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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
                                                             Columbia U.
                                                                  A. Rao
                                                                   Cisco
                                                             R. Lanphier
                                                            RealNetworks
                                                           M. Westerlund
                                                                Ericsson


draft-ietf-mmusic-rfc2326bis-01.txt
June 06, 2002
Expires: December, 2002


                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   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 mate-
   rial 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 Pro-
   posed Standard.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time proper-
   ties. RTSP provides an extensible framework to enable controlled, on-
   demand delivery of real-time data, such as audio and video. Sources



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   of data can include both live data feeds and stored clips. This pro-
   tocol is intended to control multiple data delivery 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 1889).














































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1 Introduction

1.1 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   either a single or several time-synchronized streams of continuous
   media such as audio and video. It does not typically deliver the con-
   tinuous streams itself, although interleaving of the continuous media
   stream with the control stream is possible (see Section 10.13). In
   other words, RTSP acts as a "network remote control" for multimedia
   servers.

   The set of streams to be controlled is defined by a presentation
   description. This memorandum does not define a format for a presenta-
   tion description.

   There is no notion of an RTSP connection; instead, a server maintains
   a session labeled by an identifier. An RTSP session is in no way tied
   to a transport-level connection such as a TCP connection. During an
   RTSP session, an RTSP client may open and close many reliable trans-
   port connections to the server to issue RTSP requests. Alternatively,
   it may use a connectionless transport protocol such as UDP.

   The streams controlled by RTSP may use RTP [1], but the operation of
   RTSP does not depend on the transport mechanism used to carry contin-
   uous media.

   The protocol is intentionally similar in syntax and operation to
   HTTP/1.1 [26] 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:

     + RTSP introduces a number of new methods and has a different pro-
       tocol identifier.

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

     + Both an RTSP server and client can issue requests.

     + Data is carried out-of-band by a different protocol. (There is an
       exception to this.)

     + RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
       consistent with current HTML internationalization efforts [3].

     + The Request-URI always contains the absolute URI. Because of
       backward compatibility with a historical blunder, HTTP/1.1 [26]



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       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 request a pre-
          sentation description via 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, either to play back
          media into the presentation or to record all or a subset of
          the media in a presentation. This mode is useful for dis-
          tributed teaching applications. Several parties in the confer-
          ence may take turns "pushing the remote control buttons".

     Addition of media to an existing presentation: Particularly for
          live presentations, it is useful if the server can tell the
          client about additional media becoming available.

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [26].

1.2 Requirements

   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 [4].

1.3 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [26]. Terms
   not listed here are defined as in HTTP/1.1.

     Aggregate control: The control of the multiple streams using a sin-
          gle timeline by the server. For audio/video feeds, this means
          that the client may issue a single play or pause message to
          control both the audio and video feeds.





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

     Client: The client requests continuous media data from the media
          server.

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

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

     Continuous media: Data where there is a timing relationship between
          source and sink; that is, the sink must reproduce the timing
          relationship that existed at the source. The most common exam-
          ples of continuous media are audio and motion video. Continu-
          ous media can be real-time (interactive), 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 metainformation in the form of
          entity-header fields and content in the form of an entity-
          body, as described in Section 8.

     Media initialization: Datatype/codec specific initialization.  This
          includes such things as clockrates, color tables, etc. Any
          transport-independent information which is required by a
          client for playback of a media stream occurs in the media ini-
          tialization 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 or recording 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 or a different host as the
          web server the presentation is invoked from.

     Media server indirection: Redirection of a media client to a dif-
          ferent media server.

     (Media) stream: A single media instance, e.g., an audio stream or a
          video stream as well as a single whiteboard or shared applica-
          tion group. When using RTP, a stream consists of all RTP and



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          RTCP packets created by a source within an RTP session. This
          is equivalent to the definition of a DSM-CC stream([5]).

     Message: The basic unit of RTSP communication, consisting of a
          structured sequence of octets matching the syntax defined in
          Section 15 and transmitted via a connection or a connection-
          less protocol.

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

     Presentation: A set of one or more streams presented to the client
          as a complete media feed, using a presentation description as
          defined below. In most cases in the RTSP context, this implies
          aggregate control of those streams, but does not have to.

     Presentation description: A presentation description contains
          information about one or more media streams within a presenta-
          tion, such as the set of encodings, network addresses and
          information about the content. Other IETF protocols such as
          SDP (RFC 2327 [24]) use the term "session" for a live presen-
          tation. The presentation description may take several differ-
          ent formats, including but not limited to the session descrip-
          tion 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.

     RTSP session: A complete RTSP "transaction", e.g., the viewing of a
          movie. A session typically consists of a client setting up a
          transport mechanism for the continuous media stream (SETUP),
          starting the stream with PLAY or RECORD, and closing the
          stream with TEARDOWN.

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

1.4 Protocol Properties

   RTSP has the following properties:

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





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

     Secure: RTSP re-uses web security mechanisms, either at the trans-
          port level (TLS, RFC 2246 [27]) or within the protocol itself.
          All HTTP authentication mechanisms such as basic (RFC 2616
          [26]) and digest authentication (RFC 2069 [6]) are directly
          applicable.

     Transport-independent: RTSP may use either an unreliable datagram
          protocol (UDP) (RFC 768 [7]), a reliable datagram protocol
          (RDP, RFC 1151, not widely used [8]) or a reliable stream pro-
          tocol such as TCP (RFC 793 [9]) as it implements application-
          level reliability.

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

     Control of recording devices: The protocol can control both record-
          ing and playback devices, as well as devices that can alter-
          nate between the two modes ("VCR").

     Separation of stream control and conference initiation: Stream con-
          trol is divorced from inviting a media server to a conference.
          In particular, SIP [10] or H.323 [28] 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 presenta-
          tion description must contain at least one RTSP URI.

     Proxy and firewall friendly: The protocol should be readily handled
          by both application and transport-layer (SOCKS [11]) fire-
          walls. A firewall may need to understand the SETUP method to
          open a "hole" for the UDP 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
          [12,13]) for associating labels with content. However, RTSP
          does not just add methods to HTTP since the controlling



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          continuous media requires server state in most cases.

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

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

     Capability negotiation: If basic features are disabled, there must
          be some clean mechanism for the client to determine which
          methods are not going to be implemented. This allows clients
          to present the appropriate user interface. For example, if
          seeking is not allowed, the user interface must be able to
          disallow moving a sliding position indicator.


     An earlier requirement in RTSP was multi-client capability.
     However, it was determined that a better approach was to make
     sure that the protocol is easily extensible to the multi-
     client scenario. Stream identifiers can be used by several
     control streams, so that "passing the remote" would be possi-
     ble. The protocol would not address how several clients nego-
     tiate access; this is left to either a "social protocol" or
     some other floor control mechanism.

1.5 Extending RTSP

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

     + A server may only be capable of playback thus has no need to sup-
       port the RECORD request.

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

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

   A server SHOULD implement all header fields described in Section 12.

   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 [26],
   where the methods described in [H19.5] are not likely to be supported
   across all servers.



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

     + Existing methods can be extended with new parameters, as long as
       these parameters can be safely ignored by the recipient. (This is
       equivalent to adding new parameters to an HTML tag.) 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 12.32).

     + 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 SHOULD list the meth-
       ods it supports using the Public response header.

     + A new version of the protocol can be defined, allowing almost all
       aspects (except the position of the protocol version number) to
       change.

1.6 Overall Operation

   Each presentation and media stream may be identified by an RTSP URL.
   The overall presentation and the properties of the media the presen-
   tation 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 main-
   tains 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, lan-
   guage, 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 iden-
   tified by an RTSP URL, 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 exam-
   ple, audio and video streams can be split across servers for load
   sharing.  The description also enumerates which transport methods the



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   server is capable of.

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

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

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

     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 descrip-
          tion, established by means outside the scope of this specifi-
          cation.

1.7 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may
   occur on a TCP connection while the data flows 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 con-
   trolled by RTSP requests issued sequentially on different TCP connec-
   tions.  Therefore, the server needs to maintain "session state" to be
   able to correlate RTSP requests with a stream. The state transitions
   are described in Section A.

   Many methods in RTSP do not contribute to state. However, the follow-
   ing play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEAR-
   DOWN.

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

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

     PAUSE: Temporarily halts a stream without freeing server resources.

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



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          RTSP methods that contribute to state use the Session header
          field (Section 12.37) to identify the RTSP session whose state
          is being manipulated. The server generates session identifiers
          in response to SETUP requests (Section 10.4).

1.8 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may inter-
   act with HTTP in that the initial contact with streaming content is
   often to be made through a web page. The current protocol specifica-
   tion 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
   roundtrips in web-browser-based scenarios, yet also allows for stan-
   dalone RTSP servers and clients which do not rely on HTTP at all.

   However, RTSP differs fundamentally from HTTP in that data delivery
   takes place out-of-band in a different protocol. HTTP is an asymmet-
   ric 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 not stateless; they may set parame-
   ters 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.

   While most real-time media will use RTP as a transport protocol, RTSP
   is not tied to RTP.

   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams.

2 Notational Conventions

   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 [26]).

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used in
   [H2.1]. It is described in detail in RFC 2234 [14], with the differ-
   ence that this RTSP specification maintains the "1#" notation for



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   comma-separated lists.

   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation. This is intended to give readers who were
   not involved with the formulation of the specification an understand-
   ing of why things are the way that they are in RTSP.

3 Protocol Parameters

3.1 RTSP Version

   HTTP Specification Section [H3.1] applies, with HTTP replaced by
   RTSP.

3.2 RTSP URL

   The "rtsp" and "rtspu" schemes are used to refer to network resources
   via the RTSP protocol. This section defines the scheme-specific syn-
   tax and semantics for RTSP URLs.


   rtsp_URL  =  ( "rtsp:" | "rtspu:" )
                "//" host [ ":" port ] [ abs_path ]
   host      =  <A legal Internet host domain name of IP address
                (in dotted decimal form), as defined by Section 2.1
                of RFC 1123 [15]>
   port      =  *DIGIT


   abs_path is defined in [H3.2.1].


     Note that fragment and query identifiers do not have a well-
     defined meaning at this time, with the interpretation left to
     the RTSP server.

   The scheme rtsp requires that commands are issued via a reliable pro-
   tocol (within the Internet, TCP), while the scheme rtspu identifies
   an unreliable protocol (within the Internet, UDP).

   If the port is empty or not given, port 554 is assumed. The semantics
   are that the identified resource can be controlled by RTSP at the
   server listening for TCP (scheme "rtsp") connections or UDP (scheme
   "rtspu") packets on that port of host, and the Request-URI for the
   resource is rtsp_URL.

   The use of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [16]).



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   A presentation or a stream is identified by a textual media identi-
   fier, using the character set and escape conventions [H3.2] of URLs
   (RFC 1738 [17]). URLs may refer to a stream or an aggregate of
   streams, i.e., a presentation. Accordingly, requests described in
   Section 10 can apply to either the whole presentation or an individ-
   ual stream within the presentation. Note that some request methods
   can only be applied to streams, not presentations and vice versa.

   For example, the RTSP URL:

     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 URL:

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


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


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

   The path components of the RTSP URL 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 URL.

3.3 Session Identifiers

   Session identifiers are opaque strings of arbitrary length. Linear
   white space must be URL-escaped. A session identifier MUST be chosen
   randomly and MUST be at least eight octets long to make guessing it
   more difficult. (See Section 16.)





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   session-id  =  8*( ALPHA | DIGIT | safe )


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.


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


   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



3.5 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position rela-
   tive to the beginning of the presentation. 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 event. It may be used only for live events.



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   NPT is defined as in DSM-CC: "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."  [5]

   npt-range       =  ["npt" "="] npt-range-spec
                      ; 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


   Examples:

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




     The syntax conforms to ISO 8601. The npt-sec notation is opti-
     mized 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 timestamps, using UTC (GMT).
   Fractions of a second may be indicated.


   utc-range       =  ["clock" "="] utc-range-spec
   utc-range-spec  =  ( utc-time "-" [ utc-time ] ) | ( "-" utc-time )
   utc-time        =  utc-date "T" utc-time "Z"
   utc-date        =  8DIGIT                                           ; < YYYYMMDD >




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   utc-time        =  6DIGIT [ "." fraction ]                          ; < HHMMSS.fraction >


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

     19961108T143720.25Z



3.7 Option Tags

   Option tags are unique identifiers used to designate new options in
   RTSP. These tags are used in in Require (Section 12.32) and Proxy-
   Require (Section 12.27) header fields.

   Syntax:

   option-tag  =  token


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

3.7.1 Registering New Option Tags with IANA

   When registering a new RTSP option, the following information should
   be provided:

     + Name and description of option. The name may be of any length,
       but SHOULD be no more than twenty characters long. The name MUST
       not contain any spaces, control characters or periods.

     + Indication of who has change control over the option (for exam-
       ple, IETF, ISO, ITU-T, other international standardization bod-
       ies, a consortium or a particular company or group of companies);

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

     + For proprietary options, contact information (postal and email
       address);




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4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [18]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by them-
   selves 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.

   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 [18])

   RTSP messages can be carried over any lower-layer transport protocol
   that is 8-bit clean.

   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):





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     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 12.14) 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 Con-
   tent-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 chun-
     ked transfer encoding unnecessary.

5 General Header Fields

   See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
   and Warning headers are not defined:


   general-header  =  Cache-Control  ; Section 12.9
                   |  Connection     ; Section 12.10
                   |  CSeq           ; Section 12.17
                   |  Date           ; Section 12.18
                   |  Via            ; Section 12.44


6 Request

   A request message from a client to a server or vice versa includes,
   within the first line of that message, the method to be applied to
   the resource, the identifier of the resource, and the protocol ver-
   sion in use.


   Request  =   Request-Line      ; Section 6.1
            *(  general-header    ; Section 5
            |   request-header    ; Section 6.2
            |   entity-header )   ; Section 8.1
                CRLF




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                [ message-body ]  ; Section 4.3


6.1 Request Line


   Request-Line  =  Method SP Request-URI SP RTSP-Version CRLF



   Method  =  "DESCRIBE"        ; Section 10.2
           |  "ANNOUNCE"        ; Section 10.3
           |  "GET_PARAMETER"   ; Section 10.8
           |  "OPTIONS"         ; Section 10.1
           |  "PAUSE"           ; Section 10.6
           |  "PLAY"            ; Section 10.5
           |  "RECORD"          ; Section 10.11
           |  "REDIRECT"        ; Section 10.10
           |  "SETUP"           ; Section 10.4
           |  "SET_PARAMETER"   ; Section 10.9
           |  "TEARDOWN"        ; Section 10.7
           |  extension-method



   extension-method  =  token
   Request-URI       =  "*" | absolute_URI
   RTSP-Version      =  "RTSP" "/" 1*DIGIT "." 1*DIGIT


6.2 Request Header Fields


   request-header  =  Accept             ; Section 12.1
                   |  Accept-Encoding    ; Section 12.2
                   |  Accept-Language    ; Section 12.3
                   |  Authorization      ; Section 12.6
                   |  Bandwidth          ; Section 12.7
                   |  Blocksize          ; Section 12.8
                   |  From               ; Section 12.20
                   |  If-Modified-Since  ; Section 12.23
                   |  Proxy-Require      ; Section 12.27
                   |  Range              ; Section 12.29
                   |  Referer            ; Section 12.30
                   |  Require            ; Section 12.32
                   |  Scale              ; Section 12.34
                   |  Session            ; Section 12.37




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                   |  Speed              ; Section 12.35
                   |  Transport          ; Section 12.40
                   |  User-Agent         ; Section 12.42


   Note that in contrast to HTTP/1.1 [26], RTSP requests always contain
   the absolute URL (that is, including the scheme, host and port)
   rather than just the absolute path.


     HTTP/1.1 requires servers to understand the absolute URL, 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.

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


     OPTIONS * RTSP/1.0



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
   with are defined in Table 1.

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


   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


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



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   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.


   Status-Line  =  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 11. 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:

     + 1xx: Informational - Request received, continuing process

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

     + 3xx: Redirection - Further action must be taken in order to com-
       plete the request

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

     + 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-Phrase's, are
   presented below. The reason phrases listed here are only recommended
   -- they may be replaced by local equivalents without affecting the
   protocol. Note that RTSP adopts most HTTP/1.1 [26] status codes and
   adds RTSP-specific status codes starting at x50 to avoid conflicts
   with newly defined HTTP status codes.



     Status-Code  =  "100"           ; Continue




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                  |  "200"           ; OK
                  |  "201"           ; Created
                  |  "250"           ; Low on Storage Space
                  |  "300"           ; Multiple Choices
                  |  "301"           ; Moved Permanently
                  |  "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
                  |  "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







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


   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 unrecog-
   nized 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 addi-
   tional 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.


   response-header  =  Location            ; Section 12.25
                    |  Proxy-Authenticate  ; Section 12.26
                    |  Public              ; Section 12.28
                    |  Range               ; Section 12.29
                    |  Retry-After         ; Section 12.31
                    |  RTP-Info            ; Section 12.33
                    |  Scale               ; Section 12.34
                    |  Session             ; Section 12.37
                    |  Server              ; Section 12.36
                    |  Speed               ; Section 12.35
                    |  Transport           ; Section 12.40
                    |  Unsupported         ; Section 12.41
                    |  Vary                ; Section 12.43
                    |  WWW-Authenticate    ; Section 12.45


   Response-header field names can be extended reliably only in combina-
   tion with a change in the protocol version. However, new or experi-
   mental 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



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        Code  reason
        --------------------------------------------------------
        100   Continue                          all
        --------------------------------------------------------
        200   OK                                all
        201   Created                           RECORD
        250   Low on Storage Space              RECORD
        --------------------------------------------------------
        300   Multiple Choices                  all
        301   Moved Permanently                 all
        302   Moved Temporarily                 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          SETUP
        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
        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
        --------------------------------------------------------
        500   Internal Server Error             all
        501   Not Implemented                   all
        502   Bad Gateway                       all
        503   Service Unavailable               all
        504   Gateway Timeout                   all



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        505   RTSP Version Not Supported        all
        551   Option not support                all


   Table 1: Status codes and their usage with RTSP methods

   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.

   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 metainformation about the
   entity-body or, if no body is present, about the resource identified
   by the request.


   entity-header     =  Allow             ; Section 12.5
                     |  Content-Base      ; Section 12.11
                     |  Content-Encoding  ; Section 12.12
                     |  Content-Language  ; Section 12.13
                     |  Content-Length    ; Section 12.14
                     |  Content-Location  ; Section 12.15
                     |  Content-Type      ; Section 12.16
                     |  Expires           ; Section 12.19
                     |  Last-Modified     ; Section 12.24
                     |  extension-header
   extension-header  =  message-header


   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]





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9 Connections

   RTSP requests can be transmitted in several different ways:

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

     + one connection per request/response transaction;

     + connectionless mode.

   The type of transport connection is defined by the RTSP URI (Section
   3.2). For the scheme "rtsp", a persistent connection is assumed,
   while the scheme "rtspu" calls for RTSP requests to be sent without
   setting up a connection.

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent connec-
   tions, 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

   Requests are acknowledged by the receiver unless they are sent to a
   multicast group. If there is no acknowledgement, the sender may
   resend the same message after a timeout of one round-trip time (RTT).
   The round-trip time is estimated as in TCP (RFC 1123) [15], with an
   initial round-trip value of 500 ms. An implementation MAY cache the
   last RTT measurement as the initial value for future connections.

   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 can-
     not typically take advantage of the application-layer retrans-
     mission since the transport stack will not deliver the



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     application-layer retransmission before the first attempt has
     reached the receiver. If the packet loss is caused by conges-
     tion, multiple retransmissions at different layers will exac-
     erbate the congestion.

   If RTSP is used over a small-RTT LAN, standard procedures for opti-
   mizing initial TCP round trip estimates, such as those used in T/TCP
   (RFC 1644) [19], can be beneficial.

   The Timestamp header (Section 12.39) is used to avoid the retransmis-
   sion ambiguity problem [20] and obviates the need for Karn's algo-
   rithm.

   Each request carries a sequence number in the CSeq header (Section
   12.17), which is incremented by one for each distinct request trans-
   mitted. 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).

   Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
   support UDP. The default port for the RTSP server is 554 for both UDP
   and TCP.

   A number of RTSP packets destined for the same control end point may
   be packed into a single lower-layer PDU or encapsulated into a TCP
   stream.  RTSP data MAY be interleaved with RTP and RTCP packets.
   Unlike HTTP, an RTSP message MUST contain a Content-Length header
   field whenever that message contains a payload. Otherwise, an RTSP
   packet is terminated with an empty line immediately following the
   last message header.

10 Method Definitions

   The method token indicates the method to be performed on the resource
   identified by the Request-URI case-sensitive. New methods may be
   defined in the future. Method names may not start with a $ character
   (decimal 24) and must be a token. Methods are summarized in Table 2.


   Notes on Table 2: PAUSE is recommended, but not required in that a
   fully functional server can be built that does not support this
   method, for example, for live feeds. If a server does not support a
   particular method, it MUST return 501 (Not Implemented) and a client
   SHOULD not try this method again for this server.

10.1 OPTIONS





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      method         direction   object  requirement
      -------------------------------------------------------------
      DESCRIBE       C->S        P,S     recommended
      ANNOUNCE       C->S, S->C  P,S     optional
      GET_PARAMETER  C->S, S->C  P,S     optional
      OPTIONS        C->S, S->C  P,S     required (S->C: optional)
      PAUSE          C->S        P,S     recommended
      PING           C->S, S->C  P,S     optional
      PLAY           C->S        P,S     required
      RECORD         C->S        P,S     optional
      REDIRECT       S->C        P,S     optional
      SETUP          C->S        S       required
      SET_PARAMETER  C->S, S->C  P,S     optional
      TEARDOWN       C->S        P,S     required


   Table 2: Overview of RTSP methods, their direction, and what  objects
   (P: presentation, S: stream) they operate on

   The behavior is equivalent to that described in [H9.2]. An OPTIONS
   request may be issued at any time, e.g., if the client is about to
   try a nonstandard request. It does not influence server state.

   Example:


     C->S:  OPTIONS * RTSP/1.0
            CSeq: 1
            Require: implicit-play
            Proxy-Require: gzipped-messages

     S->C:  RTSP/1.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE



   Note that these 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).

10.2 DESCRIBE

   The DESCRIBE method retrieves the description of a presentation or
   media object identified by the request URL from a server. It may use
   the Accept header to specify the description formats that the client
   understands. The server responds with a description of the requested



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   resource. The DESCRIBE reply-response pair constitutes the media ini-
   tialization phase of RTSP.

   Example:


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

     S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           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=whiteboard 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization informa-
   tion for the resource(s) that it describes. If a media client obtains
   a presentation description from a source other than DESCRIBE and that
   description contains a complete set of media initialization parame-
   ters, the client SHOULD use those parameters and not then request a
   description for the same media via RTSP.

   Additionally, servers SHOULD NOT use the DESCRIBE response as a means
   of media indirection.


     By forcing a DESCRIBE response to contain all media initial-
     ization for the set of streams that it describes, and discour-
     aging use of DESCRIBE for media indirection, we avoid looping
     problems that might result from other approaches.




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

     + via RTSP's DESCRIBE method;

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

     + via the command line or standard input (thus working as a browser
       helper application launched with an SDP file or other media ini-
       tialization format).

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as a "helper application" that accepts a media initialization
   file from standard input, command line, and/or other means that are
   appropriate to the operating environment of the client.

10.3 ANNOUNCE

   The ANNOUNCE method serves two purposes:

   When sent from client to server, ANNOUNCE posts the description of a
   presentation or media object identified by the request URL to a
   server.  When sent from server to client, ANNOUNCE updates the ses-
   sion description in real-time.

   If a new media stream is added to a presentation (e.g., during a live
   presentation), the whole presentation description should be sent
   again, rather than just the additional components, so that components
   can be deleted.

   Example:


     C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Content-Type: application/sdp
           Content-Length: 332

           v=0
           o=mhandley 2890844526 2890845468 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



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           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

     S->C: RTSP/1.0 200 OK
           CSeq: 312



10.4 SETUP

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can issue a SETUP request for a
   stream that is already playing 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). For the benefit of any
   intervening firewalls, a client must indicate the transport parame-
   ters 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 ini-
     tialization.

   The Transport header specifies the transport parameters acceptable to
   the client for data transmission; the response will contain the
   transport parameters selected by the server.


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

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;
             client_port=4588-4589;server_port=6256-6257






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   The server generates session identifiers in response to SETUP
   requests. If a SETUP request to a server includes a session identi-
   fier, the server MUST bundle this setup request into the existing
   session or return error 459 (Aggregate Operation Not Allowed) (see
   Section 11.4.10).

10.5 PLAY

   The PLAY method tells the server to start sending data via the mecha-
   nism specified in SETUP. A client MUST NOT issue a PLAY request until
   any outstanding SETUP requests have been acknowledged as successful.

   The PLAY request positions the normal play time to the beginning of
   the range specified and delivers stream data until the end of the
   range is reached. PLAY requests may be pipelined (queued); a server
   MUST queue PLAY requests to be executed in order. That is, a PLAY
   request arriving while a previous PLAY request is still active is
   delayed until the first has been completed.

     This allows precise editing.  For example, regardless of how
     closely spaced the two PLAY requests in the example below
     arrive, the server 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

     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 836
           Session: 12345678
           Range: npt=20-25

     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 837
           Session: 12345678
           Range: npt=30-



   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal. It starts playing a
   stream from the beginning unless the stream has been paused.  If a
   stream has been paused via PAUSE, stream delivery resumes at the



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   pause point.

   The Range header may also contain a time parameter.  This parameter
   specifies a time in UTC at which the playback should start. If the
   message is received after the specified time, playback is started
   immediately. The time parameter may be used to aid in synchronization
   of streams obtained from different sources.

   For a on-demand stream, the server replies with the actual range that
   will be played back. This may differ from the requested range if
   alignment of the requested range to valid frame boundaries is
   required for the media source. If no range is specified in the
   request, the current position is returned in the reply. The unit of
   the range in the reply is the same as that in the request.

   After playing the desired range, the presentation is automatically
   paused, as if a PAUSE request had been issued.

   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip. The playback is to start
   at 15:36 on 23 Jan 1997.


     C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
           CSeq: 833
           Session: 12345678
           Range: smpte=0:10:20-;time=19970123T153600Z

     S->C: RTSP/1.0 200 OK
           CSeq: 833
           Date: 23 Jan 1997 15:35:06 GMT
           Range: smpte=0:10:22-;time=19970123T153600Z
           RTP-Info:url=rtsp://audio.example.com/twister.en;seq=14783;rtptime=2345962545



   For playing back a recording of a live presentation, it may be desir-
   able 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



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           Range: clock=19961108T142300Z-19961108T143520Z
           RTP-Info:url=rtsp://audio.example.com/meeting.en;seq=53745;rtptime=484589019




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

   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.

10.6 PAUSE

   The PAUSE request causes the stream delivery to be interrupted
   (halted) temporarily. If the request URL names a stream, only play-
   back and recording of that stream is halted. For example, for audio,
   this is equivalent to muting. If the request URL names a presentation
   or group of streams, delivery of all currently active streams within
   the presentation or group is halted. After resuming playback or
   recording, 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



   The PAUSE request may contain a Range header specifying when the      |
   stream or presentation is to be halted. We refer to this point as the |
   "pause point". The 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 nor-  |
   mal play time:                                                        |




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     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                                |
           Range: npt=10-30                                              |
           RTP-Info:url=rtsp://example.com/fizzle/foo/audiotrack;seq=5712;rtptime=934207921,|
                   url=rtsp://example.com/fizzle/foo/videotrack;seq=57654;rtptime=2792482193|

     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                                |
           Range: npt=21-                                                |



   The normal play time for the stream is set to the pause point. The
   pause request becomes effective the first time the server is encoun-
   tering the time point specified in any of the currently pending PLAY
   requests. If the Range header specifies a time outside any currently
   pending PLAY requests, the error 457 (Invalid Range) is returned. If
   a media unit (such as an audio or video frame) starts presentation at
   exactly the pause point, it is not played or recorded. 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 nor-
   mal play time.

   A PAUSE request discards all queued PLAY requests. However, the pause
   point in the media stream MUST be maintained. A subsequent PLAY
   request without Range header resumes from the pause point.

   For example, if the server has play requests 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 stops after completing the first
   play request and discards the second play request.





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   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 PLAY request effectively being ignored, assum-
   ing the PAUSE request arrives before the server has started playing
   the second, overlapping range. Regardless of when the PAUSE request
   arrives, it sets the NPT to 14.

   If the server has already sent data beyond the time specified in the
   Range header, a PLAY would still resume at that point in time, as it
   is assumed that the client has discarded data after that point. This
   ensures continuous pause/play cycling without gaps.

10.7 TEARDOWN

   The TEARDOWN request stops the stream delivery for the given URI,
   freeing the resources associated with it. If the URI is the presenta-
   tion URI for this presentation, any RTSP session identifier associ-
   ated with the session is no longer valid. Unless all transport param-
   eters are defined by the session description, a SETUP request has to
   be issued before the session can be played again.

   A server that after processing the TEARDOWN still has a valid session
   MUST in the response return a session header.

   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



10.8 GET_PARAMETER

   The GET_PARAMETER request retrieves the value of a parameter of a
   presentation or stream specified in the URI. The content of the reply
   and response is left to the implementation. GET_PARAMETER with no
   entity body may be used to test client or server liveness ("ping").

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0



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           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. This method is intentionally loosely defined with
     the intention that the reply content and response content will
     be defined after further experimentation.

10.9 SET_PARAMETER

   This method requests to set the value of a parameter for a presenta-
   tion or stream specified by the URI.

   A request SHOULD 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 parame-
   ter to be set repeatedly to the same value, but it MAY disallow
   changing parameter values.

   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



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     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




     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.

10.10 REDIRECT

   A redirect request informs the client that it must connect to another
   server location. It contains the mandatory header Location, which
   indicates that the client should issue requests for that URL. It may
   contain the parameter Range, which indicates when the redirection
   takes effect. If the client wants to continue to send or receive
   media for this URI, the client MUST issue a TEARDOWN request for the
   current session and a SETUP for the new session at the designated
   host.

   This example request redirects traffic for this URI to the new server
   at the given play time:


     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 732
           Location: rtsp://bigserver.com:8001
           Range: clock=19960213T143205Z-





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10.11 RECORD

   This method initiates recording a range of media data according to
   the presentation description. The timestamp reflects start and end
   time (UTC). If no time range is given, use the start or end time pro-
   vided in the presentation description. If the session has already
   started, commence recording immediately.

   The server decides whether to store the recorded data under the
   request-URI or another URI. If the server does not use the request-
   URI, the response SHOULD be 201 (Created) and contain an entity which
   describes the status of the request and refers to the new resource,
   and a Location header.

   A media server supporting recording of live presentations MUST sup-
   port the clock range format; the smpte format does not make sense.

   In this example, the media server was previously invited to the con-
   ference indicated.


     C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
           CSeq: 954
           Session: 12345678
           Conference: 128.16.64.19/32492374




     Note: this example needs work, or needs to be removed.


10.12 PING                                                               |

   This method is a bi-directional mechanism for server or client live-  |
   ness checking. It has no side effects.                                |

   Prior to using this method, an OPTIONS method MUST 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.             |

   When a proxy is in use, PING with a * indicates a single-hop liveness |
   check, whereas PING with a URL indicates an end-to-end liveness       |
   check.                                                                |

   Example:                                                              |

     C->S: PING * RTSP/1.0                                               |



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           CSeq: 123                                                     |

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



10.13 Embedded (Interleaved) Binary Data

   Certain firewall designs and other circumstances may force a server
   to interleave RTSP methods and stream data. This interleaving should
   generally be avoided unless necessary since it complicates client and
   server operation and imposes additional overhead. 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, fol-
   lowed by the length of the encapsulated binary data as a binary, two-
   byte integer in network byte order. The stream data follows immedi-
   ately afterwards, without a CRLF, but including the upper-layer pro-
   tocol headers. Each $ block contains exactly one upper-layer protocol
   data unit, e.g., one RTP packet.

   The channel identifier is defined in the Transport header with the
   interleaved parameter(Section 12.40).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection. As a default, RTCP packets are
   sent on the first available channel higher than the RTP channel. The
   client MAY explicitly request RTCP packets on another channel. This
   is done by specifying two channels in the interleaved parameter of
   the Transport header(Section 12.40).


     RTCP is needed for synchronization when two or more streams
     are interleaved in such a fashion. Also, this provides a con-
     venient 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://foo.com/bar.file RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP/TCP;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Date: 05 Jun 1997 18:57:18 GMT



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           Transport: RTP/AVP/TCP;interleaved=0-1
           Session: 12345678

     C->S: PLAY rtsp://foo.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://foo.com/bar.file;
             seq=232433;rtptime=972948234

     S->C: $000{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $000{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $001{2 byte length}{"length" bytes  RTCP packet}



11 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 1 for a
   listing of which status codes may be returned by which requests.

11.1 Success 2xx

11.1.1 250 Low on Storage Space

   The server returns this warning after receiving a RECORD request that
   it may not be able to fulfill completely due to insufficient storage
   space. If possible, the server should use the Range header to indi-
   cate what time period it may still be able to record. Since other
   processes on the server may be consuming storage space simultane-
   ously, a client should take this only as an estimate.

11.2 Redirection 3xx

   See [H10.3].

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

11.3 Client Error 4xx




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11.4 400 Bad Request

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

11.4.1 405 Method Not Allowed

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

11.4.2 451 Parameter Not Understood

   The recipient of the request does not support one or more parameters
   contained in the request.

11.4.3 452 reserved

   This error code was removed from RFC 2326 [21] and is obsolete.

11.4.4 453 Not Enough Bandwidth

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

11.4.5 454 Session Not Found

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

11.4.6 455 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.

11.4.7 456 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.



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11.4.8 457 Invalid Range

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

11.4.9 458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not modi-
   fied.

11.4.10 459 Aggregate Operation Not Allowed

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

11.4.11 460 Only Aggregate Operation Allowed

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

11.4.12 461 Unsupported Transport

   The Transport field did not contain a supported transport specifica-
   tion.

11.4.13 462 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 param-
   eter in the Transport field.

11.5 Server Error 5xx

11.5.1 551 Option not supported

   An option given in the Require or the Proxy-Require fields was not
   supported. The Unsupported header should be returned stating the
   option for which there is no support.

12 Header Field Definitions



   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       |



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   method        direction  object requirement acronym Body
   -----------------------------------------------------------
   DESCRIBE      C->S       P,S    recommended DES     r
   ANNOUNCE      C->S, S->C P,S    optional    ANN     R
   GET_PARAMETER C->S, S->C P,S    optional    GPR     R,r
   OPTIONS       C->S       P,S    required    OPT
                 S->C              optional
   PAUSE         C->S       P,S    recommended PSE
   PING          C->S, S->C P,S    optional    PNG
   PLAY          C->S       P,S    required    PLY
   RECORD        C->S       P,S    optional    REC
   REDIRECT      S->C       P,S    optional    RDR
   SETUP         C->S       S      required    STP
   SET_PARAMETER C->S, S->C P,S    optional    SPR     R,r?
   TEARDOWN      C->S       P,S    required    TRD




   Table 3: 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:  There  has  been some usage of the body to convey
   more information in error messages  for  responses  containing  error
   codes. Some error messages seem to mandate such usage.


   syntax, meaning, and usage.  Throughout this section, we use [HX.Y]   |
   to refer to Section X.Y of the current HTTP/1.1 specification RFC     |
   2616 [26].  Examples of each header field are given.                  |

   Information about header fields in relation to methods and proxy pro- |
   cessing is summarized in Table 4.                                     |

   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;                       |




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

   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:                                                         |

     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 must 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 are abbreviated according to table 3:        |

     c: Conditional; requirements on the header field depend on the con- |
          text 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.                                    |

     t: The header field SHOULD be sent, but clients/servers need to be  |
          prepared to receive messages without that header field.  If a  |
          stream-based protocol (such as TCP) is used as a transport,    |
          then the header field MUST be sent.                            |

     *: The header field is required if the message body is not empty.   |
          See sections 12.14, 12.16 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, and a Client/Server MAY ignore the header      |
   field if present in the request or response (The exception to this    |
   rule is the Require header field discussed in 12.32). 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         |



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   applicable" means that the header field MUST NOT be present in a      |
   request. If one is placed 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 a URI. If the  |
   URI contains a comma, or semicolon, the URI MUST be enclosed in dou-  |
   ble 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.                 |



12.1 Accept

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

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

   See [H14.1] for syntax.

   Example of use:

     Accept: application/rtsl, application/sdp;level=2



12.2 Accept-Encoding

   See [H14.3]

12.3 Accept-Language

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


12.4 Accept-Ranges                                                       |





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   Header              Where  Proxy DES OPT GPR SPR ANN STP PLY REC PSE TRD RDR PNG
   ---------------------------------------------------------------------------------
   Accept                R           o   -   -   -   -   -   -   -   -   -   -   -
   Accept-Encoding       R      r    o   -   -   -   -   -   -   -   -   -   -   -
   Accept-Language       R      r    o   -   -   -   -   -   -   -   -   -   -   -
   Accept-Ranges         r           -   -   -   -   -   -   o   -   -   -   -   -
   Allow                405          -   -   -   -   m   -   m   m   m   -   -   -
   Authorization         R           o   o   o   o   o   o   o   o   o   o   o   o
   Bandwidth             R           o   -   -   o   -   o   o   -   -   -   -   -
   Blocksize             R           o   -   -   o   -   o   o   -   -   -   -   -
   Cache-Control                r    -   -   -   -   -   o   -   -   -   -   -   -
   Connection                        o   o   o   o   o   o   o   o   o   o   o   -
   Content-Base          R           -   -   o   o   o   -   -   -   -   -   -   -
   Content-Base          r           o   -   o   o   -   -   -   -   -   -   -   -
   Content-Base         4xx          o   o   o   o   o   o   o   o   o   o   o   -
   Content-Encoding      R      r    -   -   o   o   o   -   -   -   -   -   -   -
   Content-Encoding      r      r    o   -   o   o   -   -   -   -   -   -   -   -
   Content-Encoding     4xx     r    o   o   o   o   o   o   o   o   o   o   o   -
   Content-Language      R      r    -   -   o   o   o   -   -   -   -   -   -   -
   Content-Language      r      r    o   -   o   o   -   -   -   -   -   -   -   -
   Content-Language     4xx     r    o   o   o   o   o   o   o   o   o   o   o   -
   Content-Length        R      r    -   -   *   *   *   -   -   -   -   -   -   -
   Content-Length        r      r    *   -   *   *   -   -   -   -   -   -   -   -
   Content-Length       4xx     r    *   *   *   *   *   *   *   *   *   *   *   -
   Content-Location      R           -   -   o   o   o   -   -   -   -   -   -   -
   Content-Location      r           o   -   o   o   -   -   -   -   -   -   -   -
   Content-Location     4xx          o   o   o   o   o   o   o   o   o   o   o   -
   Content-Type          R           -   -   *   *   *   -   -   -   -   -   -   -
   Content-Type          r           *   -   *   *   -   -   -   -   -   -   -   -
   Content-Type         4xx          *   *   *   *   *   *   *   *   *   *   *   -
   CSeq                 Rc           m   m   m   m   m   m   m   m   m   m   m   m
   Date                        am    o   o   o   o   o   o   o   o   o   o   o   o
   Expires               r      r    o   -   -   -   -   -   -   -   -   -   -   -
   From                  R      r    o   o   o   o   o   o   o   o   o   o   o   o
   Host                              o   o   o   o   o   o   o   o   o   o   o   o
   If-Match              R      r    -   -   -   -   -   o   -   -   -   -   -   -
   If-Modified-Since     R      r    o   -   -   -   -   o   -   -   -   -   -   -
   Last-Modified         R      r    -   -   -   -   o   -   -   -   -   -   -   -
   Last-Modified         r      r    o   -   o   -   -   -   -   -   -   -   -   -
   Location              R           -   -   -   -   -   -   -   -   -   -   m   -
   Location             3xx          m   -   -   -   -   m   -   -   -   -   -   -
   Proxy-Authenticate   407    amr   m   m   m   m   m   m   m   m   m   m   m   m
   Proxy-Require         R     ar    o   o   o   o   o   o   o   o   o   o   o   o
   Public                r    admr   -  m*   -   -   -   -   -   -   -   -   -   -
   Public               501   admr  m*  m*  m*  m*  m*  m*  m*  m*  m*  m*  m*  m*



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   Range                 R           -   -   -   -   -   -   o   -   o   -   o   -
   Range                 r           -   -   -   -   -   -  m*   -   -   -   -   -
   Referer               R           o   o   o   o   o   o   o   o   o   o   o   -
   Require               R           o   o   o   o   o   o   o   o   o   o   o   o
   Retry-After        3xx,503        o   o   o   o   -   o   -   -   -   -   -   -
   RTP-Info              r           -   -   -   -   -   -   m   -   -   -   -   -
   Scale                             -   -   -   -   -   -   o   o   -   -   -   -
   Session               R           -   o   o   o   m   o   m   m   m   m   m   m
   Session               r           -   c   c   c   m   m   m   m   m   o   m   m
   Server                R           -   o   o   o   o   -   -   -   -   -   o   o
   Server                r           o   o   o   o   o   o   o   o   o   o   -   o
   Speed                             -   -   -   -   -   -   o   -   -   -   -   -
   Supported             R           o   o   o   o   o   o   o   o   o   o   o   o
   Supported             r           c   c   c   c   c   c   c   c   c   c   c   c
   Timestamp             R           o   o   o   o   o   o   o   o   o   o   o   o
   Timestamp             c           m   m   m   m   m   m   m   m   m   m   m   m
   Transport                         -   -   -   -   -   m   -   -   -   -   -   -
   Unsupported           r           c   c   c   c   c   c   c   c   c   c   c   c
   User-Agent            R          m*  m*  m*  m*  m*  m*  m*  m*  m*  m*   -   -
   User-Agent            r           -   -   -   -   -   -   -   -   -   -  m*   -
   Vary                  r           c   c   c   c   c   c   c   c   c   c   -   -
   Via                   R     amr   o   o   o   o   o   o   o   o   o   o   o   o
   Via                   c     dr    m   m   m   m   m   m   m   m   m   m   m   m
   WWW-Authenticate     401          m   m   m   m   m   m   m   m   m   m   m   m
   ---------------------------------------------------------------------------------
   Header              Where  Proxy DES OPT GPR SPR ANN STP PLY REC PSE TRD RDR PNG



   Table 4: Overview of RTSP header fields


12.5 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.

   Example of use:

     Allow: SETUP, PLAY, RECORD, SET_PARAMETER



12.6 Authorization




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   See [H14.8]

12.7 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 modem retraining.


   Bandwidth  =  "Bandwidth" ":" 1*DIGIT


   Example:

     Bandwidth: 4000



12.8 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 (416) if the value is syntactically invalid.


   Blocksize  =  "Blocksize" ":" 1*DIGIT


12.9 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 appli-
   cation, 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-direc-
   tive for a specific cache.





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   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, but rather of the stream identified by the
   SETUP request. Responses to RTSP requests are not cacheable, except
   for responses to DESCRIBE.


   Cache-Control             =  "Cache-Control" ":" 1#cache-directive
   cache-directive           =  cache-request-directive
                            |   cache-response-directive
   cache-request-directive   =  "no-cache"
                            |   "max-stale"
                            |   "min-fresh"
                            |   "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 ) ]


     no-cache: Indicates that the media stream MUST NOT be cached any-
          where. 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 link. Serious opera-
          tional 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 authenti-
          cation all depend on receiving a stream that is bit-for-bit
          identical to the original entity-body. Therefore, if a
          response includes the no-transform directive, an intermediate



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          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 net-
          work 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 con-
          straints of the request, or respond with a 504 (Gateway Time-
          out) 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 must do an end-to-end revalidation every
          time, if, based solely on the origin server's Expires, the
          cached response is stale.)

12.10 Connection

   See [H14.10]

12.11 Content-Base

   The Content-Base entity-header field may be used to specify the base
   URI for resolving relative URLs within the entity. This header field
   is described as Base in RFC 1808, which is expected to be revised.



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   Content-Base  =  "Content-Base" ":" absoluteURI


   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.

12.12 Content-Encoding

   See [H14.11]

12.13 Content-Language

   See [H14.12]

12.14 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].

12.15 Content-Location

   See [H14.14]

12.16 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.

12.17 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 num-
   ber. 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).






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   CSeq  =  "Cseq" ":" 1*DIGIT


12.18 Date

   See [H14.18].

12.19 Expires

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

     DESCRIBE response: The Expires header indicates a date and time
          after which the description 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 13 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:


   Expires  =  "Expires" ":" HTTP-date


   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.



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   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
   be non-cacheable indicates that the media stream is cacheable, unless
   indicated otherwise by a Cache-Control header field (Section 12.9).

12.20 From

   See [H14.22].

12.21 Host

   The Host HTTP request header field [H14.23] is not needed for RTSP.
   It should be silently ignored if sent.

12.22 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.

   The identifier is an opaque identifier, and thus is not specific to
   any particular session description language.

12.23 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 will be returned without any message-body.


   If-Modified-Since  =  "If-Modified-Since" ":" HTTP-date


   An example of the field is:


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



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12.24 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 or ANNOUNCE, the header field indicates the last modifica-
   tion date and time of the description, for SETUP that of the media
   stream.

12.25 Location

   See [H14.30].

12.26 Proxy-Authenticate

   See [H14.33].

12.27 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-sen-
   sitive 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 if not supported.
   Servers should treat this field identically to the Require field.

   See Section 12.32 for more details on the mechanics of this message
   and a usage example.

12.28 Public

   The Public response-header field lists the set of methods supported
   by the server. The purpose of this field is strictly to inform the
   recipient of the capabilities of the server regarding unusual meth-
   ods. 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.


        Public  =  "Public" ":" 1#method

   Example of use:

      Public: OPTIONS, MGET, MHEAD, GET, HEAD



   This header field applies only to the server directly connected to
   the client (i.e., the nearest neighbor in a chain of connections).



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   If the response passes through a proxy, the proxy MUST either remove
   the Public header field or replace it with one applicable to its own
   capabilities.

12.29 Range

   The Range request and response header field specifies a range of
   time. The range can be specified in a number of units.  This specifi-
   cation 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
   not meaningful and MUST NOT be used. The header may also contain a
   time parameter in UTC, specifying the time at which the operation is
   to be made effective. Servers supporting the Range header MUST under-
   stand the NPT range format and SHOULD understand the SMPTE range for-
   mat. The Range response header indicates what range of time is actu-
   ally being played or recorded. If the Range header is given in a time
   format that is not understood, the recipient should return 501 (Not
   Implemented).

   Ranges are half-open intervals, including the lower point, but
   excluding the upper point. In other words, a range of a-b starts
   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.


   Range             =  "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
   ranges-specifier  =  npt-range | utc-range | smpte-range


   Example:

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




     The notation is similar to that used for the HTTP/1.1 [26]
     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.



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12.30 Referer

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

12.31 Retry-After

   See [H14.37].

12.32 Require

   The Require request-header field is used by clients to query the
   server about options that it may or may not support. The server MUST
   respond to this header by using the Unsupported header to negatively
   acknowledge those options which are NOT supported.


     This is to make sure that the client-server interaction will
     proceed without delay when all options are understood by both
     sides, and only slow down if options are not understood (as in
     the case above).  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.


   Require  =  "Require" ":" 1#option-tag


   Example:

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Require: funky-feature
           Funky-Parameter: funkystuff

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

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 303

   S->C:   RTSP/1.0 200 OK
           CSeq: 303





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   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 trans-
   mitted with every exchange.

   Proxies and other intermediary devices SHOULD ignore features that
   are not understood in this field. If a particular extension requires
   that intermediate devices support it, the extension should be tagged
   in the Proxy-Require field instead (see Section 12.27).

12.33 RTP-Info

   The RTP-Info response-header field is used to set RTP-specific param-
   eters in the PLAY response.

     url: Indicates the stream URL which for which the following RTP
          parameters correspond.

     seq: Indicates the sequence number of the first packet of the
          stream. 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.

     rtptime: Indicates the RTP timestamp corresponding to the time
          value in the Range response header. (Note: For aggregate con-
          trol, 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 indi-
          cated 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 times-
          tamps to NPT. Furthermore, in order to ensure that this
          information is available at the necessary time (immedi-
          ately at startup or after a seek), and that it is deliv-
          ered reliably, this mapping is placed in the RTSP control
          channel.

          In order to compensate for drift for long, uninterrupted pre-
          sentations, RTSP clients should additionally map NPT to NTP,
          using initial RTCP sender reports to do the mapping, and later



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          reports to check drift against the mapping.

   Syntax:

   RTP-Info        =  "RTP-Info" ":" 1#rtsp-info-spec
   rtsp-info-spec  =  stream-url 1*parameter
   stream-url      =  quoted-url | unquoted-url
   unquoted-url    =  "url" "=" safe-url
                  |   ";" "mode" = <"> 1#Method <">
   quoted-url      =  "url" "=" <"> needquote-url <">
   safe-url        =  url
   needquote-url   =  url
   url             =  ( absoluteURI | relativeURI )
   parameter       =  ";" "seq" "=" 1*DIGIT
                  |   ";" "rtptime" "=" 1*DIGIT


   Additional constraint: safe-url MUST NOT contain the semicolon (";")
   or comma (",") characters. The quoted-url form SHOULD only be used
   when a URL does not meet the safe-url constraint, in order to ensure
   compatibility with implementations conformant to RFC 2326 [21].

   absoluteURI and relativeURI are defined in RFC 2396 [22].

   Example:

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



12.34 Scale

   A scale value of 1 indicates normal play or record at the normal for-
   ward 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.

   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 frag-
   ments of audio.



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   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 request contains a Range parameter, the new scale value will
   take effect at that time.


   Scale  =  "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]


   Example of playing in reverse at 3.5 times normal rate:


     Scale: -3.5



12.35 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.  Implementa-
   tion 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. If the request contains a Range parameter,
   the new speed value will take effect at that time.


   Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]


   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 presen-
   tation at a higher or lower rate is necessary. Implementors should
   keep in mind that bandwidth for the session may be negotiated before-
   hand (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.



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12.36 Server

   See [H14.38]

12.37 Session

   The Session request-header and response-header field identifies an
   RTSP session started by the media server in a SETUP response and con-
   cluded by TEARDOWN on the presentation URL. The session identifier is
   chosen by the media server (see Section 3.3) and MUST be returned in
   the SETUP response. Once a client receives a Session identifier, it
   MUST return it for any request related to that session.


   Session  =  "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]


   The timeout parameter is only allowed in a response header.  The
   server uses it to indicate to the client how long the server is pre-
   pared to wait between RTSP commands before closing the session due to
   lack of activity (see Section A). The timeout is measured in seconds,
   with a default of 60 seconds (1 minute).

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections. Control messages for more than one
   RTSP URL may be sent within a single RTSP session. Hence, it is pos-
   sible that clients use the same session for controlling many streams
   constituting a presentation, as long as all the streams come from the
   same server. (See example in Section 14). However, multiple "user"
   sessions for the same URL from the same client MUST use different
   session identifiers.

     The session identifier is needed to distinguish several deliv-
     ery requests for the same URL coming from the same client.

   The response 454 (Session Not Found) is returned if the session iden-
   tifier is invalid.


12.38 Supported                                                          |

   The Supported header field enumerates all the extensions supported by |
   the client or server. When offered in a request, the receiver MUST    |
   respond with its cooresponding Supported header.                      |

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




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   Example:                                                              |


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

   SuppoS->C:RTSP/1.0e200 OKz                                               ||



12.39 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.


   Timestamp  =  "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
   delay      =  *(DIGIT) [ "." *(DIGIT) ]


12.40 Transport

   The Transport request-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 Transport header field MAY also be used to change certain trans-
   port parameters. A server MAY refuse to change parameters of an
   existing stream.

   The server MAY return a Transport response-header field in the
   response to indicate the values actually chosen.

   A Transport request header field may contain a list of transport
   options acceptable to the client, in the form of multiple transport-
   spec entries. In that case, the server MUST return a single option
   (transport-spec) which was actually chosen.



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   A transport-spec transport option may only contain one of any given   |
   parameter within it. Parameters may be given in any order.  Addition- |
   ally, it may only contain the unicast or multicast transport parame-  |
   ter.


     The Transport header field is restricted to describing a sin-
     gle RTP 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.

   Below are the configuration parameters associated with transport:

   General parameters:

     unicast | multicast: This parameter is a mutually exclusive indica-
          tion 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 trans-
          mission MUST indicate such capability by including two full
          transport-specs with separate parameters for each.


     destination: The address to which a stream will be sent.  The       |
          client may specify the destination address with the destina-   |
          tion parameter. To avoid becoming the unwitting perpetrator of |
          a remote-controlled denial-of-service attack, a server SHOULD  |
          authenticate the client and SHOULD log such attempts before    |
          allowing the client to direct a media stream to an address not |
          chosen by the server. This is particularly important if RTSP   |
          commands are issued via UDP, but implementations cannot rely   |
          on TCP as reliable means of client identification by itself.

     source: If the source address for the stream is different than can
          be derived from the RTSP endpoint address (the server in play-
          back or the client in recording), the source address MAY be
          specified.





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          This information may also be available through SDP.  How-
          ever, 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.

     mode: The mode parameter indicates the methods to be supported for
          this session. Valid values are PLAY and RECORD. If not pro-
          vided, the default is PLAY.

     append: If the mode parameter includes RECORD, the append parameter
          indicates that the media data should append to the existing
          resource rather than overwrite it.  If appending is requested
          and the server does not support this, it MUST refuse the
          request rather than overwrite the resource identified by the
          URI. The append parameter is ignored if the mode parameter
          does not contain RECORD.

     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 10.13. The argument provides the channel number to be
          used in the $ statement. 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.


          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:

     port: This parameter provides the RTP/RTCP port pair for a multi-
          cast session. It is 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





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     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

     ssrc: The ssrc parameter indicates the RTP SSRC [23] value that
          should be (request) or will be (response) used by the media
          server. This parameter is only valid for unicast transmission.
          It identifies the synchronization source to be associated with
          the media stream, and is expressed as an eight digit hexideci-
          mal value.


   Transport           =  "Transport" ":" 1#transport-spec
   transport-spec      =  transport-id *parameter
   transport-id        =  transport-protocol "/" profile ["/" lower-transport]
                          ; no LWS is allowed inside transport-id
   transport-protocol  =  "RTP" | token
   profile             =  "AVP" | token
   lower-transport     =  "TCP" | "UDP" | token
   parameter           =  ";" ( "unicast" | "multicast" )
                      |   ";" "source" [ "=" address ]
                      |   ";" "destination" [ "=" address ]
                      |   ";" "interleaved" "=" channel [ "-" channel ]
                      |   ";" "append"
                      |   ";" "ttl" "=" ttl
                      |   ";" "layers" "=" 1*DIGIT
                      |   ";" "port" "=" port [ "-" port ]
                      |   ";" "client_port" "=" port [ "-" port ]
                      |   ";" "server_port" "=" port [ "-" port ]
                      |   ";" "source" "=" address
                      |   ";" "ssrc" "=" ssrc
                      |   ";" "mode" "=" mode-spec
   ttl                 =  1*3(DIGIT)
   port                =  1*5(DIGIT)
   ssrc                =  8*8(HEX)
   channel             =  1*3(DIGIT)
   address             =  host
   mode-spec           =  <"> 1#mode <"> | mode
   mode                =  "PLAY" | "RECORD" | token


   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



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           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

     S->C: RTSP/1.0 200 OK
           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"



12.41 Unsupported

   The Unsupported response-header field lists the features not sup-
   ported by the server. In the case where the feature was specified via
   the Proxy-Require field (Section 12.32), if there is a proxy on the
   path between the client and the server, the proxy MUST insert a
   response message with a status code of 551 (Option Not Supported).

   See Section 12.32 for a usage example.


   Unsupported  =  "Unsupported" ":" 1#option-tag


12.42 User-Agent

   See [H14.43]

12.43 Vary

   See [H14.44]

12.44 Via

   See [H14.45].

12.45 WWW-Authenticate

   See [H14.47].

13 Caching

   In HTTP, response-request pairs are cached. RTSP differs signifi-
   cantly in that respect. Responses are not cacheable, with the excep-
   tion of the presentation description returned by DESCRIBE or included



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   with ANNOUNCE. (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.

   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 issu-
   ing 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 possi-
   bly making a local copy for later reuse. The exact behavior allowed
   to the cache is given by the cache-response directives described in
   Section 12.9. A cache MUST answer any DESCRIBE requests if it is cur-
   rently 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 addi-
   tional 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.

14 Examples

   The following examples refer to stream description formats that are
   not standards, such as RTSL. The following examples are not to be
   used as a reference for those formats.

14.1 Media on Demand (Unicast)




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   Client C requests a movie from media servers A (audio.example.com )
   and V (video.example.com ). The media description is stored on a web
   server W. The media description contains descriptions of the presen-
   tation and all its streams, including the codecs that are available,
   dynamic RTP payload types, the protocol stack, and content informa-
   tion 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
         Content-Type: application/sdp

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         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
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057

   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001

   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059

   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0



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         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://video.example.com/twister/video;
        seq=12312232;rtptime=78712811

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

   A->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
        seq=876655;rtptime=1032181

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

   A->C: RTSP/1.0 200 OK
         CSeq: 3

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

   V->C: RTSP/1.0 200 OK
         CSeq: 3



   Even though the audio and video track are on two different servers,
   and 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.

14.2 Streaming of a Container file





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   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
   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 con-
   trol all the streams via a single control message using an aggregate
   URL.

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URLs.

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


   C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 164

         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         a=control:rtsp://foo/twister
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://foo/twister/audio
         m=video 0 RTP/AVP 26
         a=control:rtsp://foo/twister/video




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   C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001;
                    server_port=9000-9001
         Session: 12345678

   C->M: SETUP rtsp://foo/twister/video RTSP/1.0
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003;
                    server_port=9004-9005
         Session: 12345678

   C->M: PLAY rtsp://foo/twister RTSP/1.0
         CSeq: 4
         Range: npt=0-
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 4
         Session: 12345678
         RTP-Info: url=rtsp://foo/twister/video;
        seq=9810092;rtptime=3450012

   C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
         CSeq: 5
         Session: 12345678

   M->C: RTSP/1.0 460 Only aggregate operation allowed
         CSeq: 5

   C->M: PAUSE rtsp://foo/twister RTSP/1.0
         CSeq: 6
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: 12345678

   C->M: SETUP rtsp://foo/twister RTSP/1.0



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         CSeq: 7
         Transport: RTP/AVP;unicast;client_port=10000

   M->C: RTSP/1.0 459 Aggregate operation not allowed
         CSeq: 7




   In the first instance of failure, the client tries to pause one
   stream (in this case video) of the presentation. This is disallowed
   for that presentation by the server. In the second instance, the
   aggregate URL may not be used for SETUP and one control message is
   required per stream to set up transport parameters.

     This keeps the syntax of the Transport header simple and
     allows easy parsing of transport information by firewalls.

14.3 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 descrip-
   tion for request URLs, rather than assuming that a consistent URL may
   always be used throughout. Here's 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

       S->C  RTSP/1.0 200 OK
             CSeq: 1
             Content-base: rtsp://foo.com/test.wav/
             Content-type: application/sdp
             Content-length: 48

             v=0
             o=- 872653257 872653257 IN IP4 172.16.2.187
             s=mu-law wave file
             i=audio test
             t=0 0
             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;



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                        client_port=6970-6971;mode="PLAY"
             CSeq: 2

       S->C  RTSP/1.0 200 OK
             Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
                        server_port=6970-6971;mode="PLAY"
             CSeq: 2
             Session: 2034820394

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

       S->C  RTSP/1.0 200 OK
             CSeq: 3
             Session: 2034820394
             RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
               seq=981888;rtptime=3781123



   Note the different URL in the SETUP command, and then the switch back
   to the aggregate URL 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.

   In this special case, it is recommended that servers be forgiving of
   implementations that send:


       C->S  PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
             CSeq: 3



   In the worst case, servers should send back:


       S->C  RTSP/1.0 460 Only aggregate operation allowed
             CSeq: 3



   One would also hope that server implementations are also forgiving of
   the following:





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       C->S  SETUP rtsp://foo.com/test.wav RTSP/1.0
             Transport: rtp/avp/udp;client_port=6970-6971;mode="PLAY"
             CSeq: 2



   Since there is only a single stream in this file, it's not ambiguous
   what this means.

14.4 Live Media Presentation Using Multicast

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


   C->W: GET /concert.sdp HTTP/1.1
         Host: www.example.com

   W->C: HTTP/1.1 200 OK
         Content-Type: application/x-rtsl

         <session>
           <track src="rtsp://live.example.com/concert/audio">
         </session>

   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 44

         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

   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
         Transport: RTP/AVP;multicast;destination=224.2.0.1;



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                    port=3456-3457;ttl=16
         Session: 0456804596

   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
         Session: 0456804596



14.5 Recording

   The conference participant client C asks the media server M to record
   the audio and video portions of a meeting. The client uses the
   ANNOUNCE method to provide meta-information about the recorded ses-
   sion to the server.



   C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
         CSeq: 90
         Content-Type: application/sdp
         Content-Length: 121

         v=0
         o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
         s=IETF Meeting, Munich - 1
         i=The thirty-ninth IETF meeting will be held in Munich, Germany
         u=http://www.ietf.org/meetings/Munich.html
         e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
         p=IETF Channel 1 +49-172-2312 451
         c=IN IP4 224.0.1.11/127
         t=3080271600 3080703600
         a=tool:sdr v2.4a6
         a=type:test
         m=audio 21010 RTP/AVP 5
         c=IN IP4 224.0.1.11/127
         a=ptime:40
         m=video 61010 RTP/AVP 31
         c=IN IP4 224.0.1.12/127

   M->C: RTSP/1.0 200 OK
         CSeq: 90

   C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0



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         CSeq: 91
         Transport: RTP/AVP;multicast;destination=224.0.1.11;
                    port=21010-21011;mode=record;ttl=127

   M->C: RTSP/1.0 200 OK
         CSeq: 91
         Session: 50887676
         Transport: RTP/AVP;multicast;destination=224.0.1.11;
                    port=21010-21011;mode=record;ttl=127

   C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
         CSeq: 92
         Session: 50887676
         Transport: RTP/AVP;multicast;destination=224.0.1.12;
                    port=61010-61011;mode=record;ttl=127

   M->C: RTSP/1.0 200 OK
         CSeq: 92
         Transport: RTP/AVP;multicast;destination=224.0.1.12;
                    port=61010-61011;mode=record;ttl=127

   C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
         CSeq: 93
         Session: 50887676
         Range: clock=19961110T1925-19961110T2015

   M->C: RTSP/1.0 200 OK
         CSeq: 93



15 Syntax

   The RTSP syntax is described in an augmented Backus-Naur form (BNF)
   as used in RFC 2068 [2].

15.1 Base Syntax


   OCTET           =  <any 8-bit sequence of data>
   CHAR            =  <any US-ASCII character (octets 0 - 127)>
   UPALPHA         =  <any US-ASCII uppercase letter "A".."Z">
   LOALPHA         =  <any US-ASCII lowercase letter "a".."z">
   ALPHA           =  UPALPHA | LOALPHA
   DIGIT           =  <any US-ASCII digit "0".."9">
   CTL             =  <any US-ASCII control character
                      (octets 0 - 31) and DEL (127)>




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   CR              =  <US-ASCII CR, carriage return (13)>
   LF              =  <US-ASCII LF, linefeed (10)>
   SP              =  <US-ASCII SP, space (32)>
   HT              =  <US-ASCII HT, horizontal-tab (9)>
   <">             =  <US-ASCII double-quote mark (34)>
   BACKSLASH       =  <US-ASCII backslash (92)>
   CRLF            =  CR LF
   LWS             =  [CRLF] 1*( SP | HT )
   TEXT            =  <any OCTET except CTLs>
   tspecials       =  "(" | ")" | "<" | ">" | "@"
                  |   "," | ";" | ":" | BACKSLASH | <">
                  |   "/" | "[" | "]" | "?" | "="
                  |   "{" | "}" | SP | HT
   token           =  1*<any CHAR except CTLs or tspecials>
   quoted-string   =  ( <"> *(qdtext) <"> )
   qdtext          =  <any TEXT except <">>
   quoted-pair     =  BACKSLASH CHAR
   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>
   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


16 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:

     Authentication Mechanisms: RTSP and HTTP share common authentica-
          tion schemes, and thus should follow the same prescriptions
          with regards to authentication. See chapter 15.1 of [2] for
          client authentication issues, and chapter 15.2 of [2] for
          issues regarding support for multiple authentication mecha-
          nisms.





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     Abuse of Server Log Information: RTSP and HTTP servers will presum-
          ably 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 sensi-
          tive 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 URLs are opaque
          handles that do not necessarily have file system semantics, it
          is anticipated that many implementations will translate por-
          tions of the request URLs directly to file system calls. In
          such cases, file systems SHOULD follow the precautions out-
          lined in [H15.5], such as checking for ".." in path compo-
          nents.

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

     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 should be fol-
          lowed.

     DNS Spoofing: Presumably, given the longer connection times typi-
          cally 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 one another, then it must
          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 invali-
          date resources over which they have no authority. ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [26], as of this writing) and also of the previous RFC2068



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   [2], 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-ini-
          tiated 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 relation between a transport
          layer connection and an RTSP session, it is possible for a
          malicious client to issue requests with random session identi-
          fiers which would affect unsuspecting clients. The server
          SHOULD use a large, random and non-sequential session identi-
          fier to minimize the possibility of this kind of attack.

     Authentication: Servers SHOULD implement both basic and digest [6]
          authentication. In environments requiring tighter security for
          the control messages, transport layer mechanisms such as TLS
          (RFC 2246 [27]) SHOULD be used.

     Stream issues: RTSP only provides for stream control. Stream deliv-
          ery 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.

17 IANA Considerations




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   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 shall contain, what specification is needed when adding a entry
   with IANA, and finally the entries that this document needs to regis-
   ter. See also the section 1.5 "Extending RTSP".

   The sections describing how to register an item uses some of the
   requirements level described in RFC 2434 [29], namely " First Come,
   First Served", "Specification Required", and "Standards Action".

   A registration request to IANA MUST contain the following informa-
   tion:

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

     + Indication of who has change control over the option (for exam-
       ple, IETF, ISO, ITU-T, other international standardization bod-
       ies, a consortium or a particular company or group of companies);

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

     + For proprietary options, contact information (postal and email
       address);

17.1 Option-tags

17.1.1 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 con-
   tains named entries representing certain functionality.

   The usage of option-tags is explained in section 3.7 and 10.1.

17.1.2 Registering New Option Tags with IANA

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

   The name of the option 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, control characters or periods. Any
   proprietary option SHOULD have as the first part of the name a vendor



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   tag, which identifies the company/person.

17.1.3 Registered entries

   The following options 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.

     record-basic: The minimal implementation for record operations
          according to section D.

     play-setup: The use of teardown and setup in play state.

     record-setup: The use of setup and teardown in record state.

17.2 RTSP Methods

17.2.1 Description

   What a method is, is described in section 10.  Extending the protocol
   with new methods allow for totally new functionality.

17.2.2 Registering New Methods with IANA

   A new method can only be registered through an IETF standards action.
   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:

     + A method name which follows the BNF rules for methods.

     + 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.

     + A list or table specifying which of the registered headers that
       are allowed to use with the method in request or/and response.

     + Describe how the method relates to network proxies.

17.2.3 Registered entries




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   This specification, RFCXXXX, registers 12 methods: DESCRIBE,
   ANNOUNCE, GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, RECORD, REDI-
   RECT, SETUP, SET_PARAMETER, and TEARDOWN.

17.3 RTSP Headers

17.3.1 Description

   By specifying new headers a method(s) can be enhanced in many differ-
   ent ways. An unknown header will be ignored by the receiving entity.
   If the new header is vital for a certain functionality, a option 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
   option tag.

   Unregistered headers SHALL have a name starting with "X-" to signal
   that it is a experimental header.

17.3.2 Registering New Headers with IANA

   A specification is required to register a header.

   The specification MUST contain the following information:

     + The header name following the BNF definition.

     + A BNF specification of how information (if any) is carried in the
       header.

     + A list or table specifying when the header may be used, encom-
       passing all methods, their request or response, the direction
       (C->S or S->C).

     + How the header shall be handled by proxies.

     + A description of the purpose of the header.

17.3.3 Registered entries

   All headers specified in section 12 in RFC XXXX are to be registered.

17.4 Parameters

17.4.1 Description

   A Parameter allow the counterpart to set something with the owner of
   the parameter. Both the client and the server can have parameters.




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17.4.2 Registering New Parameters with IANA

   Any Parameter is registered on a first come, first served basis.  The
   following rules apply for parameters:

     + The parameter name is a BNF token. The name SHOULD not be more
       than 20 characters long. Any proprietary parameter should start
       the name with a vendor tag, as clearly as possible identify the
       company or person.

     + Any non proprietary parameter MUST in the form of BNF specify
       what value types that are associated with the parameter.

17.4.3 Registered entries

   For the moment no known parameters are defined in RFC XXXX.

A RTSP Protocol State Machine

   The RTSP session state machine describe the behavior of the protocol
   from RTSP session initialization through RTSP session termination.

   State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier. The session may contain
   zero 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.

   This state machine is one possible representation that helps explain
   how the protocol works and when different requests are allowed.  We
   find it a reasonable representation but does not mandate it, and
   other representations can be created.

A.1 States

   The state machine contains five 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-nm: Ready state without any medias.

     Ready: Session is ready to start playing or recording.

     Play: Session is playing, i.e. sending media stream data in the
          direction S->C.




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     Record: Session is recording, i.e. sending media stream data in the
          direction C->S.

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
          variable is not mandated.

A.3 Abbreviations

   To make the state tables more compact a number of abbreviations are
   used, which are explained below.

     PP: Pause Point, the point in the presentation time line at which
          the presentation was paused.

     Prs: Presentation, the complete multimedia presentation.

     IFI: IF Implemented.

     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 shall change to.

   The response to valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column. The excep-
   tions shall be given a response according to the response column. If



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   the request does not meet the requisite, is erroneous or 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 3xx will result in that the session is ended and its state is
   changed to Init. 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 SHOULD result in ending of the session.

   The server will timeout the session after the period of time speci-
   fied 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 URL is equal to the media
   URL. For NRM>1 the presentation URL 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    3xx 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
   ANNOUNCE      C->S, IFI record.
   ANNOUNCE      S->C,             Update SES descr.


   Table 5: None state-machine changing events


   The methods in Table 5 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 6 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
   URL and/or server by a 3xx response.



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            Action  Requisite       New State  Response
            -------------------------------------------------
            SETUP                     Ready    NRM=1, RP=0.0
            SETUP   Needs Redirect    Init     3xx Redirect


   Table 6: State: Init

     Action         Requisite       New State  Response
     --------------------------------------------------------------
     SETUP                            Ready    NRM=1,RP=0.0
     SETUP          Needs Redirect    Init     3xx
     TEARDOWN       URL=*             Init     No session hdr.
     Timeout                          Init
     S->C:REDIRECT  Range hdr         Play     Set RedP
     S->C:REDIRECT  no range hdr      Init     Stop Media Playout
     RedP reached                     Init


   Table 7: State: Ready-nm


   The Ready-nm state has no media streams and therefore can't play or
   record. This state exist so that all session related parameters and
   resources can be kept while changing media stream(s). As seen in
   Table 7 the operations are limited to setting up a new media or tear-
   ing down the session. The established session can also be redirected
   with the REDIRECT method.


   In the Ready state, see Table 8, some of the actions are depending on
   the number of media streams 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
   URL) 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 either * or the pre-
   sentations URI the whole session is torn down. If a media URL is used
   in the TEARDOWN request the session will remain and a session header
   MUST be returned in the response. The number of media streams remain-
   ing after tearing down a media stream determines the new state.


   The Play state table, see Table 9, is the largest. The table contains
   an number of request that has presentation URL as a prerequisite on
   the request URL, this is due to the exclusion of non-aggregated
   stream control in sessions with more than one media stream.



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  Action         Requisite          New State  Response
  ---------------------------------------------------------------------
  SETUP          New URL              Ready    NRM+=1
  SETUP          Setten up URL        Ready    Change transport param.
  TEARDOWN       URL=*                Init     No session hdr
  TEARDOWN       Prs URL,NRM>1        Init     No session hdr
  TEARDOWN       md URL,NRM=1       Ready-nm   Session hdr, NRM=0
  TEARDOWN       md URL,NRM>1         Ready    Session hdr, NRM-=1
  PLAY           Prs URL, No range    Play     Play from RP
  PLAY           Prs URL, Range       Play     according to range
  RECORD                             Record
  S->C:REDIRECT  Range hdr            Ready    Set RedP
  S->C:REDIRECT  no range hdr         Init
  Timeout                             Init
  RedP reached                        Init


   Table 8: State: Ready


Action         Requisite          New State  Response
------------------------------------------------------------------------
PAUSE          PrsURL,No range      Ready    Set RP to present point
PAUSE          PrsURL,Range>now     Play     Set RP & PP to given point
PAUSE          PrsURL,Range<=now    Ready    Set RP to present pos.
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 URL,IFI          Play     NRM+=1, 200, RTP-Info
SETUP          New URL              Play     501
SETUP          Setuped URL          Play     Change transport param.
TEARDOWN       URL=*                Init     No session hdr
TEARDOWN       Prs URL,NRM>1        Init     No session hdr
TEARDOWN       md URL,NRM=1,IFI   Ready-nm   Session hdr
TEARDOWN       md URL,NRM>1,IFI     Play     Session hdr
TEARDOWN       md URL               Play     501
S->C:REDIRECT  Range hdr            Play     Set RedP
S->C:REDIRECT  no range hdr         Init     Stop Media Playout
RedP reached                        Init     Stop Media playout
Timeout                             Init


   Table 9: State: Play

   To avoid inconsistencies between the client and server, automatic
   state transitions are avoided. This can be seen at for example "End



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   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 auto-
   matic 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 transi-
   tions it must be prepared for getting error message as the state may
   or may not have changed.

   SETUP and TEARDOWN requests with media URLs in aggregated sessions
   may not be handled by the server as it is optional functionality. Use
   the service discovery mechanism with OPTIONS to find out in before-
   hand if the server implements it. If the functionality is not imple-
   mented but still tried by the client a "501 Not Implemented" response
   SHALL be received.


      Action         Requisite         New State  Response
      ------------------------------------------------------------
      PAUSE                              Ready
      Out-of-disc                       Record    Stop recording
      TEARDOWN       URL=*               Init     No session hdr
      TEARDOWN       Prs URL,NRM>1       Init     No session hdr
      TEARDOWN       md URL,NRM=1,IFI  Ready-nm   Session hdr
      TEARDOWN       md URL,NRM>1,IFI   Record    Session hdr
      TEARDOWN       md URL             Record    501
      S->C:REDIRECT  Range hdr          Record    Set RedP
      S->C:REDIRECT  w/o range hdr       Init     Stop Recording
      RedP reached                       Init     Stop Recording
      Timeout                            Init


   Table 10: State: Record


   The Record state Table 10 has only one event which is unique for this
   table, namely the "out-of-disc" event.  This event will happen if the
   recording server runs out of disc space. The state machine will
   remain in the Record state but the server will not be able to perform
   the actions related to the state.


     Something is needed to signal the client the fact that the
     server run out of disc space and not was capable of recording
     the data sent by the client.





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B Interaction with RTP

   RTSP allows media clients to control selected, non-contiguous sec-
   tions of media presentations, rendering those streams with an RTP
   media layer[23]. 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.

   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.  First we play NPT 10 through 15, then 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 40,000 through 55,200.


     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.

   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the above
   restriction. Combined RTSP/RTP media clients should use the RTP-Info
   field to determine whether incoming RTP packets were sent before or
   after a seek.

   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.

   For scaling (see Section 12.34), 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 12.35) of one, the
   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.

   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 12.33) header provides
   the first sequence number of the next segment.





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C Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 2327 [24]) may be used to
   describe streams or presentations in RTSP.  This description is typi-
   cally returned in reply to a DESCRIBE request on a URL from a server
   to a client, received via HTTP from a server to a client, or sent in
   an ANNOUNCE method from the client to the server.

   This appendix describes how an SDP file determines the operation of
   an RTSP session.  SDP 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).

C.1 Definitions

   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 [24]):

C.1.1 Control URL

   The "a=control:" attribute is used to convey the control URL. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URL to be used for con-
   trolling that particular media stream. If found at the session level,
   the attribute indicates the URL for aggregate control.

   Example:

     a=control:rtsp://example.com/foo



   This attribute may contain either relative and absolute URLs, follow-
   ing the rules and conventions set out in RFC 1808 [25].  Implementa-
   tions should look for a base URL in the following order:

     1.   the RTSP Content-Base field;

     2.   the RTSP Content-Location field;

     3.   the RTSP request URL.

   If this attribute contains only an asterisk (*), then the URL is
   treated as if it were an empty embedded URL, and thus inherits the
   entire base URL.




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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 synchro-
   nization. If the session is unicast, the port number serves as a rec-
   ommendation from the server to the client; the client still has to
   include it in its SETUP request and may ignore this recommendation.
   If the server has no preference, it SHOULD set the port number value
   to zero.

   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 pay-
   load type is a static payload type from RFC 1890 [1], no other infor-
   mation is required. In case it is a dynamic payload type, the media
   attribute "rtpmap" is used to specify what the media is.  The "encod-
   ing name" within the "rtpmap" attribute may be one of those specified
   in RFC 1890 (Sections 5 and 6), or an experimental encoding with a
   "X-" prefix as specified in SDP (RFC 2327 [24]). Codec-specific
   parameters are not specified in this field, but rather in the "fmtp"
   attribute described below.  Implementors seeking to register new
   encodings should follow the procedure in RFC 1890 [1]. If the media
   type is not suited to the RTP AV profile, then it is recommended that
   a new profile be created and the appropriate profile name be used in
   lieu of "RTP/AVP" in the "m=" field.

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 the packetization
   interval is conveyed using the "ptime" attribute.

C.1.5 Range of Presentation

   The "a=range" attribute defines the total time range of the stored
   session. (The length of live sessions can be deduced from the "t" and
   "r" parameters.) Unless the presentation contains media streams of
   different durations, the length attribute is a session-level
   attribute.  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.



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   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203



C.1.6 Time of Availability

   The "t=" field MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control. With
   aggregate 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. With non-aggregate control, the
   values should reflect the actual period for which the session is
   available in keeping with SDP semantics, and not depend on other
   means (such as the life of the web page containing the description)
   for this purpose.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. However, for on-demand unicast streams and some multicast
   streams, the destination address is specified by the client via the
   SETUP request. Unless the media content has a fixed destination
   address, the "c=" field is to be set to a suitable null value. For
   addresses of type "IP4", this value is "0.0.0.0".

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 12.22) 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.

   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




     One could argue that the "o=" field provides identical func-
     tionality. However, it does so in a manner that would put



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     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 and multiple
   media sections are specified, each section MUST have the control URL
   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
   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 URL 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 ini-
   tialization 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 con-
   trolled as a whole. In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is rela-
   tive, it is resolved to absolute URLs according to Section C.1.1
   above.

   If the presentation comprises only a single stream, the media-level
   "a=control:" attribute may be omitted altogether. However, if the
   presentation contains more than one stream, each media stream section
   MUST contain its own "a=control" attribute.



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   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I contain
   i=<more info>
   c=IN IP4 0.0.0.0
   t=0 0
   a=control:rtsp://example.com/movie/
   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 URLs rtsp://exam-
   ple.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set
   up the video and audio streams, respectively. The URL rtsp://exam-
   ple.com/movie/ controls the whole 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.

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

     + Generate the following requests: SETUP, TEARDOWN, and one of PLAY
       (i.e., a minimal playback client) or RECORD (i.e., a minimal
       recording client). If RECORD is implemented, ANNOUNCE MUST be
       implemented as well.

     + Include the following headers in requests: CSeq, Connection, Ses-
       sion, Transport. If ANNOUNCE is implemented, the capability to
       include headers Content-Language, Content-Encoding, Content-
       Length, and Content-Type should be as well.

     + Parse and understand the following headers in responses: CSeq,
       Connection, Session, Transport, Content-Language, Content-Encod-
       ing, Content-Length, Content-Type. If RECORD is implemented, the
       Location header must be understood as well. RTP-compliant imple-
       mentations should also implement RTP-Info.




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     + 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.

     + Expect and respond to asynchronous requests from the server, such
       as ANNOUNCE. This does not necessarily mean that it should imple-
       ment the ANNOUNCE method, merely that it MUST respond positively
       or negatively to any request received from the server.

   Though not required, the following are RECOMMENDED.

     + Implement RTP/AVP/UDP as a valid transport.

     + Inclusion of the User-Agent header.

     + Understand SDP session descriptions as defined in Appendix C

     + Accept media initialization formats (such as SDP) from standard
       input, command line, or other means appropriate to the operating
       environment to act as a "helper application" for other applica-
       tions (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 addi-
   tionally be able to do the following:

     + generate the PAUSE request;

     + 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 fol-
   lowing:

     + recognize the 401 (Unauthorized) status code;





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     + parse and include the WWW-Authenticate header;

     + implement Basic Authentication and Digest Authentication.

D.2 Server

   A minimal server implementation MUST be able to do the following:

     + Implement the following methods: SETUP, TEARDOWN, OPTIONS and
       either PLAY (for a minimal playback server) or RECORD (for a min-
       imal recording server).

       If RECORD is implemented, ANNOUNCE SHOULD be implemented as well.

     + Include the following headers in responses: Connection, Content-
       Length, Content-Type, Content-Language, Content-Encoding, Trans-
       port, Public. The capability to include the Location header
       should be implemented if the RECORD method is. RTP-compliant
       implementations should also implement the RTP-Info field.

     + Parse and respond appropriately to the following headers in
       requests: Connection, Session, Transport, Require.

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial implemen-
   tations and/or to be a "good citizen".

     + Implement RTP/AVP/UDP as a valid transport.

     + Inclusion of the Server header.

     + Implement the DESCRIBE method.

     + 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 addi-
   tionally be able to do the following:





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     + Recognize the Range header, and return an error if seeking is not
       supported.

     + 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:

     + Include and parse the Range header, with NPT units. Implementa-
       tion of SMPTE units is recommended.

     + Include the length of the media presentation in the media ini-
       tialization information.

     + 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 informa-
     tion to determine if the clip is seekable, and visably disable
     seeking features for clips for which the length information is
     unavailable. A common use of the presentation length is to
     implement a "slider bar" which serves as both a progress indi-
     cator 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:

     + Generate the 401 (Unauthorized) status code when authentication
       is required for the resource.

     + Parse and include the WWW-Authenticate header

     + Implement Basic Authentication and Digest Authentication

E Changes

   Since RFC 2326, the following issues were addressed:

     + http://rtsp.org/bug448521 - URLs in Rtp-Info need to be quoted




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     + http://rtsp.org/bug448525 - Syntax for SSRC should be clarified

     + http://rtsp.org/bug461083 - Body w/o Content-Length clarification

     + http://rtsp.org/bug477407 - Transport BNF doesn't properly deal
       with semicolon and comma

     + http://rtsp.org/bug477413 - Transport BNF: mode parameter issues

     + http://rtsp.org/bug477416 - BNF error section 3.6 NPT

     + http://rtsp.org/bug477421 - When to send response

     + http://rtsp.org/bug507347 - Removal of destination redirection

     + http://rtsp.org/bug477404 - Expanded the table to something use-
       ful, proxy indications still missing.

     + http://rtsp.org/bug477419 - Started updating to rfc2616 by adding
       public header. Section references in header chapter needs update.

     + http://rtsp.org/bug500803 - Rewritten the complete chapter on the
       state machine. Needs review.

     + http://rtsp.org/bug513753 - Created a IANA section defining four
       registries.

   Note that this list does not reflect minor changes in wording or cor-
   rection of typographical errors.

   A word-by-word diff from RFC 2326 can be found at
   http://rtsp.org/2002/drafts

F 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




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   Robert Lanphier
   RealNetworks
   P.O. Box 91123
   Seattle, WA 98111-9223
   USA
   electronic mail: robla@real.com

   Magnus Westerlund
   Ericsson AB, ERA/TVA/A
   Torshamsgatan 23
   SE-164 80 STOCKHOLM
   SWEDEN
   electronic mail: magnus.westerlund@ericsson.com

G 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 men-
   tioned, the following individuals have contributed to this specifica-
   tion:

   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, Philipp Hoschka, Anne Jones, Anders Klemets, Ruth Lang,
   Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas Marshall,
   Rob McCool, Aravind Narasimhan, David Oran, Joerg Ott, Maria
   Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,
   Igor Plotnikov, Jonathan Sergent, Pinaki Shah, David Singer, Jeff
   Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, and
   David Walker.

   [1] H. Schulzrinne, "RTP profile for audio and video conferences with
   minimal control," RFC 1890, Internet Engineering Task Force, Jan.
   1996.

   [2] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-Lee,
   "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Engi-
   neering Task Force, Jan. 1997.

   [3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst, "Internationaliza-
   tion of the hypertext markup language," RFC 2070, Internet Engineer-
   ing Task Force, Jan.  1997.



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   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [5] ISO/IEC, "Information technology -- generic coding of moving pic-
   tures and associated audio informaiton -- part 6: extension for digi-
   tal storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [6] J. Franks, P. Hallam-Baker, and J. Hostetler, "An extension to
   HTTP: digest access authentication," RFC 2069, Internet Engineering
   Task Force, Jan.  1997.

   [7] J. Postel, "User datagram protocol," RFC STD 6, 768, Internet
   Engineering Task Force, Aug. 1980.

   [8] B. Hinden and C. Partridge, "Version 2 of the reliable data pro-
   tocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. 1990.

   [9] J. Postel, "Transmission control protocol," RFC STD 7, 793,
   Internet Engineering Task Force, Sept. 1981.

   [10] 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.

   [11] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.

   [12] J. Miller, P. Resnick, and D. Singer, "Rating services and rat-
   ing systems (and their machine readable descriptions)," Recommenda-
   tion REC-PICS-services-961031, W3C (World Wide Web Consortium),
   Boston, Massachusetts, Oct. 1996.

   [13] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
   distribution label syntax and communication protocols," Recommenda-
   tion REC-PICS-labels-961031, W3C (World Wide Web Consortium), Boston,
   Massachusetts, Oct. 1996.

   [14] D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
   tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov. 1997.

   [15] B. Braden, "Requirements for internet hosts - application and
   support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
   1989.





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   [16] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
   Internet Engineering Task Force, Apr. 1996.

   [17] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
   locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
   1994.

   [18] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   2279, Internet Engineering Task Force, Jan. 1998.

   [19] B. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," RFC 1644, Internet Engineering Task Force, July 1994.

   [20] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
   Reading, Massachusetts: Addison-Wesley, 1994.

   [21] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [22] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   identifiers (URI): generic syntax," RFC 2396, Internet Engineering
   Task Force, Aug.  1998.

   [23] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," RFC 1889, Internet
   Engineering Task Force, Jan. 1996.

   [24] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.

   [25] R. Fielding, "Relative uniform resource locators," RFC 1808,
   Internet Engineering Task Force, June 1995.

   [26] R. Fielding, "Hypertext Transfer Protocol -- HTTP/1.1," RFC
   2616, Internet Engineering Task Force, June 1999.

   [27] T. Dierks, C. Allen, "The TLS Protocol, Version 1.0," RFC 2246,
   Internet Engineering Task Force, Januari 1999.

   [28] International Telecommunication Union, "Visual telephone systems
   and equipment for local area networks which provide a non-guaranteed
   quality of service," Recommendation H.323, Telecommunications Stan-
   darization Sector of ITU, Geneva, Switzerland, May 1996.

   [29] T. Narten, H. Alvestrand, "Guidelines for Writing an IANA Con-
   siderations Section in RFCs," RFC2434, Internet Engineering Task
   Force, October 1998.



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Full Copyright Statement

   Copyright (C) The Internet Society (2002). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published and
   distributed, in whole or in part, without restriction of any kind,
   provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this docu-
   ment itself may not be modified in any way, such as by removing the
   copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of develop-
   ing Internet standards in which case the procedures for copyrights
   defined in the Internet Standards process must be followed, or as
   required to translate it into languages other than English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MER-
   CHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

























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                           Table of Contents


1          Introduction  . . . . . . . . . . . . . . . . . . . . . .   3
1.1        Purpose . . . . . . . . . . . . . . . . . . . . . . . . .   3
1.2        Requirements  . . . . . . . . . . . . . . . . . . . . . .   4
1.3        Terminology . . . . . . . . . . . . . . . . . . . . . . .   4
1.4        Protocol Properties . . . . . . . . . . . . . . . . . . .   6
1.5        Extending RTSP  . . . . . . . . . . . . . . . . . . . . .   8
1.6        Overall Operation . . . . . . . . . . . . . . . . . . . .   9
1.7        RTSP States . . . . . . . . . . . . . . . . . . . . . . .  10
1.8        Relationship with Other Protocols . . . . . . . . . . . .  11
2          Notational Conventions  . . . . . . . . . . . . . . . . .  11
3          Protocol Parameters . . . . . . . . . . . . . . . . . . .  12
3.1        RTSP Version  . . . . . . . . . . . . . . . . . . . . . .  12
3.2        RTSP URL  . . . . . . . . . . . . . . . . . . . . . . . .  12
3.3        Session Identifiers . . . . . . . . . . . . . . . . . . .  13
3.4        SMPTE Relative Timestamps . . . . . . . . . . . . . . . .  14
3.5        Normal Play Time  . . . . . . . . . . . . . . . . . . . .  14
3.6        Absolute Time . . . . . . . . . . . . . . . . . . . . . .  15
3.7        Option Tags . . . . . . . . . . . . . . . . . . . . . . .  16
3.7.1      Registering New Option Tags with IANA . . . . . . . . . .  16
4          RTSP Message  . . . . . . . . . . . . . . . . . . . . . .  17
4.1        Message Types . . . . . . . . . . . . . . . . . . . . . .  17
4.2        Message Headers . . . . . . . . . . . . . . . . . . . . .  17
4.3        Message Body  . . . . . . . . . . . . . . . . . . . . . .  17
4.4        Message Length  . . . . . . . . . . . . . . . . . . . . .  17
5          General Header Fields . . . . . . . . . . . . . . . . . .  18
6          Request . . . . . . . . . . . . . . . . . . . . . . . . .  18
6.1        Request Line  . . . . . . . . . . . . . . . . . . . . . .  19
6.2        Request Header Fields . . . . . . . . . . . . . . . . . .  19
7          Response  . . . . . . . . . . . . . . . . . . . . . . . .  20
7.1        Status-Line . . . . . . . . . . . . . . . . . . . . . . .  20
7.1.1      Status Code and Reason Phrase . . . . . . . . . . . . . .  21
7.1.2      Response Header Fields  . . . . . . . . . . . . . . . . .  23
8          Entity  . . . . . . . . . . . . . . . . . . . . . . . . .  25
8.1        Entity Header Fields  . . . . . . . . . . . . . . . . . .  25
8.2        Entity Body . . . . . . . . . . . . . . . . . . . . . . .  25
9          Connections . . . . . . . . . . . . . . . . . . . . . . .  26
9.1        Pipelining  . . . . . . . . . . . . . . . . . . . . . . .  26
9.2        Reliability and Acknowledgements  . . . . . . . . . . . .  26
10         Method Definitions  . . . . . . . . . . . . . . . . . . .  27
10.1       OPTIONS . . . . . . . . . . . . . . . . . . . . . . . . .  27
10.2       DESCRIBE  . . . . . . . . . . . . . . . . . . . . . . . .  28
10.3       ANNOUNCE  . . . . . . . . . . . . . . . . . . . . . . . .  30
10.4       SETUP . . . . . . . . . . . . . . . . . . . . . . . . . .  31
10.5       PLAY  . . . . . . . . . . . . . . . . . . . . . . . . . .  32
10.6       PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . .  34



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10.7       TEARDOWN  . . . . . . . . . . . . . . . . . . . . . . . .  36
10.8       GET_PARAMETER . . . . . . . . . . . . . . . . . . . . . .  36
10.9       SET_PARAMETER . . . . . . . . . . . . . . . . . . . . . .  37
10.10      REDIRECT  . . . . . . . . . . . . . . . . . . . . . . . .  38
10.11      RECORD  . . . . . . . . . . . . . . . . . . . . . . . . .  39
10.12      PING  . . . . . . . . . . . . . . . . . . . . . . . . . .  39
10.13      Embedded (Interleaved) Binary Data  . . . . . . . . . . .  40
11         Status Code Definitions . . . . . . . . . . . . . . . . .  41
11.1       Success 2xx . . . . . . . . . . . . . . . . . . . . . . .  41
11.1.1     250 Low on Storage Space  . . . . . . . . . . . . . . . .  41
11.2       Redirection 3xx . . . . . . . . . . . . . . . . . . . . .  41
11.3       Client Error 4xx  . . . . . . . . . . . . . . . . . . . .  41
11.4       400 Bad Request . . . . . . . . . . . . . . . . . . . . .  42
11.4.1     405 Method Not Allowed  . . . . . . . . . . . . . . . . .  42
11.4.2     451 Parameter Not Understood  . . . . . . . . . . . . . .  42
11.4.3     452 reserved  . . . . . . . . . . . . . . . . . . . . . .  42
11.4.4     453 Not Enough Bandwidth  . . . . . . . . . . . . . . . .  42
11.4.5     454 Session Not Found . . . . . . . . . . . . . . . . . .  42
11.4.6     455 Method Not Valid in This State  . . . . . . . . . . .  42
11.4.7     456 Header Field Not Valid for Resource . . . . . . . . .  42
11.4.8     457 Invalid Range . . . . . . . . . . . . . . . . . . . .  43
11.4.9     458 Parameter Is Read-Only  . . . . . . . . . . . . . . .  43
11.4.10    459 Aggregate Operation Not Allowed . . . . . . . . . . .  43
11.4.11    460 Only Aggregate Operation Allowed  . . . . . . . . . .  43
11.4.12    461 Unsupported Transport . . . . . . . . . . . . . . . .  43
11.4.13    462 Destination Unreachable . . . . . . . . . . . . . . .  43
11.5       Server Error 5xx  . . . . . . . . . . . . . . . . . . . .  43
11.5.1     551 Option not supported  . . . . . . . . . . . . . . . .  43
12         Header Field Definitions  . . . . . . . . . . . . . . . .  43
12.1       Accept  . . . . . . . . . . . . . . . . . . . . . . . . .  46
12.2       Accept-Encoding . . . . . . . . . . . . . . . . . . . . .  46
12.3       Accept-Language . . . . . . . . . . . . . . . . . . . . .  46
12.4       Accept-Ranges . . . . . . . . . . . . . . . . . . . . . .  46
12.5       Allow . . . . . . . . . . . . . . . . . . . . . . . . . .  48
12.6       Authorization . . . . . . . . . . . . . . . . . . . . . .  48
12.7       Bandwidth . . . . . . . . . . . . . . . . . . . . . . . .  49
12.8       Blocksize . . . . . . . . . . . . . . . . . . . . . . . .  49
12.9       Cache-Control . . . . . . . . . . . . . . . . . . . . . .  49
12.10      Connection  . . . . . . . . . . . . . . . . . . . . . . .  51
12.11      Content-Base  . . . . . . . . . . . . . . . . . . . . . .  51
12.12      Content-Encoding  . . . . . . . . . . . . . . . . . . . .  52
12.13      Content-Language  . . . . . . . . . . . . . . . . . . . .  52
12.14      Content-Length  . . . . . . . . . . . . . . . . . . . . .  52
12.15      Content-Location  . . . . . . . . . . . . . . . . . . . .  52
12.16      Content-Type  . . . . . . . . . . . . . . . . . . . . . .  52
12.17      CSeq  . . . . . . . . . . . . . . . . . . . . . . . . . .  52
12.18      Date  . . . . . . . . . . . . . . . . . . . . . . . . . .  53
12.19      Expires . . . . . . . . . . . . . . . . . . . . . . . . .  53



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12.20      From  . . . . . . . . . . . . . . . . . . . . . . . . . .  54
12.21      Host  . . . . . . . . . . . . . . . . . . . . . . . . . .  54
12.22      If-Match  . . . . . . . . . . . . . . . . . . . . . . . .  54
12.23      If-Modified-Since . . . . . . . . . . . . . . . . . . . .  54
12.24      Last-Modified . . . . . . . . . . . . . . . . . . . . . .  55
12.25      Location  . . . . . . . . . . . . . . . . . . . . . . . .  55
12.26      Proxy-Authenticate  . . . . . . . . . . . . . . . . . . .  55
12.27      Proxy-Require . . . . . . . . . . . . . . . . . . . . . .  55
12.28      Public  . . . . . . . . . . . . . . . . . . . . . . . . .  55
12.29      Range . . . . . . . . . . . . . . . . . . . . . . . . . .  56
12.30      Referer . . . . . . . . . . . . . . . . . . . . . . . . .  57
12.31      Retry-After . . . . . . . . . . . . . . . . . . . . . . .  57
12.32      Require . . . . . . . . . . . . . . . . . . . . . . . . .  57
12.33      RTP-Info  . . . . . . . . . . . . . . . . . . . . . . . .  58
12.34      Scale . . . . . . . . . . . . . . . . . . . . . . . . . .  59
12.35      Speed . . . . . . . . . . . . . . . . . . . . . . . . . .  60
12.36      Server  . . . . . . . . . . . . . . . . . . . . . . . . .  61
12.37      Session . . . . . . . . . . . . . . . . . . . . . . . . .  61
12.38      Supported . . . . . . . . . . . . . . . . . . . . . . . .  61
12.39      Timestamp . . . . . . . . . . . . . . . . . . . . . . . .  62
12.40      Transport . . . . . . . . . . . . . . . . . . . . . . . .  62
12.41      Unsupported . . . . . . . . . . . . . . . . . . . . . . .  66
12.42      User-Agent  . . . . . . . . . . . . . . . . . . . . . . .  66
12.43      Vary  . . . . . . . . . . . . . . . . . . . . . . . . . .  66
12.44      Via . . . . . . . . . . . . . . . . . . . . . . . . . . .  66
12.45      WWW-Authenticate  . . . . . . . . . . . . . . . . . . . .  66
13         Caching . . . . . . . . . . . . . . . . . . . . . . . . .  66
14         Examples  . . . . . . . . . . . . . . . . . . . . . . . .  67
14.1       Media on Demand (Unicast) . . . . . . . . . . . . . . . .  67
14.2       Streaming of a Container file . . . . . . . . . . . . . .  69
14.3       Single Stream Container Files . . . . . . . . . . . . . .  72
14.4       Live Media Presentation Using Multicast . . . . . . . . .  74
14.5       Recording . . . . . . . . . . . . . . . . . . . . . . . .  75
15         Syntax  . . . . . . . . . . . . . . . . . . . . . . . . .  76
15.1       Base Syntax . . . . . . . . . . . . . . . . . . . . . . .  76
16         Security Considerations . . . . . . . . . . . . . . . . .  77
17         IANA Considerations . . . . . . . . . . . . . . . . . . .  79
17.1       Option-tags . . . . . . . . . . . . . . . . . . . . . . .  80
17.1.1     Description . . . . . . . . . . . . . . . . . . . . . . .  80
17.1.2     Registering New Option Tags with IANA . . . . . . . . . .  80
17.1.3     Registered entries  . . . . . . . . . . . . . . . . . . .  81
17.2       RTSP Methods  . . . . . . . . . . . . . . . . . . . . . .  81
17.2.1     Description . . . . . . . . . . . . . . . . . . . . . . .  81
17.2.2     Registering New Methods with IANA . . . . . . . . . . . .  81
17.2.3     Registered entries  . . . . . . . . . . . . . . . . . . .  81
17.3       RTSP Headers  . . . . . . . . . . . . . . . . . . . . . .  82
17.3.1     Description . . . . . . . . . . . . . . . . . . . . . . .  82
17.3.2     Registering New Headers with IANA . . . . . . . . . . . .  82



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17.3.3     Registered entries  . . . . . . . . . . . . . . . . . . .  82
17.4       Parameters  . . . . . . . . . . . . . . . . . . . . . . .  82
17.4.1     Description . . . . . . . . . . . . . . . . . . . . . . .  82
17.4.2     Registering New Parameters with IANA  . . . . . . . . . .  83
17.4.3     Registered entries  . . . . . . . . . . . . . . . . . . .  83
A          RTSP Protocol State Machine . . . . . . . . . . . . . . .  83
A.1        States  . . . . . . . . . . . . . . . . . . . . . . . . .  83
A.2        State variables . . . . . . . . . . . . . . . . . . . . .  84
A.3        Abbreviations . . . . . . . . . . . . . . . . . . . . . .  84
A.4        State Tables  . . . . . . . . . . . . . . . . . . . . . .  84
B          Interaction with RTP  . . . . . . . . . . . . . . . . . .  89
C          Use of SDP for RTSP Session Descriptions  . . . . . . . .  90
C.1        Definitions . . . . . . . . . . . . . . . . . . . . . . .  90
C.1.1      Control URL . . . . . . . . . . . . . . . . . . . . . . .  90
C.1.2      Media Streams . . . . . . . . . . . . . . . . . . . . . .  91
C.1.3      Payload Type(s) . . . . . . . . . . . . . . . . . . . . .  91
C.1.4      Format-Specific Parameters  . . . . . . . . . . . . . . .  91
C.1.5      Range of Presentation . . . . . . . . . . . . . . . . . .  91
C.1.6      Time of Availability  . . . . . . . . . . . . . . . . . .  92
C.1.7      Connection Information  . . . . . . . . . . . . . . . . .  92
C.1.8      Entity Tag  . . . . . . . . . . . . . . . . . . . . . . .  92
C.2        Aggregate Control Not Available . . . . . . . . . . . . .  93
C.3        Aggregate Control Available . . . . . . . . . . . . . . .  93
D          Minimal RTSP implementation . . . . . . . . . . . . . . .  94
D.1        Client  . . . . . . . . . . . . . . . . . . . . . . . . .  94
D.1.1      Basic Playback  . . . . . . . . . . . . . . . . . . . . .  95
D.1.2      Authentication-enabled  . . . . . . . . . . . . . . . . .  95
D.2        Server  . . . . . . . . . . . . . . . . . . . . . . . . .  96
D.2.1      Basic Playback  . . . . . . . . . . . . . . . . . . . . .  96
D.2.2      Authentication-enabled  . . . . . . . . . . . . . . . . .  97
E          Changes . . . . . . . . . . . . . . . . . . . . . . . . .  97
F          Author Addresses  . . . . . . . . . . . . . . . . . . . .  98
G          Acknowledgements  . . . . . . . . . . . . . . . . . . . .  99


















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