Description

   When a client and server try to determine what part and functionality
   of the RTSP specification and any future extensions that its counter
   part implements there is need for a namespace.  This registry
   contains named entries representing certain functionality.

   The usage of feature-tags is explained in section 10 and 11.1.

20.1.2 Registering New Feature-tags with IANA

   The registering of feature-tags is done on a first come, first served
   basis.

   The name of the feature MUST follow these rules: The name may be of
   any length, but SHOULD be no more than twenty characters long. The
   name MUST not contain any spaces, or control characters.  The
   registration SHALL indicate if the feature tag applies to servers
   only, proxies only or both server and proxies. Any proprietary
   feature SHALL have as the first part of the name a vendor tag, which
   identifies the organization.

20.1.3 Registered entries

   The following feature-tags are in this specification defined and
   hereby registered. The change control belongs to the Authors and the
   IETF MMUSIC WG.

        play.basic: The minimal implementation for playback operations
             according to section D. Applies for both servers and
             proxies.

        play.scale: Support of scale operations for media playback.
             Applies only for servers.

        play.speed: Support of the speed functionality for playback.
             Applies only for servers

20.2 RTSP Methods

20.2.1 Description

   What a method is, is described in section 11.  Extending the protocol



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   with new methods allow for totally new functionality.

20.2.2 Registering New Methods with IANA

   A new method MUST be registered through an IETF standard track
   document. The reason is that new methods may radically change the
   protocols behavior and purpose.

   A specification for a new RTSP method MUST consist of the following
   items:

        o A method name which follows the BNF rules for methods.

        o A clear specification on what action and response a request
          with the method will result in. Which directions the method is
          used, C -> S or S -> C or both. How the use of headers, if
          any, modifies the behavior and effect of the method.

        o A list or table specifying which of the registered headers
          that are allowed to use with the method in request or/and
          response.

        o Describe how the method relates to network proxies.

20.2.3 Registered Entries

   This specification, RFCXXXX, registers 10 methods: DESCRIBE,
   GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, REDIRECT, SETUP,
   SET_PARAMETER, and TEARDOWN.

20.3 RTSP Status Codes

20.3.1 Description

   A status code is the three digit numbers used to convey information
   in RTSP response messages, see  7.  The number space is limited and
   care should be taken not to fill the space.

20.3.2 Registering New Status Codes with IANA

   A new status code can only be registered by an IETF standards track
   document. A specification for a new status code MUST specify the
   following:

        o The requested number.

        o A description what the status code means and the expected
          behavior of the sender and receiver of the code.



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20.3.3 Registered Entries

   RFCXXX, registers the numbered status code defined in the BNF entry
   "Status-Code" except "extension-code" in section 18.2.2.

20.4 RTSP Headers

20.4.1 Description

   By specifying new headers a method(s) can be enhanced in many
   different ways. An unknown header will be ignored by the receiving
   entity. If the new header is vital for a certain functionality, a
   feature-tag for the functionality can be created and demanded to be
   used by the counter-part with the inclusion of a Require header
   carrying the feature-tag.

20.4.2 Registering New Headers with IANA

   A public available specification is required to register a header.
   The specification SHOULD be a standards document, preferable an IETF
   RFC.

   The specification MUST contain the following information:

        o The name of the header.

        o A BNF specification of the header syntax.

        o A list or table specifying when the header may be used,
          encompassing all methods, their request or response, the
          direction (C -> S or S -> C).

        o How the header is to be handled by proxies.

        o A description of the purpose of the header.

20.4.3 Registered entries

   All headers specified in section 14 in RFCXXXX are to be registered.

   Furthermore the following RTSP headers defined in other
   specifications are registered:

        o x-wap-profile defined in [37].

        o x-wap-profile-diff defined in [37].

        o x-wap-profile-warning defined in [37].



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        o x-predecbufsize defined in [37].

        o x-initpredecbufperiod defined in [37].

        o x-initpostdecbufperiod defined in [37].

   The use of "X-" is NOT RECOMMENDED but the above headers in the
   register list was defined prior to the clarification.

20.5 Transport Header registries

   The transport header contains a number of parameters which have
   possibilities for future extensions. Therefore registries for these
   needs to be defined.

20.5.1 Transport Protocols

   A registry for the parameter transport-protocol SHALL be defined with
   the following rules:

        o Registering require an public available standards
          specification.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 1 value:

        o Use of the RTP  [17] protocol for media transport. The usage
          is explained in RFC XXXX, appendix B.1.

20.5.2 Profile

   A registry for the parameter profile SHALL be defined with the
   following rules:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.




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        o A definition of which Transport protocol(s) that this profile
          is valid for.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 1 value:

        o The "RTP profile for audio and video conferences with minimal
          control"  [3] MUST only be used when the transport
          specification's transport-protocol is "RTP".

20.5.3 Lower Transport

   A registry for the parameter lower-transport SHALL be defined with
   the following rules:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A text describing how the registered value are used in RTSP.

   This specification registers 2 values:

        UDP: Indicates the use of the "User datagram protocol"  [9] for
             media transport.

        TCP: Indicates the use Transmission control protocol  [10] for
             media transport.

20.5.4 Transport modes

   A registry for the transport parameter mode SHALL be defined with the
   following rules:

        o Registering requires an IETF standard tracks document.

        o A contact person or organization with address and email.

        o A value definition that are following the BNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.



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   This specification registers 2 values:

        PLAY: See RFC XXXX.

        RECORD: See RFC XXXX.

20.6 Cache Directive Extensions

   There exist a number of cache directives which can be sent in the
   Cache-Control header. A registry for this cache directives SHALL be
   defined with the following rules:

        o Registering requires an IETF standard tracks document.

        o A registration is required to contain a contact person.

        o Name of the directive and a definition of the value, if any.

        o Specification if it is an request or response directive.

        o A describing text that explains how the cache directive is
          used for RTSP controlled media streams.

   This specification registers the following values:

        no-cache:

        public:

        private:

        no-transform:

        only-if-cached:

        max-stale:

        min-fresh:

        must-revalidate:

        proxy-revalidate:

        max-age:

20.7 Accept-Credentials policies

   In section  17.3.1 three policies for how to handle certificates.



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   Further policies may be defined and SHALL be registered with IANA
   using the following rules:

        o Registering requires an IETF standard tracks document.

        o A registration is required name a contact person.

        o Name of the policy.

        o A describing text that explains how the policy works for
          handling the certificates.

   This specification registers the following values:

        Any

        Proxy

        User

20.8 URI Schemes

   This specification defines two URI schemes ("rtsp" and "rtsps") and
   reserves a third one ("rtspu").

   This will need to be done in accordance with RFC 2717.

20.9 SDP attributes

   This specification defines two SDP [2] attributes that it is
   requested that IANA register.



   SDP Attribute ("att-field"):

        Attribute name:     range
        Long form:          Media Range Attribute
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition.

        Attribute name:     control
        Long form:          RTSP control URI
        Type of name:       att-field



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        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             Absolute or Relative URIs.

        Attribute name:     etag
        Long form:          Entity Tag
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition





A RTSP Protocol State Machine

   The RTSP session state machine describes the behavior of the protocol
   from RTSP session initialization through RTSP session termination.

   The State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier. The session may contain
   one or more media streams depending on state. If a single media
   stream is part of the session it is in non-aggregated control. If two
   or more is part of the session it is in aggregated control.

   The below state machine is a normative description of the protocols
   behavior. However, in case of ambiguity with the earlier parts of
   this specification, the description in the earlier parts SHALL take
   precedence.

A.1 States

   The state machine contains three states, described below. For each
   state there exist a table which shows which requests and events that
   is allowed and if they will result in a state change.

        Init: Initial state no session exist.

        Ready: Session is ready to start playing.

        Play: Session is playing, i.e. sending media stream data in the
             direction S -> C.




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A.2 State variables

   This representation of the state machine needs more than its state to
   work. A small number of variables are also needed and is explained
   below.

        NRM: The number of media streams part of this session.

        RP: Resume point, the point in the presentation time line at
             which a request to continue will resume from. A time format
             for the variable is not mandated.

A.3 Abbreviations

   To make the state tables more compact a number of abbreviations are
   used, which are explained below.

        IFI: IF Implemented.

        md: Media

        PP: Pause Point, the point in the presentation time line at
             which the presentation was paused.

        Prs: Presentation, the complete multimedia presentation.

        RedP: Redirect Point, the point in the presentation time line at
             which a REDIRECT was specified to occur.

        SES: Session.

A.4 State Tables

   This section contains a table for each state. The table contains all
   the requests and events that this state is allowed to act on.  The
   events which is method names are, unless noted, requests with the
   given method in the direction client to server (C -> S). In some
   cases there exist one or more requisite. The response column tells
   what type of response actions should be performed. Possible actions
   that is requested for an event includes: response codes, e.g. 200,
   headers that MUST be included in the response, setting of state
   variables, or setting of other session related parameters. The new
   state column tells which state the state machine changes to.

   The response to valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column. The
   exceptions needs to be given a response according to the response
   column. If the request does not meet the requisite, is erroneous or



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   some other type of error occur the appropriate response code MUST be
   sent. If the response code is a 4xx the session state is unchanged. A
   response code of 3rr will result in that the session is ended and its
   state is changed to Init. A response code of 304 results in no state
   change. However there exist restrictions to when a 3xx response may
   be used. A 5xx response SHALL not result in any change of the session
   state, except if the error is not possible to recover from. A
   unrecoverable error SHALL result the ending of the session. As it in
   the general case can't be determined if it was a unrecoverable error
   or not the client will be required to test. In the case that the next
   request after a 5xx is responded with 454 (Session Not Found) the
   client knows that the session has ended.

   The server will timeout the session after the period of time
   specified in the SETUP response, if no activity from the client is
   detected.  Therefore there exist a timeout event for all states
   except Init.

   In the case that NRM=1 the presentation URI is equal to the media
   URI. For NRM>1 the presentation URI MUST be other than any of the
   medias that are part of the session. This applies to all states.





   Event         Prerequisite    Response
   ______________________________________________________________
   DESCRIBE      Needs REDIRECT  3rr Redirect
   DESCRIBE                      200, Session description
   OPTIONS       Session ID      200, Reset session timeout timer
   OPTIONS                       200
   SET_PARAMETER Valid parameter 200, change value of parameter
   GET_PARAMETER Valid parameter 200, return value of parameter


   Table 13: None state-machine changing events


   The methods in Table 13 do not have any effect on the state machine
   or the state variables. However some methods do change other session
   related parameters, for example SET_PARAMETER which will set the
   parameter(s) specified in its body.


   The initial state of the state machine, see Table 14 can only be left
   by processing a correct SETUP request. As seen in the table the two
   state variables are also set by a correct request. This table also
   shows that a correct SETUP can in some cases be redirected to another


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       Action           Requisite       New State  Response
       _____________________________________________________________
       SETUP                              Ready    NRM=1, RP=0.0
       SETUP            Needs Redirect    Init     3rr Redirect
       S -> C:REDIRECT  No Session hdr    Init     Terminate all SES


   Table 14: State: Init

   URI and/or server by a 3rr response.


   Action           Requisite          New State  Response

_____________________________________________________________________
   SETUP            New URI              Ready    NRM+=1
   SETUP            Setten up URI        Ready    Change transport param
   TEARDOWN         Prs URI,NRM>1        Init     No session hdr
   TEARDOWN         md URI,NRM=1         Init     No Session hdr, NRM=0
   TEARDOWN         md URI,NRM>1         Ready    Session hdr, NRM-=1
   PLAY             Prs URI, No range    Play     Play from RP
   PLAY             Prs URI, Range       Play     according to range
   PAUSE            Prs URI              Ready    Return PP
   S -> C:REDIRECT  Range hdr            Ready    Set RedP
   S -> C:REDIRECT  no range hdr         Init     Session is removed
   Timeout                               Init
   RedP reached                          Ready    TEARDOWN of session


   Table 15: State: Ready


   In the Ready state, see Table 15, some of the actions are depending
   on the number of media streams (NRM) in the session, i.e. aggregated
   or non-aggregated control. A setup request in the ready state can
   either add one more media stream to the session or if the media
   stream (same URI) already is part of the session change the transport
   parameters. TEARDOWN is depending on both the Request-URI and the
   number of media stream within the session. If the Request-URI is the
   presentations URI the whole session is torn down. If a media URI is
   used in the TEARDOWN request and more than one media exist in the
   session, the session will remain and a session header MUST be
   returned in the response. If only a single media stream remains in
   the session when performing a TEARDOWN with a media URI the session
   is removed. The number of media streams remaining after tearing down
   a media stream determines the new state.


   The Play state table, see Table 16, is the largest. The table



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   Action           Requisite         New State  Response
   _____________________________________________________________________
   PAUSE            PrsURI,No range     Ready    Set RP to present point
   PAUSE            PrsURI,Range>now    Play     Set RP & PP to given p.
   PAUSE            PrsURI,Range<now    Ready    Set RP to Range Hdr.
   PP reached                           Ready    RP = PP
   End of media     All media           Play     No action, RP = Invalid
   End of media     >1 Media plays      Play     No action
   End of range                         Play     Set RP = End of range
   SETUP            New URI             Play     455
   SETUP            Setuped URI         Play     455
   SETUP            Setuped URI, IFI    Play     Change transport param.
   TEARDOWN         Prs URI,NRM>1       Init     No session hdr
   TEARDOWN         md URI,NRM=1        Init     No Session hdr, NRM=0
   TEARDOWN         md URI              Play     455
   S -> C:REDIRECT  Range hdr           Play     Set RedP
   S -> C:REDIRECT  no range hdr        Init     Session is removed
   RedP reached                         Play     TEARDOWN of session
   Timeout                              Init     Stop Media playout


   Table 16: State: Play

   contains an number of requests that has presentation URI as a
   prerequisite on the Request-URI, this is due to the exclusion of
   non-aggregated stream control in sessions with more than one media
   stream.

   To avoid inconsistencies between the client and server, automatic
   state transitions are avoided. This can be seen at for example "End
   of media" event when all media has finished playing, the session
   still remain in Play state. An explicit PAUSE request MUST be sent to
   change the state to Ready. It may appear that there exist two
   automatic transitions in "RedP reached" and "PP reached", however
   they are requested and acknowledge before they take place. The time
   at which the transition will happen is known by looking at the range
   header. If the client sends request close in time to these
   transitions it needs to be prepared for getting error message as the
   state may or may not have changed.

B Media Transport Alternatives

   This section defines how certain combinations of protocols, profiles
   and lower transports are used. This includes the usage of the
   Transport header's general source and destination parameters
   "src_addr" and "dest_addr".




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B.1 RTP

   This section defines the interaction of RTSP with respect to the RTP
   protocol [17]. It also defines any necessary media transport
   signalling with regards to RTP.

   The available RTP profiles and lower layer transports are described
   below along with rules on signalling the available combinations.

B.1.1 AVP

   The usage of the "RTP Profile for Audio and Video Conferences with
   Minimal Control" [3] when using RTP for media transport over
   different lower layer transport protocols is defined below in regards
   to RTSP.

   One such case is defined within this document, the use of embedded
   (interleaved) binary data as defined in section  12.  The usage of
   this method is indicated by include the "interleaved" parameter.

   When using embedded binary data the "src_addr" and "dest_addr" SHALL
   NOT be used. This addressing and multiplexing is used as defined with
   use of channel numbers and the interleaved parameter.

B.1.2 AVP/UDP

   This part describes sending of RTP [17] over lower transport layer
   UDP [9] according to the profile "RTP Profile for Audio and Video
   Conferences with Minimal Control" defined in RFC 3551 [3]. This
   profiles requires one or two uni- or bi-directional UDP flows per
   media stream. The first UDP flow is for RTP and the second is for
   RTCP. Embedding of RTP data with the RTSP messages, in accordance
   with section 12, SHOULD NOT be performed when RTSP messages are
   transported over unreliable transport protocols, like UDP [9].

   The RTP/UDP and RTCP/UDP flows can be established in two ways using
   the Transport header's parameters. The way provided in RFC 2326 was
   to use the necessary parameters from the set of "source",
   "destination", "client_port", and "server_port". This has the
   advantage of being compatible with all RTP capable RTSP servers and
   clients. However this method does not provide the means to specify
   non-continues port ranges for RTP and RTCP. The other way is to use
   the parameters "src_addr", and "dest_addr".  This method provides
   total flexibility in specifying address and port number for each
   transport flow. However the disadvantage is that it is not supported
   by non-updated clients, i.e. clients not supporting the "play.basic"
   feature-tag.




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   When using the "source", "destination", "client_port", and
   "server_port" the packets are be addressed in the following way for
   media playback:

        o RTP/UDP packet from the server to the client SHALL be sent to
          the address specified in the "destination" parameter and first
          even port number given in client_port range. If only an RTP
          port is to be specified, then only that even port number SHALL
          be given, i.e. no range including an odd number SHALL be used.

        o The server SHOULD send its RTP/UDP packets from the address
          specified in "source" parameter and from the first even port
          number specified in "server_port" parameter.


        o When the range specified in the "client_port" parameter
          contains at least two port numbers, the RTCP/UDP packets from
          server to client SHALL be sent to the address specified in the
          "destination" parameter and using the first odd port number
          belonging to the range specified in the client_port parameter.

        o The Server SHOULD send its RTCP/UDP packets from the address
          specified in "source" parameter and from the first odd port
          number greater than the RTP port number specified in
          "server_port" parameter.

        o RTCP/UDP packets from the client to the server SHALL be sent
          to the address specified in the "source" parameter and first
          odd port number greater than the RTP port number given in
          server_port range.

        o The client SHOULD send its RTCP/UDP packets from the address
          specified in "destination" parameter and from the first odd
          port number specified in client_port" parameter.

   The usage of "src_addr" and "dest_addr" parameters to specify the
   address and port numbers is performed in the following way for media
   playback, i.e. Mode=PLAY:

        o The "src_addr" and "dest_addr" parameters MUST contain either
          1 or 2 address and port pairs.

        o Each address and port pair MUST contain both and address and a
          port number.

        o The first address and port pair given in either of the
          parameters applies to the RTP stream. The second address and
          port pair if present applies to the RTCP stream.



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        o The RTP/UDP packets from the server to the client SHALL be
          sent to the address and port given by first address and port
          pair of the "dest_addr" parameter.

        o The RTCP/UDP packets from the server to the client SHALL be
          sent to the address and port given by the second address and
          port pair of the "dest_addr" parameter. If no second pair is
          given RTCP SHALL NOT be sent.

        o The RTCP/UDP packets from the client to the server SHALL be
          sent to the address and port given by the second address and
          port pair of the "src_addr" parameter. If no second pair is
          given RTCP SHALL NOT be sent.

        o RTP and RTCP Packets SHOULD be sent from the corresponding
          receiver port, i.e. RTCP packets from server should be sent
          from the "src_addr" parameters second address port pair.

B.1.3 AVP/TCP

   Note that this combination is not yet defined using sperate TCP
   connections. However the use of embedded (interleaved) binary data
   transported on the RTSP connection is possible as specified in
   section  12. When using this declared combination of interleaved
   binary data the RTSP messages MUST be transported over TCP.


        A possible future for this profile would be to define the
        use of a combination of the two drafts "Connection-Oriented
        Media Transport in SDP" [38] and "Framing RTP and RTCP
        Packets over Connection-Oriented Transport" [39]. However
        as this work is not finished, this functionality is
        unspecified.

B.1.4 Handling NPT Jumps in the RTP Media Layer

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer[17]. Such control allows jumps to be created in NPT
   timeline of the RTSP session. For example, jumps in NPT can be caused
   by multiple ranges in the range specifier of a PLAY request or
   through a "seek" opertaion on an RTSP session which involves a PLAY,
   PAUSE, PLAY scenario where a new NPT is set for the session. The
   media layer rendering the RTP stream should not be affected by jumps
   in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be
   continuous and monotonic across jumps of NPT.





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        We cannot assume that the RTSP client can communicate with
        the RTP media agent, as the two may be independent
        processes.  If the RTP timestamp shows the same gap as the
        NPT, the media agent will assume that there is a pause in
        the presentation. If the jump in NPT is large enough, the
        RTP timestamp may roll over and the media agent may believe
        later packets to be duplicates of packets just played out.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.


      C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15

      S->C: RTSP/1.0 200 OK
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15
        RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0;
              rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s



   Immediately after the end of the play range, the client follows up
   with a request to PLAY from a new NPT.


   C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0
         CSeq: 5
         Session: abcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 5
         Session: abcdefg



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         Range: npt=15-15

   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20;

   S->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=18-20
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=50;
                   rtptime=40100



   The ensuing RTP data stream is depicted below:

      S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
      S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
       . . .
      S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s



   In this example, first, NPT 10 through 15 is played, then the client
   request the server to skip ahead and play NPT 18 through 20. The
   first segment is presented as RTP packets with sequence numbers 0
   through 49 and timestamp 0 through 39,200. The second segment
   consists of RTP packets with sequence number 50 through 69, with
   timestamps 40,100 through 55,200. While there is a gap in the NPT,
   there is no gap in the sequence number space of the RTP data stream.

   The RTP timestamp gap is present in the above example due to the time
   it takes to perform the second play request, in this case 12.5 ms
   (100/8000). To avoid this gap in playback due to the time it takes to
   perform RTSP requests, a PLAY request with multiple ranges needs to
   be specified. That would result in the following example:


      C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15;npt=18-20

      S->C: RTSP/1.0 200 OK
        CSeq: 4
        Session: abcdefg



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        Range: npt=10-15
        RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0;
              rtptime=0



   The ensuing RTP data stream is depicted below:



      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
      S -> C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
      S -> C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
       . . .
      S -> C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s





B.1.5 Handling RTP Timestamps after PAUSE

   During a PAUSE / PLAY interaction in an RTSP session, the duration of
   time for which the RTP transmission was halted MUST be reflected in
   the RTP timestamp of each RTP stream. The duration can be calculated
   for each RTP stream as the time elapsed from when the last RTP packet
   was sent before the PAUSE request was received and when the first RTP
   packet was sent after the subsequent PLAY request was received. The
   duration includes all latency incurred and processing time required
   to complete the request.


        The RTP RFC [17] states that: The RTP timestamp for each
        unit[packet] would be related to the wallclock time at
        which the unit becomes current on the virtual presentation
        timeline.

   In order to satisfy the requirements of [17], the RTP timestamp space
   needs to increase continuously with real time.  While this is not
   optimal for stored media, it is required for RTP and RTCP to function
   as intended. Using a continuous RTP timestamp space allows the same
   timestamp model for both stored and live media and allows better
   opportunity to integrate both types of media under a single control.

   As an example, assume a clock frequency of 8000 Hz, a packetization



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   interval of 100 ms and an initial sequence number and timestamp of
   zero.


   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15;

   S->C: RTSP/1.0 200 OK
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=0;
              rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s



   The client then sends a PAUSE request:


   C->S: PAUSE rtsp://xyz/fizzle RTSP/1.0
         CSeq: 5
         Session: abdcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=10.4-15



   20 seconds elapse and then the client sends a PLAY request. In
   addition the server requires 15 ms to process the request:


   C->S: PLAY rtsp://xyz/fizzle RTSP/1.0
         CSeq: 6



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         Session: abcdefg

   S->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=10.4-15
         RTP-Info: url= rtsp://xyz/fizzle/audiotrack;seq=4;
                 rtptime=164400



   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s



   First, NPT 10 through 10.3 is played, then a PAUSE is received by the
   server. After 20 seconds a PLAY is received by the server which take
   15ms to process. The duration of time for which the session was
   paused is reflected in the RTP timestamp of the RTP packets sent
   after this PLAY request.

   A client can use the RTSP range header and RTP-Info header to map NPT
   time of a presentation with the RTP timestamp.

   Note: In RFC 2326 [1], this matter was not clearly defined and was
   misunderstood commonly. Therefore, clients SHOULD expect servers to
   break the continuity of the RTP timestamp space in various arbitrary
   manners after a PAUSE request. In these cases, it is RECOMMENDED that
   clients accept the RTP stream after the pause with appropriate
   mappings provided by the RTP-Info and Range headers.

B.1.6 RTSP / RTP Integration

   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the above
   restrictions. Combined RTSP/RTP media clients should use the RTP-Info
   field to determine whether incoming RTP packets were sent before or
   after a seek or before or after a PAUSE.

B.1.7 Scaling with RTP

   For scaling (see Section 14.39), RTP timestamps should correspond to
   the playback timing. For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 14.40) of one, the



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   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.

        Note: The above scaling puts requirements on the media
        codec or a media stream to support it. For example motion
        JPEG or other non-predictive video coding can easier handle
        the above example.

B.1.8 Maintaining NPT synchronization with RTP timestamps

   The client can maintain a correct display of NPT by noting the RTP
   timestamp value of the first packet arriving after repositioning.
   The sequence parameter of the RTP-Info (Section 14.38) header
   provides the first sequence number of the next segment.

B.1.9 Continuous Audio

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request. This allows the client to
   perform playout delay adaptation.

B.1.10 Multiple Sources in an RTP Session

   Note that more than one SSRC MAY be sent in the media stream.
   However, without further extensions RTSP can't synchronize more than
   the single one indicated in the Transport header. In these cases RTCP
   needs to be used for synchronization.

B.1.11 Usage of SSRCs and the RTCP BYE Message During an RTSP Session

   The RTCP BYE message indicates the end of use of a given SSRC. If all
   sources leave an RTP session, it can, in most cases, be assumed to
   have ended. Therefore, a client or server SHALL NOT send a RTCP BYE
   message until it has finished using a SSRC. A server SHOULD keep
   using a SSRC until the RTP session is terminated. Prologing the use
   of a SSRC allows the established synchronization context associated
   with that SSRC to be used to sychronize subsequent PLAY requests even
   if the PLAY response is late. Additionally, changing the server side
   SSRC will prevent the server from synchronizing the new SSRC within
   RTSP as it is connected to the one declared in the ssrc parameter in
   the Transport header.

   An SSRC collision with the SSRC that transmits media does also have
   consequences, as it will force the media sender to change its SSRC in
   accordance with the RTP specification  [17].  This will result in a
   loss of synchronization context, and require any receiver to wait for
   RTCP sender reports for all media requiring synchronization before



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   being able to play out synchronized. Due to these reasons a client
   joining a session should take care to not select the same SSRC as the
   server. Any SSRC signalled in the Transport header SHOULD be avoided.
   Also a client detecting a collision prior to sending any RTP or RTCP
   messages can also select a new SSRC.

B.2 Future Additions

   It is the intention that any future protocol or profile regarding
   both for media delivery and lower transport should be easy to add to
   RTSP. This section provides the necessary steps that needs to be
   meet.

   The following things needs to be considered when adding a new
   protocol of profile for use with RTSP:

        o The protocol or profile needs to define a name tag
          representing it. This tag is required to be a ABNF "token" to
          be possible to use in the Transport header specification.

        o The useful combinations of protocol/profile/lower-layer needs
          to be defined and for each combination declare the necessary
          parameters to use in the Transport header.

        o For new media protocols the interaction with RTSP needs to be
          addressed. One important factor will be the media
          synchronization.

   See the IANA section (20) for information how to register new
   attributes.

C Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 2327 [2]) may be used to
   describe streams or presentations in RTSP. This description is
   typically returned in reply to a DESCRIBE request on an URI from a
   server to a client, or received via HTTP from a server to a client.

   This appendix describes how an SDP file determines the operation of
   an RTSP session. SDP as is provides no mechanism by which a client
   can distinguish, without human guidance, between several media
   streams to be rendered simultaneously and a set of alternatives
   (e.g., two audio streams spoken in different languages). However the
   SDP extension "Grouping of Media Lines in the Session Description
   Protocol (SDP)" [40] may provide such functionality depending on
   need. Also future grouping semantics may in the future be developed.

C.1 Definitions



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   The terms "session-level", "media-level" and other key/attribute
   names and values used in this appendix are to be used as defined in
   SDP (RFC 2327 [2]):

C.1.1 Control URI

   The "a=control:" attribute is used to convey the control URI.  This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URI to be used for
   controlling that particular media stream. If found at the session
   level, the attribute indicates the URI for aggregate control
   (presentation URI). The session level URI SHALL be different from any
   media level URI. The presence of a session level control attribute
   SHALL be interpreted as support for aggregated control. The control
   attribute SHALL be present on media level unless the presentation
   only contains a single media stream, in which case the attribute MAY
   only be present on the session level.


   control-attribute  =  "a=" "control" ":" url


   Example:

     a=control:rtsp://example.com/foo



   This attribute MAY contain either relative and absolute URIs,
   following the rules and conventions set out in RFC 2396 [13].
   Implementations SHALL look for a base URI in the following order:

        1.   the RTSP Content-Base field; .IP 2.  the RTSP Content-
             Location field; .IP 3.  the RTSP Request-URI.

   If this attribute contains only an asterisk (*), then the URI SHALL
   be treated as if it were an empty embedded URI, and thus inherit the
   entire base URI.

   The URI handling for SDPs from container files need special
   consideration. For example in a container file with the URI:
   "rtsp://example.com/container.mp4". Lets assume this URI as base URI,
   and a media level URI:  "rtsp://example.com/container.mp4/trackID=2".
   A relative media level URI that resolves in accordance with RFC 2396
   [13] to the above given media URI are: "container.mp4/trackID=2". It
   is usually not desirable to need to include in or modify the SDP
   stored within the container file with the server local name of the
   container file. To avoid this, one can modify the base URI used to



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   include a trailing slash, e.g.  "rtsp://example.com/container.mp4/".
   In this case the relative URI for the media will only need to be:
   "trackID=2". However this will also mean that using "*" in the SDP
   will result in control URI including the trailing slash, i.e.
   "rtsp://example.com/container.mp4/".

C.1.2 Media Streams

   The "m=" field is used to enumerate the streams. It is expected that
   all the specified streams will be rendered with appropriate
   synchronization. If the session is a multicast, the port number
   indicated SHOULD be used for reception. The client MAY try to
   override the destination port, through the Transport header.  The
   servers MAY allow this, the response will indicate if allowed or not.
   If the session is unicast, the port number is the ones RECOMMENDED by
   the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.
   The client MAY ignore this recommendation. If the server has no
   preference, it SHOULD set the port number value to zero.

   The "m=" lines contain information about what transport protocol,
   profile, and possibly lower-layer is to be used for the media stream.
   The combination of transport, profile and lower layer, like
   RTP/AVP/UDP needs to be defined for how to be used with RTSP.  The
   currently defined combinations are defined in section B, further
   combinations MAY be specified.

   TODO: Write something about the usage of Grouping of media line, RFC
   3388 [40].

   Example:

     m=audio 0 RTP/AVP 31



C.1.3 Payload Type(s)

   The payload type(s) are specified in the "m=" field. In case the
   payload type is a static payload type from RFC 3551 [3], no other
   information may be required. In case it is a dynamic payload type,
   the media attribute "rtpmap" is used to specify what the media is.
   The "encoding name" within the "rtpmap" attribute may be one of those
   specified in RFC 3551 (Sections 5 and 6), or an MIME type registered
   with IANA, or an experimental encoding as specified in SDP (RFC 2327
   [2]). Codec-specific parameters are not specified in this field, but
   rather in the "fmtp" attribute described below.




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C.1.4 Format-Specific Parameters

   Format-specific parameters are conveyed using the "fmtp" media
   attribute. The syntax of the "fmtp" attribute is specific to the
   encoding(s) that the attribute refers to. Note that some of the
   format specific parameters may be specified outside of the fmtp
   parameters, like for example the "ptime" attribute for most audio
   encodings.

C.1.5 Range of Presentation

   The "a=range" attribute defines the total time range of the stored
   session or an individual media. Non-seekable live sessions can be
   indicated, while the length of live sessions can be deduced from the
   "t" and "r" SDP parameters.

   The attribute is both a session and a media level attribute. For
   presentations that contains media streams of the same durations, the
   range attribute SHOULD only be used at session-level. In case of
   different length the range attribute MUST be given at media level for
   all media, and SHOULD NOT be given at session level. If the attribute
   is present at both media level and session level the media level
   values SHALL be used.

   The unit is specified first, followed by the value range. The units
   and their values are as defined in Section 3.4, 3.5 and 3.6 and MAY
   be extended with further formats. Any open ended range (start-), i.e.
   without stop range, is of unspecified duration and SHALL be
   considered as non-seekable content unless this property is
   overridden.

   This attribute is defined in ABNF [5] as:

   a-range-def = "a" "=" "range" ":" ranges-specifier CRLF


   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203
     Non seekable stream of unknown duration:
     a=range:npt=0-



C.1.6 Time of Availability

   The "t=" field MUST contain suitable values for the start and stop



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   times for both aggregate and non-aggregate stream control.  The
   server SHOULD indicate a stop time value for which it guarantees the
   description to be valid, and a start time that is equal to or before
   the time at which the DESCRIBE request was received. It MAY also
   indicate start and stop times of 0, meaning that the session is
   always available.

   For sessions that are of live type, i.e. specific start time, unknown
   stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
   to indicate the start time of the event. The stop time SHOULD be
   given so that the live event will with high probability have ended at
   that time, while still not be unnecessary long into the future.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. For a media destination address that is a IPv6 one, the SDP
   extension defined in [21] needs to be used.  For on-demand unicast
   streams and some multicast streams, the destination address MAY be
   specified by the client via the SETUP request, thus overriding any
   specified address. To identify streams without a fixed destination
   address, where the client is required to specify a destination
   address, the "c=" field SHOULD be set to a null value. For addresses
   of type "IP4", this value SHALL be "0.0.0.0", and for type "IP6",
   this value SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address
   according to RFC 3513 [22].

C.1.8 Entity Tag

   The optional "a=etag" attribute identifies a version of the session
   description. It is opaque to the client. SETUP requests may include
   this identifier in the If-Match field (see section 14.25) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.


   a-etag-def   =  "a" "=" "etag" ":" etag-string CRLF
   etag-string  =  1*(%x01-09/%x0B-0C/%x0E-FF)


   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a







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        One could argue that the "o=" field provides identical
        functionality. However, it does so in a manner that would
        put constraints on servers that need to support multiple
        session description types other than SDP for the same piece
        of media content.

C.2 Aggregate Control Not Available

   If a presentation does not support aggregate control no session level
   "a=control:" attribute is specified. For a SDP with multiple media
   sections specified, each section will have its own control URI
   specified via the "a=control:" attribute.

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I came from a web page
   e=adm@example.com
   c=IN IP4 0.0.0.0
   t=0 0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.com/movie.vid



   Note that the position of the control URI in the description implies
   that the client establishes separate RTSP control sessions to the
   servers audio.com and video.com

   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URIs,
   and a session-level "a=control:" attribute which is used as the
   Request-URI for aggregate control. If the media-level URI is
   relative, it is resolved to absolute URIs according to Section C.1.1
   above.




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   Example:


   C->M: DESCRIBE rtsp://example.com/movie RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Base: rtsp://example.com/movie/
         Content-Length: 164

         v=0
         o=- 2890844256 2890842807 IN IP4 204.34.34.32
         s=I contain
         i=<more info>
         e=adm@example.com
         c=IN IP4 0.0.0.0
         t=0 0
         a=control:*
         m=video 8002 RTP/AVP 31
         a=control:trackID=1
         m=audio 8004 RTP/AVP 3
         a=control:trackID=2



   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URIs
   rtsp://example.com/movie/trackID=1 and
   rtsp://example.com/movie/trackID=2 to set up the video and audio
   streams, respectively. The URI rtsp://example.com/movie/ , which is
   resolved from the "*", controls the whole presentation (movie).

   A client is not required to issues SETUP requests for all streams
   within an aggregate object. Servers should allow the client to ask
   for only a subset of the streams.

C.4 RTSP external SDP delivery

   There are some considerations that needs to be made when the session
   description is delivered to client outside of RTSP, for example in
   HTTP or email.

   First of all the SDP needs to contain absolute URIs, relative will in
   most cases not work as the delivery will not correctly forward the
   base URI. And as SDP might be temporarily stored on file system



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   before being loaded into an RTSP capable client, thus if possible to
   transport the base URI it still would need to be merged into the
   file.

   The writing of the SDP session availability information, i.e. "t="
   and "r=", needs to be carefully considered. When the SDP is fetched
   by the DESCRIBE method it is with very high probability that the it
   is valid. However the same are much less certain for SDPs distributed
   using other methods. Therefore the publisher of the SDP should take
   care to follow the recommendations about availability in the SDP
   specification [2].

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

        o Generate the following requests: SETUP, TEARDOWN, PLAY.

        o Include the following headers in requests: CSeq, Connection,
          Session, Transport.

        o Parse and understand the following headers in responses:
          CSeq, Connection, Session, Transport, Content-Language,
          Content-Encoding, Content-Length, Content-Type.

        o Understand the class of each error code received and notify
          the end-user, if one is present, of error codes in classes 4xx
          and 5xx. The notification requirement may be relaxed if the
          end-user explicitly does not want it for one or all status
          codes.

        o Expect and respond to asynchronous requests from the server,
          such as REDIRECT. This does not necessarily mean that it
          should implement the REDIRECT method, merely that it MUST
          respond positively or negatively to any request received from
          the server.

   Though not required, the following are RECOMMENDED.

        o Implement RTP/AVP/UDP as a valid transport.

        o Inclusion of the User-Agent header.

        o Understand SDP session descriptions as defined in Appendix C

        o Accept media initialization formats (such as SDP) from



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          standard input, command line, or other means appropriate to
          the operating environment to act as a "helper application" for
          other applications (such as web browsers).


        There may be RTSP applications different from those
        initially envisioned by the contributors to the RTSP
        specification for which the requirements above do not make
        sense. Therefore, the recommendations above serve only as
        guidelines instead of strict requirements.

D.1.1 Basic Playback

   To support on-demand playback of media streams, the client MUST
   additionally be able to do the following:

        o generate the PAUSE request;

        o implement the REDIRECT method, and the Location header.

D.1.2 Authentication-enabled

   In order to access media presentations from RTSP servers that require
   authentication, the client MUST additionally be able to do the
   following:

        o recognize the 401 (Unauthorized) status code;

        o parse and include the WWW-Authenticate header;

        o implement Basic Authentication and Digest Authentication.

D.2 Server

   A minimal server implementation MUST be able to do the following:

        o Implement the following methods: SETUP, TEARDOWN, OPTIONS and
          PLAY.

        o Include the following headers in responses:  Connection,
          Content-Length, Content-Type, Content-Language, Content-
          Encoding, Timestamp, Transport, Proxy-Supported, Public, and
          Via, and Unsupported. RTP-compliant implementations MUST also
          implement the RTP-Info field.

        o Parse and respond appropriately to the following headers in
          requests: Connection, Proxy-Require, Session, Transport, and
          Require.



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   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial
   implementations and/or to be a "good citizen".

        o Implement RTP/AVP/UDP as a valid transport.

        o Inclusion of the Server, Cache-Control Date, and Expires
          headers.

        o Implement the DESCRIBE method.

        o Generate SDP session descriptions as defined in Appendix C


        There may be RTSP applications different from those
        initially envisioned by the contributors to the RTSP
        specification for which the requirements above do not make
        sense. Therefore, the recommendations above serve only as
        guidelines instead of strict requirements.

D.2.1 Basic Playback

   To support on-demand playback of media streams, the server MUST
   additionally be able to do the following:

        o Recognize the Range header, and return an error if seeking is
          not supported.

        o Implement the PAUSE method.

   In addition, in order to support commonly-accepted user interface
   features, the following are highly recommended for on-demand media
   servers:

        o Include and parse the Range header, with NPT units.
          Implementation of SMPTE units is recommended.

        o Include the length of the media presentation in the media
          initialization information.

        o Include mappings from data-specific timestamps to NPT. When
          RTP is used, the rtptime portion of the RTP-Info field may be
          used to map RTP timestamps to NPT.


        Client implementations may use the presence of length
        information to determine if the clip is seekable, and
        visably disable seeking features for clips for which the



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        length information is unavailable. A common use of the
        presentation length is to implement a "slider bar" which
        serves as both a progress indicator and a timeline
        positioning tool.

   Mappings from RTP timestamps to NPT are necessary to ensure correct
   positioning of the slider bar.

D.2.2 Authentication-enabled

   In order to correctly handle client authentication, the server MUST
   additionally be able to do the following:

        o Generate the 401 (Unauthorized) status code when
          authentication is required for the resource.

        o Parse and include the WWW-Authenticate header

        o Implement Basic Authentication and Digest Authentication

E Requirements for Unreliable Transport of RTSP messages


   This section provides any one intending to define how to transport of
   RTSP messages over a unreliable transport protocol with some
   information learned by the attempt in RFC 2326 [1]. RFC 2326 define
   both an URI scheme and some basic functionality for transport of RTSP
   messages over UDP, however it was not sufficient for reliable usage
   and successful interoperability.

   The RTSP scheme defined for unreliable transport of RTSP messages was
   "rtspu". It has been reserved by this specification as at least one
   commercial implementation exist, thus avoiding any collisions in the
   name space.

   The following considerations should exist for operation of RTSP over
   an unreliable transport protocol:

        o Request shall be acknowledged by the receiver. If there is no
          acknowledgement, the sender may resend the same message after
          a timeout of one round-trip time (RTT). Any retransmissions
          due to lack of acknowledgement must carry the same sequence
          number as the original request.

        o The round-trip time can be estimated as in TCP (RFC 1123)
          [41], with an initial round-trip value of 500 ms. An
          implementation may cache the last RTT measurement as the
          initial value for future connections.



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        o If RTSP is used over a small-RTT LAN, standard procedures for
          optimizing initial TCP round trip estimates, such as those
          used in T/TCP (RFC 1644) [42], can be beneficial.

        o The Timestamp header (Section 14.44) is used to avoid the
          retransmission ambiguity problem [43] and obviates the need
          for Karn's algorithm.

        o The registered default port for UDP for the RTSP server is
          554.

        o RTSP messages can be carried over any lower-layer transport
          protocol that is 8-bit clean.

        o RTSP messages are vulnerable to bit errors and SHOULD NOT be
          subjected to them.

        o Source authentication, or at least validation that RTSP
          messages comes from the same entity becomes extremely
          important, as session hijacking may be substantially easier
          for RTSP message transport using an unreliable protocol like
          UDP than for TCP.

   There exist two RTSP header thats primarily are intended for being
   used by the unreliable handling of RTSP messages and which will be
   maintained:

        CSeq See section  14.19

        Timestamp See section  14.44


F Backwards Compatibility Considerations

   This section contains notes on issues about backwards compatibility
   with clients or servers being implemented according to RFC 2326 [1].
   Any mechanism described in this section is intended for a migration
   period and are expected to be possible to phase out.

F.1 Requirement on Pause before Play in Play mode

   The behavior in Play mode after having run to the end of a media
   stream has been changed, see section  11.4. For state handling
   consistency, a client is now required to issue an PAUSE prior to a
   PLAY request. However as this could make a RFC 2326 client become
   stuck after having played a media stream to its end, the following
   mitigation is suggested:




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   If a server receives a PLAY request when in play state and all media
   has finished the requested play out, the server MAY interpret this as
   a PLAY request received in ready state.

   However the server SHALL NOT do the above if the client has shown any
   support for this or newer specifications, for example by sending a
   Supported header with the play.basic feature tag.

F.2 Usage of persistent connections

   Some implementations according to RFC 2326, requires the client to
   use persistent connection. The client closing the connection may
   result in that the server removes the session. To achieve
   interoperability with old servers any client is strongly RECOMMENDED
   to use persistent connections.


G Open Issues

   This section contains a list of open issues that still needs to be
   resolved. However also any open issues in the bug tracker at
   http://rtspspec.sourceforge.net should also be considered.

        1.   Is the example in Section 16.4 valid?

        2.   Should the SDP appendix contain any text in regards to the
             grouping of media line?

        3.   Following resolved Issue needs text: "Should refusal by
             server to perform media redirection have its own error
             code?"  http://rtsp.org/bug991609.

        4.   Need to shape up language in relation to the following
             issue: "Is current methods to prevent undesired media
             redirection sufficient." http://rtsp.org/bug889699

        5.   Shape up language to what was decided in San Diego on
             issue:  "Lacking Specification text for "Implicit
             Redirect?""  http://rtsp.org/bug742348

        6.   Need write up on issue: "Should further explanation on
             proxies be written?" http://rtsp.org/bug631148

        7.   Needs to add explicit white spacing for the syntax.
             Consider to copy the RFC 3261 concept to include white
             spacing in separators a COLON, SEMI, etc.

        8.   ABNF Syntax needs to be run through verifier.



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        9.   The proxy indications in the two header tables in section
             14 needs review.

        10.  Should the Allow header be possible to use optional in
             request or responses besides the now specified 405 error
             code?

        11.  The minimal implementation needs to be checked to see if it
             complies with the specification. All shall, must and
             shoulds needs to be included in the minimal. Feature-tags
             for these needs to be defined. Further feature-tags needs
             to be discussed.

        12.  The list specifying which status codes are allowed on which
             request methods seem to be in error and need review.

        13.  There is need for clearer rule in regards to Transport
             parameters changes in mid session. It needs to be
             determined if there should be any clarification on how and
             which Transport header parameters that may be changed.

        14.  Normative suggestion is needed for doing RTSP session keep
             alives. Currently there are too many options being
             suggested by RTSP such as OPTIONS with Session ID, PING,
             SET_PARAMETER. This leads to interoperability problems,
             maintenance issues and additional development for
             implementers for little gain.

H Changes

H.1 Issues Addressed

   Compared to RFC 2326, the following issues has been addressed:

        o The Transport header has been changed in the following way:

          - The ABNF has been changed to define that extensions are
            possible, and that unknown extension parameters are to be
            ignored.

          - To prevent backwards compatibility issues, any extension or
            new parameter requires the usage of a feature tag combined
            with the Require header.

          - Syntax unclarities with the Mode parameter has been
            resolved.

          - Syntax error with ";" for multicast and unicast has been



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            resolved.

          - Two new addressing parameters has been defined, src_addr and
            dest_addr. These allow one to specify more than one complete
            address and port tuple if needed.

          - Support for IPv6 explicit addresses in all address fields
            has been included.

          - To handle URI definitions that contain ";" or "," a quoted
            URI format has been introduced.

          - Defined IANA registries for the transport headers
            parameters, transport-protocol, profile, lower-transport,
            and mode.

          - The transport headers interleaved parameter's text was made
            more strict and use formal requirements levels. However no
            change on how it is used was made.

          - It has been clarified that the client can't request of the
            server to use a certain RTP SSRC, using a request with the
            transport parameter SSRC.

          - Syntax definition for SSRC has been clarified to require 8*8
            HEX. It has also been extend to allow multiple values for
            clients supporting this version.

          - Updated the text on the transport headers "destination" and
            "dest_addr" parameters regarding what security precautions
            the server is required to perform.

          - The embedded (interleaved) binary data and its transport
            parameter was clarified to being symmetric and that it is
            the server that sets the channel numbers.


H.2 Changes made to the protocol and specification

 o The Range formats has been changed in the following way:

   - The NPT format has been given a initial NPT identifier that should
     be used, if missing NPT is assumed.

   - All formats now support initial open ended formats of type "npt=-
     10".

 o RTSP message handling has been changed in the following way:



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   - RTSP messages now uses URIs rather then URLs.

   - It has been clarified that a 4xx message due to missing CSeq header
     shall be returned without a CSeq header.

   - Rules for how to handle timing out RTSP messages has been added.

 o The HTTP references has been updated to RFC 2616 and RFC 2617. This
   has resulted in that the Public, and the Content-Base header needed
   to be defined in the RTSP specification. Known effects on RTSP due to
   HTTP clarifications:

   - Content-Encoding header can include encoding of type "identity".

 o The state machine section has completely been rewritten. It includes
   now more details and are also more clear about the model used.

 o A IANA section has been included with contains a number of registries
   and their rules. This will allow us to use IANA to keep track of all
   RTSP extensions.

 o Than transport of RTSP messages has seen the following changes:

   - The use of UDP for RTSP message transport has been deprecated due
     to missing interest and to broken specification.

   - The rules for how TCP connections is to be handled has been
     clarified. Now it is made clear that servers should not close the
     TCP connection unless they have been unused for significant time.

   - Strong recommendations why server and clients should use persistent
     connections has also been added.

   - There is now a requirement to handle non-persistent connections as
     this provides great fault tolerance.

   - Added wording on the usage of Connection:Close for RTSP.

   - specified usage of TLS for RTSP messages, including a scheme to
     approve a proxies TLS connection to the next hop.

 o The following header related changes have been made:

   - Accept-Ranges response header is added. This header clarifies which
     range formats that can be used for a resource.

   - Clarified that Range header allows multiple ranges to allow for
     creating editing list.



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   - Fixed the missing definitions for the Cache-Control header. Also
     added to the syntax definition the missing delta-seconds for max-
     stale and min-fresh parameters.

   - Put requirement on CSeq header that the value is increased by one
     for each new RTSP request. A Recommendation to start at 1 has also
     been added.

   - Added requirement that the Date header must be used for all
     messages with entity. Also the Server should always include it.

   - Removed possibility of using Range header with Scale header to
     indicate when it is to be activated, since it can't work as
     defined. Also added rule that lack of Scale header in response
     indicates lack of support for the header. Feature-tags for scaled
     playback has been defined.

   - The Speed header must now be responded to indicate support and the
     actual speed going to be used. A feature-tag is defined. Notes on
     congestion control was also added.

   - The Supported header was borrowed from SIP to help with the feature
     negotiation in RTSP.

   - Clarified that the Timestamp header can be used to resolve
     retransmission ambiguities.

   - The Session header text has been expanded with a explanation on
     keep alive and which methods to use.

   - It has been clarified how the Range header formats is used to
     indicate pause points.

   - Clarified that RTP-Info URIs that are relative, uses the Request-
     URI as base URI. Also clarified that the URI that must be used is
     the SETUP.

   - Added text that requires the Range to always be present in PLAY
     responses. Clarified what should be sent in case of live streams.

   - The quoted URI format may also be used with the RTP-Info header.
     Backwards compatibility issues exist with such usage, thus it can
     only be used for implementations following this specification.

   - The headers table has been updated using a structured borrowed from
     SIP. This table carries much more information and should provide a
     good overview of the available headers.




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   - It has been is clarified that any message with a message body is
     required to have a Content-Length header. This was the case in RFC
     2326 but could be misinterpreted.

   - To resolve functionality around ETag. The ETag and If-None-Match
     header has been added from HTTP with necessary clarification in
     regards to RTSP operation.

   - Imported the Public header from HTTP RFC 2068 [19] since it has
     been removed from HTTP due to lack of use. Public is used quite
     frequently in RTSP.

   - Clarified rules for populating the Public header so that it is an
     intersection of the capabilities of all the RTSP agents in a chain.

 o The minimal implementation specification has been changed:

   - Required Timestamp, Via, and Unsupported headers for a minimal
     server implementation.

   - Recommended that Cache-Control, Expires and Date headers be
     supported by server implementations.

 o The Protocol Syntax has been changed in the following way:

   - All BNF definitions are updated according to the rules defined in
     RFC 2234 [5] and has been gathered in a separate section  18.

   - The BNF for the User-Agent and Server headers has been corrected so
     now only the description is in the HTTP specification.

   - The definition in the introduction of the RTSP session has been
     changed.

   - The protocol has been made fully IPv6 capable. Certain of the
     functionality, like using explicit IPv6 addresses in fields
     requires that the protocol support this updated specification.

   - Added a fragment part to the RTSP URI. This seem to be indicated by
     the note below the definition however it was not part of the BNF.

   - The CHAR rule has been changed to exclude NULL.

 o The Status codes has been changed in the following way:

   - The use of status code 303 "See Other" has been deprecated as it
     does not make sense to use in RTSP.




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   - When sending response 451 and 458 the response body should contain
     the offending parameters.

   - Clarification on when a 3rr redirect status code can be received
     has been added. This includes receiving 3rr as a result of request
     within a established session. This provides clarification to a
     previous unspecified behavior.

   - Removed the 250 (Low On Storage Space) status code as it only is
     relevant to recording which is deprecated.

 o The following functionality has been deprecated from the protocol:

   - The use of Queued Play.

   - The use of PLAY method for keep-alive in play state.

   - The RECORD and ANNOUNCE methods and all related functionality. Some
     of the syntax has been removed.

   - The possibility to use timed execution of methods with the time
     parameter in the Range header.

   - The description on how rtspu works is not part of the core
     specification and will require external description. Only that it
     exist is defined here and some requirements for the the transport
     is provided.

 o Text specifying the special behavior of PLAY for live content.

 o The following changes has been made in relation to methods:

   - The OPTIONS method has been clarified with regards to the use of
     the Public and Allow headers.

   - The RECORD and ANNOUNCE methods are removed as they are lacking
     implementation and not considered necessary in the core
     specification. Any work on these methods should be done as a
     extension document to RTSP.

   - Added text clarifying the usage of SET_PARAMETER for keep-alive and
     usage without any body.

   - Added a backwards compatibility resolution for how to handle the
     new state machine without automatic state transition, for example
     for returning to ready when finished playing.

 o Wrote a new section about how to setup different media transport



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   alternatives and their profiles, and lower layer protocols. This
   resulted that the appendix on RTP interaction was moved there instead
   in the part describing RTP. The section also includes guidelines what
   to think of when writing usage guidelines for new protocols and
   profiles.

 o Added a new section describing the available mechanisms to determine
   if functionality is supported, called "Capability Handling". Renamed
   option-tags to feature-tags.

 o Added a contributors section with people who has contribute actual
   text to the specification.

 o Added a section Use Cases that describes the major use cases for
   RTSP.

 o Clarified the usage of a=range and how to indicate live content that
   are not seekable with this header.

   Note that this list does not reflect minor changes in wording or
   correction of typographical errors.

   A word-by-word diff from RFC 2326 can be found at http://rtsp.org/

I Author Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail: schulzrinne@cs.columbia.edu

   Anup Rao
   Cisco
   USA
   electronic mail: anrao@cisco.com

   Robert Lanphier
   RealNetworks
   P.O. Box 91123
   Seattle, WA 98111-9223
   USA
   electronic mail: robla@real.com

   Magnus Westerlund
   Ericsson AB, EAB/TVA/A



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   Torshamsgatan 23
   SE-164 80 STOCKHOLM
   SWEDEN
   electronic mail: magnus.westerlund@ericsson.com

   Aravind Narasimhan
   Princeton, NJ
   USA
   electronic mail: aravind.narasimhan@gmail.com

J Contributors

   The following people has made written contribution included in the
   specification:

        o Tom Marshall has contributed with text about the usage of 3rr
          status codes.

        o Thomas Zheng has contributed with text regarding the usage of
          the Range in PLAY responses.

        o Sean Sheedy has contributed the text regarding the timing out
          of RTSP messages.

        o Fredrik Lindholm has contributed with text for the RTSP
          security framework.

   The following persons has provided detailed comments on the updated
   version of the specification:

        o Stephan Wenger

K Acknowledgements

   This draft is based on the functionality of the original RTSP draft
   submitted in October 1996. It also borrows format and descriptions
   from HTTP/1.1.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already
   mentioned, the following individuals have contributed to this
   specification:

   Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
   Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
   Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
   Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
   John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,



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   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
   Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
   Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
   Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
   Chesire, David Walker, Geetha Srikantan, Stephan Wenger, Pekka Pessi,
   and Mela Martti.

L Normative References

   [1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [2] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.

   [3] H. Schulzrinne and S. Casner, "RTP profile for audio and video
   conferences with minimal control," RFC 3551, Internet Engineering
   Task Force, July 2003.

   [4] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, L. Masinter, P.
   J. Leach, and T. Berners-Lee, "Hypertext transfer protocol --
   HTTP/1.1," RFC 2616, Internet Engineering Task Force, June 1999.

   [5] "Augmented BNF for syntax specifications: ABNF," RFC 2234,
   Internet Engineering Task Force, Nov. 1997.

   [6] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [7] T. Dierks and C. Allen, "The TLS protocol version 1.0," RFC 2246,
   Internet Engineering Task Force, Jan. 1999.

   [8] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. J.
   Leach, A. Luotonen, and L. Stewart, "HTTP authentication: Basic and
   digest access authentication," RFC 2617, Internet Engineering Task
   Force, June 1999.

   [9] J. B. Postel, "User datagram protocol," RFC 768, Internet
   Engineering Task Force, Aug. 1980.

   [10] J. B. Postel, "Transmission control protocol," RFC 793, Internet
   Engineering Task Force, Sept. 1981.

   [11] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
   Internet Engineering Task Force, Apr. 1996.




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   [12] R. Hinden, B. E. Carpenter, and L. Masinter, "Format for literal
   IPv6 addresses in URL's," RFC 2732, Internet Engineering Task Force,
   Dec. 1999.

   [13] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   identifiers (URI): generic syntax," RFC 2396, Internet Engineering
   Task Force, Aug.  1998.

   [14] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   2279, Internet Engineering Task Force, Jan. 1998.

   [15] NIST, "Fips pub 180-1:secure hash standard," tech. rep.,
   National Institute of Standards and Technology, Apr. 1995.

   [16] R. Housley, W. Polk, W. Ford, and D. Solo, "Internet X.509
   public key infrastructure certificate and certificate revocation list
   (CRL) profile," RFC 3280, Internet Engineering Task Force, Apr. 2002.

   [17] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," RFC 3550, Internet
   Engineering Task Force, July 2003.

   [18] E. Rescorla, "HTTP over TLS," RFC 2818, Internet Engineering
   Task Force, May 2000.

   [19] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, and T.
   Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068,
   Internet Engineering Task Force, Jan. 1997.

   [20] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
   considerations section in RFCs," RFC 2434, Internet Engineering Task
   Force, Oct. 1998.

   [21] S. Olson, G. Camarillo, and A. B. Roach, "Support for IPv6 in
   session description protocol (SDP)," RFC 3266, Internet Engineering
   Task Force, June 2002.

   [22] R. Hinden and S. E. Deering, "Internet protocol version 6 (ipv6)
   addressing architecture," RFC 3513, Internet Engineering Task Force,
   Apr. 2003.

M Informative References

   [23] T. Z. M. Westerlund, "How to make real-time streaming protocol
   (rtsp) traverse network address translators (nat) and interact with
   firewalls.," internet draft, Internet Engineering Task Force, Feb.
   2004.  Work in progress.




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   [24] A. Narasimhan, "Mute and unmute extension to rtsp," internet
   draft, Internet Engineering Task Force, Feb. 2002.  Work in progress.

   [25] P. Gentric, "Rtsp stream switching," internet draft, Internet
   Engineering Task Force, Jan. 2004.  Work in progress.

   [26] A. L. G. Srikantan, J. Murata, "Streaming relays," internet
   draft, Internet Engineering Task Force, Dec. 2003.  Work in progress.

   [27] F. Yergeau, G. Nicol, G. C. Adams, and M. Duerst,
   "Internationalization of the hypertext markup language," RFC 2070,
   Internet Engineering Task Force, Jan.  1997.

   [28] H. Schulzrinne, "A comprehensive multimedia control architecture
   for the Internet," in Proc. International Workshop on Network and
   Operating System Support for Digital Audio and Video (NOSSDAV), (St.
   Louis, Missouri), May 1997.

   [29] International Telecommunication Union, "Visual telephone systems
   and equipment for local area networks which provide a non-guaranteed
   quality of service," Recommendation H.323, Telecommunication
   Standardization Sector of ITU, Geneva, Switzerland, May 1996.

   [30] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.

   [31] J. Miller, P. Resnick, and D. Singer, "Rating services and
   rating systems (and their machine readable descriptions),"
   Recommendation REC-PICS-services-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [32] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
   distribution label syntax and communication protocols,"
   Recommendation REC-PICS-labels-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [33] D. L. Mills, "Network time protocol (version 3) specification,
   implementation," RFC 1305, Internet Engineering Task Force, Mar.
   1992.

   [34] ISO/IEC, "Information technology -- generic coding of moving
   pictures and associated audio informaiton -- part 6: extension for
   digital storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [35] ISO/IEC, "Data elements and interchange formats -- information
   interchange -- representation of dates and times," Published standard



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   ISO 8601, International Organization for Standardization ISO/IEC,
   Geneva, Switzerland, Dec. 2000.

   [36] S. Josefsson and I. W. Ed., "The base16, base32, and base64 data
   encodings," RFC 3548, Internet Engineering Task Force, July 2003.

   [37] Third Generation Partnership Project (3GPP), "Transparent end-
   to-end packet-switched streaming service (pss); protocols and
   codecs," Technical Specification 26.234, Third Generation Partnership
   Project (3GPP), Dec. 2002.

   [38] D. Yon, "Connection-oriented media transport in sdp," internet
   draft, Internet Engineering Task Force, Mar. 2003.  Work in progress.

   [39] J. Lazzaro, "Framing rtp and rtcp packets over connection-
   oriented transport," internet draft, Internet Engineering Task Force,
   Oct. 2003.  Work in progress.

   [40] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
   "Grouping of media lines in the session description protocol (SDP),"
   RFC 3388, Internet Engineering Task Force, Dec. 2002.

   [41] "Requirements for Internet hosts - application and support," RFC
   1123, Internet Engineering Task Force, Oct. 1989.

   [42] R. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," RFC 1644, Internet Engineering Task Force, July 1994.

   [43] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
   Reading, Massachusetts: Addison-Wesley, 1994.


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