Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                                   T. Zeng
Expires: November 8, 2012 May 7, 27, 2013                                  November 23, 2012

 The Evaluation of Different Network Addres Address Translator (NAT) Traversal
 Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
                draft-ietf-mmusic-rtsp-nat-evaluation-05
                draft-ietf-mmusic-rtsp-nat-evaluation-06

Abstract

   This document describes several Network Address Translator (NAT)
   traversal techniques that was were considered to be used for establishing
   the RTP media flows controlled by the Real-time Streaming Protocol
   (RTSP).  Each technique includes a description on how it would be
   used, the security implications of using it and any other deployment
   considerations it has.  There are also disussions discussions on how NAT
   traversal techniques relates to firewalls and how each technique can
   be applied in different use cases.  These findings where used when
   selecting the NAT traversal for RTSP 2.0 standardized
   in the MMUSIC WG. 2.0.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  5
     1.1.  Network Address Translators  . . . . . . . . . . . . . . .  6
     1.2.  Firewalls  . . . . . . . . . . . . . . . . . . . . . . . .  7
     1.3.  Glossary . . . . . . . . . . . . . . . . . . . . . . . . .  7
     1.4.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  8
   2.  Detecting the loss of NAT mappings . . . . . . . . . . . . . .  8
   3.  Requirements on NAT-Traversal  . . . . . . . . . . . . . . . .  9
   4.  NAT Traversal Techniques . . . . . . . . . . . . . . . . . . . 10
     4.1.  Stand-Alone STUN . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
       4.1.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 11
       4.1.2.  Using STUN to traverse NAT without server
               modifications  . . . . . . . . . . . . . . . . . . . . 11
       4.1.3.  Embedding STUN in RTSP  ALG considerations . . . . . . . . . . . . . . . . 13
       4.1.4.  Discussion On Co-located STUN Server . . 14
       4.1.4.  Deployment Considerations  . . . . . . . 14
       4.1.5.  ALG considerations . . . . . . . 14
       4.1.5.  Security Considerations  . . . . . . . . . . . 15
       4.1.6.  Deployment Considerations . . . . 15
     4.2.  Server Embedded STUN . . . . . . . . . . 15
       4.1.7.  Security Considerations . . . . . . . . . 16
       4.2.1.  Introduction . . . . . . 17
     4.2.  ICE . . . . . . . . . . . . . . . 16
       4.2.2.  Embedding STUN in RTSP . . . . . . . . . . . . 18
       4.2.1.  Introduction . . . . 16
       4.2.3.  Discussion On Co-located STUN Server . . . . . . . . . 17
       4.2.4.  ALG considerations . . . . . . . . 18
       4.2.2.  Using ICE in RTSP . . . . . . . . . . 17
       4.2.5.  Deployment Considerations  . . . . . . . . 19
       4.2.3.  Implementation burden of ICE . . . . . . 17
       4.2.6.  Security Considerations  . . . . . . . 20
       4.2.4.  Deployment Considerations . . . . . . . . 19
     4.3.  ICE  . . . . . . 21
       4.2.5.  Security Consideration . . . . . . . . . . . . . . . . 21
     4.3.  Symmetric RTP . . . . . 19
       4.3.1.  Introduction . . . . . . . . . . . . . . . . . 22
       4.3.1.  Introduction . . . . 19
       4.3.2.  Using ICE in RTSP  . . . . . . . . . . . . . . . . . 22
       4.3.2.  Necessary RTSP extensions . 20
       4.3.3.  Implementation burden of ICE . . . . . . . . . . . . . 22
       4.3.3. 21
       4.3.4.  Deployment Considerations  . . . . . . . . . . . . . . 23
       4.3.4. 22
       4.3.5.  Security Consideration . . . . . . . . . . . . . . . . 23
       4.3.5.  A Variation to Symmetric RTP . . . . 22
     4.4.  Latching . . . . . . . . . 24
     4.4.  Application Level Gateways . . . . . . . . . . . . . . . . 26 22
       4.4.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 26 22
       4.4.2.  Outline On how ALGs for  Necessary RTSP work extensions  . . . . . . . . . . 27 . . . . 23
       4.4.3.  Deployment Considerations  . . . . . . . . . . . . . . 28 23
       4.4.4.  Security Considerations  . . . . Consideration . . . . . . . . . . . 28
     4.5.  TCP Tunneling . . . . . 24
     4.5.  A Variation to Latching  . . . . . . . . . . . . . . . . . 28 25
       4.5.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 28 25
       4.5.2.  Usage of TCP tunneling in  Necessary RTSP extensions  . . . . . . . . . . . . 29 . . 26
       4.5.3.  Deployment Considerations  . . . . . . . . . . . . . . 29 27
       4.5.4.  Security Considerations  . . . . . . . . . . . . . . . 29 27
     4.6.  TURN (Traversal Using Relay NAT)  Three Way Latching . . . . . . . . . . . . . 30
       4.6.1.  Introduction . . . . . . . 27
       4.6.1.  Introduction . . . . . . . . . . . . . . 30 . . . . . . . 28
       4.6.2.  Usage of TURN with  Necessary RTSP extensions  . . . . . . . . . . . . . . . 30 28
       4.6.3.  Deployment Considerations  . . . . . . . . . . . . . . 31
       4.6.4. 28
     4.7.  Application Level Gateways . . . . . . . . . . . . . . . . 28
       4.7.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 29
       4.7.2.  Outline On how ALGs for RTSP work  . . . . . . . . . . 29
       4.7.3.  Deployment Considerations  . . . . . . . . . . . . . . 30
       4.7.4.  Security Considerations  . . . . . . . . . . . . . . . 32
   5.  Firewalls 31

     4.8.  TCP Tunneling  . . . . . . . . . . . . . . . . . . . . . . 31
       4.8.1.  Introduction . . . . . 33
   6.  Comparision . . . . . . . . . . . . . . . . 31
       4.8.2.  Usage of NAT traversal techniques TCP tunneling in RTSP . . . . . . . . . . . 33
   7.  IANA . 31
       4.8.3.  Deployment Considerations  . . . . . . . . . . . . . . 32
       4.8.4.  Security Considerations  . . . . . . . . . . . . . . . 32
     4.9.  TURN (Traversal Using Relay NAT) . . . . . . . . . . . . . 32
       4.9.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 32
       4.9.2.  Usage of TURN with RTSP  . . . . . . . . . . . . . . . 33
       4.9.3.  Deployment Considerations  . . . . . . . . . . . . . . 34
   8.
       4.9.4.  Security Considerations  . . . . . . . . . . . . . . . 35
   5.  Firewalls  . . . . 34
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35
   10. Informative References
   6.  Comparison of NAT traversal techniques . . . . . . . . . . . . 36
   7.  IANA Considerations  . . . . . . . . 35
   Authors' Addresses . . . . . . . . . . . . . 38
   8.  Security Considerations  . . . . . . . . . . . 37

1. . . . . . . . . 38
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 39
   10. Informative References . . . . . . . . . . . . . . . . . . . . 39
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41

1.  Introduction

   Today there is a proliferate deployment of different flavors of
   Network Address Translator (NAT) boxes that in many cases only
   loosely follows follow standards [RFC3022][RFC2663][RFC3424]].
   [RFC3022][RFC2663][RFC3424][RFC4787][RFC5382].  NATs cause
   discontinuity in address realms [RFC3424], therefore an application
   protocol, such as Real-time Streaming Protocol (RTSP)
   [RFC2326][I-D.ietf-mmusic-rfc2326bis], needs to deal with such
   discontinuities caused by NATs.  The problem is that, being a media
   control protocol managing one or more media streams, RTSP carries
   network address and port information within its protocol messages.
   Because of this, even if RTSP itself, when carried over Transmission
   Control Protocol (TCP) [RFC0793] for example, may is not be blocked by NATs,
   its media streams may be blocked by NATs.  This will occur unless
   special protocol provisions are added to support NAT-
   traversal. NAT-traversal.

   Like NATs, firewalls (FWs) Firewalls are also middle boxes that need to be
   considered.  Firewalls helps prevent unwanted traffic from getting in
   or out of the protected network.  RTSP is designed such that a
   firewall can be configured to let RTSP controlled media streams to go
   through with minimal implementation effort.  The minimal effort is to
   implement an Application Level Gateway (ALG) to interpret RTSP
   parameters.  There is also a large class of firewalls, commonly home
   firewalls, that uses a similar filtering behavior to what NAT has.
   This type of firewalls can be handled using the same solution as
   employed for NAT traversal instead of relying on ALGs.

   This document describes several NAT-traversal mechanisms for RTSP
   controlled media streaming.  These  Many of these NAT solutions fall into
   the category of "UNilateral Self-Address Fixing (UNSAF)" as defined
   in [RFC3424] and quoted below:

   "UNSAF is a process whereby some originating process attempts to
   determine or fix the address (and port) by which it is known - e.g.
   to be able to use address data in the protocol exchange, or to
   advertise a public address from which it will receive connections."

   Following the guidelines spelled out in RFC 3424, we describe the
   required RTSP protocol extensions for each method, transition
   strategies, and security concerns.

   This document is capturing the evaluation done in the process to
   recommend FW/NAT Firewall/NAT traversal methods for RTSP streaming servers
   based on RFC 2326 [RFC2326] as well as the RTSP 2.0 core spec
   [I-D.ietf-mmusic-rfc2326bis].  The evaluation is focused on NAT
   traversal for the media streams carried over User Datagram Protocol
   (UDP) [RFC0768].  Where [RFC0768] with Real-time Transport Protocol (RTP) [RFC3550]
   over UDP being the main case for such usage.  The findings should be
   applicable to other protocols as long as they have similar
   properties.

   The resulting ICE-based RTSP NAT traversal mechanism is specified in
   "A Network Address Translator (NAT) Traversal mechanism for media
   controlled by Real-Time Streaming Protocol (RTSP)"
   [I-D.ietf-mmusic-rtsp-nat].

1.1.  Network Address Translators

   Readers are urged to refer to "IP Network Address Translator (NAT)
   Terminology and Considerations" [RFC2663] for information on NAT
   taxonomy and terminology.  Traditional NAT is the most common type of
   NAT device deployed.  Readers may refer to "Traditional IP Network
   Address Translator (Traditional NAT)" [RFC3022] for detailed
   information on traditional NAT.  Traditional NAT has two main
   varieties -- Basic NAT and Network Address/Port Translator (NAPT).

   NAPT is by far the most commonly deployed NAT device.  NAPT allows
   multiple internal hosts to share a single public IP address
   simultaneously.  When an internal host opens an outgoing TCP or UDP
   session through a NAPT, the NAPT assigns the session a public IP
   address and port number, so that subsequent response packets from the
   external endpoint can be received by the NAPT, translated, and
   forwarded to the internal host.  The effect is that the NAPT
   establishes a NAT mapping to translate the (private IP address,
   private port number) tuple to a (public IP address, public port
   number) tuple, and vice versa, for the duration of the session.  An
   issue of relevance to peer-to-peer applications is how the NAT
   behaves when an internal host initiates multiple simultaneous
   sessions from a single (private IP, private port) endpoint to
   multiple distinct endpoints on the external network.  In this
   specification, the term "NAT" refers to both "Basic NAT" and "Network
   Address/Port Translator (NAPT)".

   This document uses the term "address and port mapping" as the
   translation between an external address and port and an internal
   address and port.  Note that this is not the same as an "address
   binding" as defined in RFC 2663.  There exist a number of address and
   port mapping behaviors described in more detail in Section 4.1 of
   "Network Address Translation (NAT) Behavioral Requirements for
   Unicast UDP" [RFC4787].

   NATs also have a filtering behavior on traffic arriving on the
   external side.  Such behavior effects affects how well different methods for
   NAT traversal works through these NATs.  See Section 5 of "Network
   Address Translation (NAT) Behavioral Requirements for Unicast UDP"

   [RFC4787] for more information on the different types of filtering
   that have been identified.

1.2.  Firewalls

   A firewall (FW) is a security gateway that enforces certain access control
   policies between two network administrative domains: a private domain
   (intranet) and a external domain, e.g. public Internet.  Many
   organizations use firewalls to prevent privacy intrusions and
   malicious attacks to corporate computing resources in the private
   intranet [RFC2588].

   A comparison between NAT and FW Firewall is given below:

   1.  A firewall must sit between two network administrative domains,
       while NAT does not have to sit between two domains.

   2.  NAT does not in itself provide security, although some access
       control policies can be implemented using address translation
       schemes.  The inherent filtering behaviours are commonly mistaken
       for real security policies.

   It should be noted that many NAT devices intended for Residential or
   small office/
   home office/home office (SOHO) use include both NATs and firewall
   functionality.

   In the rest of this memo we use the phrase "NAT traversal"
   interchangeably with "FW "Firewall traversal", "NAT/FW traversal" and "NAT/
   Firewall "NAT/Firewall
   traversal".

1.3.  Glossary

   Address-Dependent Mapping:  The NAT reuses the port mapping for
         subsequent packets sent from the same internal IP address and
         port to the same external IP address, regardless of the
         external port.  See [RFC4787].

   Address and Port-Dependent Mapping:  The NAT reuses the port mapping
         for subsequent packets sent from the same internal IP address
         and port to the same external IP address and port while the
         mapping is still active.  See [RFC4787].

   ALG:  Application Level Gateway, an entity that can be embedded in a
         NAT or other middlebox to perform the application layer
         functions required for a particular protocol to traverse the
         NAT/middlebox.

   Endpoint-Independent Mapping:  The NAT reuses the port mapping for
         subsequent packets sent from the same internal IP address and
         port to any external IP address and port.  See [RFC4787].

   ICE:  Interactive Connectivity Establishment, see [RFC5245].

   DNS:  Domain Name Service

   DDOS:

   DoS:  Denial of Service

   DDoS: Distributed Denial Of of Service attacks

   NAT:  Network Address Translator, see [RFC3022].

   NAPT: Network Address/Port Translator, see [RFC3022].

   RTP:  Real-time Transport Protocol, see [RFC3550].

   RTSP: Real-Time Streaming Protocol, see [RFC2326] and
         [I-D.ietf-mmusic-rfc2326bis].

   RTT:  Round Trip Times.

   SDP:  Session Description Protocol, see [RFC4566].

   SSRC: Synchronization source in RTP, see [RFC3550].

1.4.  Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in RFC
   2119 [RFC2119].

2.  Detecting the loss of NAT mappings

   Several NAT traversal techniques in the next chapter make use of the
   fact that the NAT UDP mapping's external address and port can be
   discovered.  This information is then utilized to traverse the NAT
   box.  However any such information is only good while the mapping is
   still valid.  As the IAB's UNSAF document [RFC3424] points out, the
   mapping can either timeout or change its properties.  It is therefore
   important for the NAT traversal solutions to handle the loss or
   change of NAT mappings, according to RFC3424.

   First, since NATs may also dynamically reclaim or readjust address/
   port translations, "keep-alive" and periodic re-polling may be
   required according to RFC 3424.  Secondly, it is possible to detect
   and recover from the situation where the mapping has been changed or
   removed.  The loss of a mapping can be detected when no traffic
   arrives for a while.  Below we will give some recommendation on how
   to detect loss of NAT mappings when using RTP/RTCP under RTSP
   control.

   A RTP session normally has both RTP and RTCP streams.  The loss of a
   RTP mapping can only be detected when expected traffic does not
   arrive.  If a client does not receive data within a few seconds after
   having received the "200 OK" response to a PLAY request, there are
   likely some middleboxes blocking the traffic.  However, for a
   receiver to be more certain to detect the case where no RTP traffic
   was delivered due to NAT trouble, one should monitor the RTCP Sender
   reports.  The sender report carries a field telling how many packets
   the server has sent.  If that has increased and no RTP packets has
   arrived for a few seconds it is likely the RTP mapping has been
   removed.

   The loss of mapping for RTCP is simpler to detect.  RTCP is normally
   sent periodically in each direction, even during the RTSP ready
   state.  If RTCP packets are missing for several RTCP intervals, the
   mapping is likely to be lost.  Note that if neither RTCP packets nor RTSP
   messages are received by the RTSP server for a while, the RTSP server
   has the option to delete the corresponding RTP session, SSRC and RTSP
   session ID, because either the client can not get through a middle
   box NAT/FW, NAT/Firewall, or that the client is mal-functioning.

3.  Requirements on NAT-Traversal

   This section considers the set of requirements for the evaulation evaluation of
   RTSP NAT traversal solutions.

   RTSP is a client-server protocol.  Typically services service providers deploy
   RTSP servers in the public address realm.  However, there are use
   cases where the reverse is true: RTSP clients are connecting from
   public address realm to RTSP servers behind home NATs.  This is the
   case for instance when home surveillance cameras running as RTSP
   servers intend to stream video to cell phone users in the public
   address realm through a home NAT.  In terms of requirements, the
   first requirement should be to solve the RTSP NAT traversal problem
   for RTSP servers deployed in a public network, i.e. no NAT at the
   server side.

   The list of feature requirements for RTSP NAT solutions are given
   below:

   1.  MUST  Must work for all flavors of NATs, including NATs with address
       and port restricted filtering.

   2.  MUST  Must work for firewalls (subject to pertinent firewall
       administrative policies), including those with ALGs.

   3.  SHOULD  Should have minimal impact on clients in the open and not dual-
       hosted.  RTSP dual-hosting means that the RTSP signalling
       protocol and the media protocol (e.g.  RTP) are implemented on
       different computers with different IP addresses.

       *  For instance, no extra delay from RTSP connection till arrival
          of media

   4.  SHOULD  Should be simple to use/implement/administer that so people actually
       turn them on

       *  Otherwise people will resort to TCP tunneling through NATs

       *  Address discovery for NAT traversal should take place behind
          the scene, automatically,
          if possible

   5.  SHOULD  Should authenticate dual-hosted client transport handler to
       prevent DDOS DDoS attacks.

   The last requirement addresses the Distributed Denial-Of-Service
   (DDOS) Denial-of-Service
   (DDoS) threat, which relates to NAT traversal as explained below.

   During NAT traversal, when the RTSP server determines the media
   destination (Address (address and port) for the client, the result may be that
   the public IP address of the RTP receiver host is different than the
   public IP address of the RTSP client host.  This posts a DDOS DDoS threat
   that has significant amplification potentials because the RTP media
   streams in general consist of large number of IP packets.  DDOS  DDoS
   attacks occur if the attacker fakes the messages in the NAT traversal
   mechanism to trick the RTSP server into believing that the client's
   RTP receiver is located in on a separate host.  For example, user A may
   use his RTSP client to direct the RTSP server to send video RTP
   streams to target.example.com in order to degrade the services
   provided by target.example.com.  Note a simple preventative measure
   commonly deployed is for the RTSP server to disallow the cases where
   the client's RTP receiver has a different public IP address than that
   of the RTSP client.  However, in some applications (e.g., centralized
   conferencing), dual-hosted RTSP/RTP clients have valid use cases.
   The key is how to authenticate the messages exchanged during  With the increased deployment of NAT
   traversal process.  Message authentication is middleboxes
   by operators, i.e. carrier grade NAT (CGN), the reusing of a big challenge in public
   IP address for many customers reduces the
   current wired and wireless networking environment.  It may be
   necessary protection provided.  Also
   in the immediate future to deploy NAT traversal solutions
   that do not some applications (e.g., centralized conferencing), dual-hosted
   RTSP/RTP clients have full message authentication, but provide upgrade
   path valid use cases.  The key is how to add authentication features in
   authenticate the messages exchanged during the future. NAT traversal process.

4.  NAT Traversal Techniques

   There exist exists a number of potential NAT traversal techniques that can
   be used to allow RTSP to traverse NATs.  They have different features
   and are applicable to different topologies; their cost is costs are also
   different.  They also vary in security levels.  In the following
   sections, each technique is outlined in details with discussions on the
   corresponding advantages and disadvantages.

   The main evaluation was done prior to 2007 and are based on what was
   available then.  This section includes NAT traversal techniques that
   have not been formally specified anywhere else.  The overview section
   of this document may be the only publicly available specification of
   some of the NAT traversal techniques.  However that is no not a real
   barrier against doing an evaluation of the NAT traversal technique.
   Some other techniques have been recommended against or are currently (at the time of writing) in a state of
   flux no longer
   possible due to ongoing standardization work on these techniques works' outcome or has
   been not been progressed their failure to
   progress within IETF after the intiial initial evaluation in this document,
   e.g.  RTP No-Op [I-D.ietf-avt-rtp-no-op].

4.1.  Stand-Alone STUN

4.1.1.  Introduction

   STUN - "Simple

   Session Traversal of UDP Through Network Address Translators"
   [RFC3489][RFC5389] Utilities for NAT (STUN) [RFC5389] is a
   standardized protocol that allows a client to use secure means to
   discover the presence of a NAT between himself itself and the STUN server.
   The client uses the STUN server to discover the address mappings
   assigned by the NAT.  STUN is a client-server protocol.  The STUN
   client sends a request to a STUN server and the server returns a
   response.  There are two types of STUN requests messages - Binding Requests, sent over UDP, Requests
   and Shared Secret Requests, sent
   over TLS over TCP. Indications.  Binding requests are used when determining a
   client's external address and solicits a response from the STUN
   server with the seen address.

   The first version of STUN [RFC3489] included categorization and
   parameterization of NATs.  This was abandoned in the updated version
   [RFC5389] due to it being unreliable. unreliable and brittle.  Some of the below
   discussed methods are based on RFC3489 functionality which will be
   called out and the downside of that will be part of the
   characterization.

4.1.2.  Using STUN to traverse NAT without server modifications

   This section describes how a client can use STUN to traverse NATs to
   RTSP servers without requiring server modifications.  Note that this
   method has limited applicability and requires the server to be
   available in the external/public address realm in regards to the
   client located behind a NAT(s).

   Limitations:

   o  The server must be located in either a public address realm or the
      next hop external address realm in regards to the client.

   o  The client may only be located behind NATs that performing
      Endpoint Independent perform "Endpoint-
      Independent" or Address Dependent "Address-Dependent" Mappings.  Clients behind NATs
      that do Address "Address and Port Dependent Port-Dependent" Mappings cannot use this
      method.  See [RFC4787] for full definition of these terms.

   o  Based on the discontinued middlebox classification of the replaced
      STUN specification [RFC3489].  Thus brittle and unreliable.

   Method:

   A RTSP client using RTP transport over UDP can use STUN to traverse a
   NAT(s) in the following way:

   1.  Use STUN to try to discover the type of NAT, and the timeout
       period for any UDP mapping on the NAT.  This is RECOMMENDED recommend to be
       performed in the background as soon as IP connectivity is
       established.  If this is performed prior to establishing a
       streaming session the delays in the session establishment will be
       reduced.  If no NAT is detected, normal SETUP SHOULD should be used.

   2.  The RTSP client determines the number of UDP ports needed by
       counting the number of needed media transport protocols sessions
       in the multi-media presentation.  This information is available
       in the media description protocol, e.g.  SDP [RFC4566].  For
       example, each RTP session will in general require two UDP ports,
       one for RTP, and one for RTCP.

   3.  For each UDP port required, establish a mapping and discover the
       public/external IP address and port number with the help of the
       STUN server.  A successful mapping looks like: client's local
       address/port <-> public address/port.

   4.  Perform the RTSP SETUP for each media.  In the transport header
       the following parameter SHOULD should be included with the given values:
       "dest_addr" [I-D.ietf-mmusic-rfc2326bis] or "destination" +
       "client_port" [RFC2326] with the public/external IP address and
       port pair for both RTP and RTCP.  To be certain that this works
       servers must allow a client to setup the RTP stream on any port,
       not only even ports and with non-continuous non-contiguous port numbers for RTP
       and RTCP.  This requires the new feature provided in the update
       to RFC2326 [I-D.ietf-mmusic-rfc2326bis].  The server should
       respond with a transport header containing an "src_addr" or
       "source parameter"
       "source" + "server_port" parameters with the RTP and RTCP source
       IP address and port of the media stream.

   5.  To keep the mappings alive, the client SHOULD should periodically send
       UDP traffic over all mappings needed for the session.  For the
       mapping carrying RTCP traffic the periodic RTCP traffic may be are
       likely enough.  For mappings carrying RTP traffic and for
       mappings carrying RTCP packets at too low a frequency, keep-alive
       messages
       SHOULD should be sent.  As keep alive messages, one could use
       the RTP No-Op packet [I-D.ietf-avt-rtp-no-op] to the streaming
       server's discard port (port number 9).  The drawback of using RTP
       No-Op is that the payload type number must be dynamically
       assigned through RTSP first.  Otherwise STUN could be used for
       the keep-alive as well as empty UDP packets.

   If a UDP mapping is lost, the above discovery process must be
   repeated.  The media stream also needs to be SETUP again to change
   the transport parameters to the new ones.  This will cause a glitch
   in media playback.

   To allow UDP packets to arrive from the server to a client behind a
   "Address Dependent" filtering NAT, the client must first send a UDP
   packet to establish filtering state in the NAT.  The client, before
   sending a RTSP PLAY request, must send a so called FW hole-punching
   packet (such as a RTP No-Op packet) on each mapping, to the IP
   address given as the servers source address.  To create minimum
   problems for the server these UDP packets SHOULD should be sent to the
   server's discard port (port number 9).  Since UDP packets are
   inherently unreliable, to ensure that at least one UDP message passes
   the NAT, FW hole-punching packets should be retransmitted a reasonable
   number of times.

   For a an "Address and Port Dependent" filtering NAT the client must
   send messages to the exact ports used by the server to send UDP
   packets before sending a RTSP PLAY request.  This makes it possible
   to use the above described process with the following additional
   restrictions: for each port mapping, FW hole-punching packets need to be
   sent first to the server's source address/port.  To minimize
   potential effects on the server from these messages the following
   type of FW hole punching packets
   MUST must be sent.  RTP: an empty or less
   than 12 bytes UDP packet.  RTCP: A correctly formatted RTCP RR or SR
   message.  The above described adaptations for restricted NATs will
   not work unless the server includes the "src_addr" in the "Transport"
   header (which is the "source" transport parameter in RFC2326).

   This method is also brittle because it relies on that assumes one can use STUN to
   classify the NAT behavior. behavior, which was found to be problematic
   [RFC5389].  If the NAT changes the properties of the existing mapping
   and filtering state for example due to load, then the methods will
   fail.

4.1.3.  Embedding STUN in  ALG considerations

   If a NAT supports RTSP

   This section outlines the adaptation ALG (Application Level Gateway) and embedding is not
   aware of the STUN within
   RTSP.  This enables STUN to be used to traverse any type of NAT,
   including symmetric NATs.  This would require protocol changes.

   This NAT traversal solution has limitations:

   1.  It does not work if both RTSP option, service failure may happen,
   because a client and RTSP server are behind
       separate NATs.

   2.  The RTSP server may, for security reasons, refuse to send media
       streams to an IP different from the discovers its public IP address and port numbers,
   and inserts them in its SETUP requests.  When the client RTSP
       requests.

   Deviations from STUN as defined ALG processes
   the SETUP request it may change the destination and port number,
   resulting in RFC 3489: unpredictable behavior.  An ALG should not update
   address fields which contains addresses other than the NATs internal
   address domain.  In cases where the ALG modifies fields unnecessarily
   two alternatives exist:

   1.  We allow RTSP applications  Use TLS to have encrypt the option to perform STUN
       "Shared Secret Request" through RTSP, via extension RTSP TCP connection to RTSP; prevent the ALG
       from reading and modifying the RTSP messages.

   2.  We require  Turn off the STUN server to based NAT traversal mechanism

   As it may be co-located on RTSP server's media
       output ports.

   In order difficult to allow binding discovery without authentication, determine why the STUN
   server embedded in failure occurs, the usage
   of TLS protected RTSP application must ignore authentication tag,
   and message exchange at all times would avoid this
   issue.

4.1.4.  Deployment Considerations

   For the Stand-Alone usage of STUN client embedded in RTSP application must use dummy
   authentication tag.

   If the following applies:

   Advantages:

   o  STUN server is co-located with RTSP server's media output port, an
   RTSP client using RTP transport over UDP a solution first used by SIP applications.  As shown
      above, with little or no changes, the RTSP application can use re-use
      STUN to traverse ALL
   types of NATs.  In as a NAT traversal solution, avoiding the case pit-fall of port and address dependent mapping
   NATs, the party inside the NAT must initiate UDP traffic.  The STUN
   Bind Request, being solving
      a UDP packet itself, can serve as the traffic
   initiating packet.  Subsequently, both the problem twice.

   o  Using STUN Binding Response
   packets and the RTP/RTCP packets can traverse the NAT, regardless of
   whether the does not require RTSP server or modifications; it only
      affects the RTSP client is behind NAT.

   Likewise, if an RTSP server is behind implementation.

   Disadvantages:

   o  Requires a NAT, then an embedded STUN server must co-locate on the RTSP client's RTCP port.  Also it will
   become deployed in the client that needs to disclose his destination public address
   rather than the server so space.

   o  Only works with NATs that the server correctly can determine its
   NAT external source perform endpoint independent and address for the media streams.  In this case, we
   assume that the client has
      dependent mappings.  Address and Port-Dependent filtering NATs
      create some means of establishing TCP connection issues.

   o  Brittle to NATs changing the properties of the RTSP server behind NAT so as to exchange RTSP messages mapping and
      filtering.

   o  Does not work with
   the port and address dependent mapping NATs without
      server modifications.

   o  Will mostly not work if a NAT uses multiple IP addresses, since
      RTSP server.

   To minimize delay, we servers generally require that all media streams to use the same
      IP as used in the RTSP server supporting this
   option must inform its client connection to prevent becoming a DDoS tool.

   o  Interaction problems exist when a RTSP-aware ALG interferes with
      the RTP use of STUN for NAT traversal unless TLS secured RTSP message
      exchange is used.

   o  Using STUN requires that RTSP servers and RTCP ports from where clients support the
   server intend
      updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], because
      it is no longer possible to send out guarantee that RTP and RTCP packets, respectively.  This
   can be done ports are
      adjacent to each other, as required by using the "server_port" parameter in RFC2326, "client_port" and the
   "src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis].  Both are
      "server_port" parameters in RFC2326.

   Transition:

   The usage of STUN can be phased out gradually as the RTSP Transport header.  But in general this strategy will require
   that one first do one SETUP request per media to learn the step of a
   STUN capable server
   ports, then perform or client should be to check the presence of
   NATs.  The removal of STUN checks, followed by a subsequent SETUP
   to change capability in the client port and destination address implementations
   will have to what was learned
   during the STUN checks. wait until there is absolutely no need to use STUN.

4.1.5.  Security Considerations

   To be certain that RTCP works correctly prevent the RTSP end-point (server or
   client) will be required to send and receive RTCP packets server from being used as Denial of Service (DoS)
   attack tools the
   same port.

4.1.4.  Discussion On Co-located STUN Server

   In order RTSP Transport header parameter "destination" and
   "dest_addr" are generally not allowed to use STUN point to traverse "address and port dependent"
   filtering or mapping NATs any IP address
   other than the STUN server needs to be co-located with one the streaming RTSP message originates from.  The RTSP server media output ports.  This creates a de-
   multiplexing problem: we must be able to differentiate a STUN packet
   from a media packet.  This will be done based on heuristics.  A
   common heuristics
   is only prepared to make an exception to this rule when the first byte in client is
   trusted (e.g., through the packet, which works fine
   between STUN and RTP use of a secure authentication process, or RTCP where
   through some secure method of challenging the first byte happens destination to be
   different, but may verify
   its willingness to accept the RTP traffic).  Such a restriction means
   that STUN in general does not work as well with other media transport
   protocols.

4.1.5.  ALG considerations

   If a NAT supports for use cases where RTSP ALG (Application Level Gateway) and is not
   aware of the media
   transport go to different addresses.

   STUN traversal option, service failure may happen,
   because a client discovers its public IP combined with destination address and port numbers,
   and inserts them in its SETUP requests, when the restricted RTSP ALG processes has the SETUP request it may change same
   security properties as the destination and port number,
   resulting in unpredictable behavior.  An ALG should not update
   address fields which contains addresses other than core RTSP.  It is protected from being
   used as a DoS attack tool unless the NATs internal
   address domain.  In cases where attacker has the ALG modifies fields unnecessary
   two alternatives exist:

   1.  The usage of TLS ability to encrypt
   spoof the RTSP TCP connection to prevent
       the ALG from reading and modifying the carrying RTSP messages.

   2.  To turn off the STUN based NAT traversal mechanism

   As it may be difficult to determine why the failure occurs, the usage
   of TLS protected RTSP

   Using STUN's support for message exchange at all times would avoid this
   issue.

4.1.6.  Deployment Considerations

   For the non-embedded usage authentication and secure transport
   of RTSP messages, attackers cannot modify STUN the following applies:

   Advantages:

   o  STUN is a solution first used by SIP applications.  As shown
      above, with little responses or no changes, RTSP application can re-use STUN
   messages (TLS) to change media destination.  This protects against
   hijacking, however as a NAT traversal solution, avoiding client can be the pit-fall initiator of solving a
      problem twice.

   o  Using STUN does not require an attack,
   these mechanisms cannot securely prevent RTSP server modifications; it only
      affects servers being used as
   DoS attack tools.

4.2.  Server Embedded STUN

4.2.1.  Introduction

   This Section describes an alternative to the client implementation.

   Disadvantages:

   o  Requires a stand-alone STUN server deployed usage
   in the public address space.

   o  Only works with NATs previous section that perform endpoint independent and address
      dependent mappings.  Port has quite significantly different
   behavior.

4.2.2.  Embedding STUN in RTSP

   This section outlines the adaptation and address dependent filtering NATs
      create some issues.

   o  Brittle embedding of STUN within
   RTSP.  This enables STUN to NATs changing the properties be used to traverse any type of the NAT mapping and
      filtering.

   o  Does not work with port and NAT,
   including address dependent and Port-Dependent mapping NATs without
      server modifications.

   o  Will mostly NATs.  This would
   require RTSP level protocol changes.

   This NAT traversal solution has limitations:

   1.  It does not work if a NAT uses multiple IP addresses, since both RTSP client and RTSP server generally requires all are behind
       separate NATs.

   2.  The RTSP server may, for security reasons, refuse to send media
       streams to an IP different from the IP in the client RTSP
       requests.

   Deviations from STUN as defined in RFC 5389:

   1.  The RTSP application must provision the client with an identity
       and shared secret to use in the STUN authentication;

   2.  We require STUN server to be co-located on RTSP server's media
       source ports.

   If STUN server is co-located with RTSP server's media source port, an
   RTSP client using RTP transport over UDP can use STUN to traverse ALL
   types of NATs.  In the case of port and address dependent mapping
   NATs, the party on the inside of the NAT must initiate UDP traffic.
   The STUN Binding Request, being a UDP packet itself, can serve as the
   traffic initiating packet.  Subsequently, both the STUN Binding
   Response packets and the RTP/RTCP packets can traverse the NAT,
   regardless of whether the RTSP server or the RTSP client is behind
   NAT (however only one of the can be behind a NAT).

   Likewise, if an RTSP server is behind a NAT, then an embedded STUN
   server must be co-located on the RTSP client's RTCP port.  Also it
   will become the client that needs to disclose his destination address
   rather than the server, so the server can correctly determine its NAT
   external source address for the media streams.  In this case, we
   assume that the client has some means of establishing TCP connection
   to the RTSP server behind NAT so as to exchange RTSP messages with
   the RTSP server, potentially using a proxy or static rules.

   To minimize delay, we require that the RTSP server supporting this
   option must inform the client about the RTP and RTCP ports from where
   the server will send out RTP and RTCP packets, respectively.  This
   can be done by using the "server_port" parameter in RFC2326, and the
   "src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis].  Both are in
   the RTSP Transport header.  But in general this strategy will require
   that one first do one SETUP request per media streams to use learn the same
      IP as used in server
   ports, then perform the RTSP connection to prevent becoming a DDOS tool.

   o  Interaction problems exist when STUN checks, followed by a RTSP-aware ALG interferes with subsequent SETUP
   to change the client port and destination address to what was learned
   during the use of STUN for NAT traversal unless TLS secured RTSP message
      exchange is used.

   o  Using STUN requires checks.

   To be certain that RTSP servers and clients support RTCP works correctly the
      updated RTSP specification, because it is no longer possible end-point (server or
   client) will be required to
      guarantee that RTP send and receive RTCP ports are adjacent to each other, as
      required by packets from the "client_port"
   same port.

4.2.3.  Discussion On Co-located STUN Server

   In order to use STUN to traverse "address and "server_port" parameters in
      RFC2326.

   Transition:

   The usage of port dependent"
   filtering or mapping NATs the STUN can server needs to be phased out gradually as co-located with
   the first step of a
   STUN capable streaming server or client should media output ports.  This creates a de-
   multiplexing problem: we must be able to check the presence of
   NATs. differentiate a STUN packet
   from a media packet.  This will be done based on heuristics.  The removal of
   existing STUN capability heuristics is the first byte in the client implementations
   will have packet and the
   Magic Cookie field (added in RFC5389), which works fine between STUN
   and RTP or RTCP where the first byte happens to wait until there is absolutely no need be different.  Thanks
   to use STUN. the magic cookie field it is unlikely that other protocols would
   be mistaken for a STUN packet, but not assured.

4.2.4.  ALG considerations

   The same ALG traversal considerations as for Stand-Alone STUN applies
   (Section 4.1.3).

4.2.5.  Deployment Considerations

   For the "Embedded STUN" method the following applies:

   Advantages:

   o  STUN is a solution first used by SIP applications.  As shown
      above, with little or no changes, RTSP application can re-use STUN
      as a NAT traversal solution, avoiding the pit-fall of solving a
      problem twice.

   o  STUN has built-in message authentication features, which makes it
      more secure. secure against hi-jacking attacks.  See next section for an
      in-depth security discussion.

   o  This solution works as long as there is only one RTSP end point endpoint in
      the private address realm, regardless of the NAT's type.  There
      may even be multiple NATs (see figure Figure 1 in RFC3489). [RFC5389]).

   o  Compares  Compared to other UDP based NAT traversal methods in this
      document, STUN requires little new protocol development (since
      STUN is already a IETF standard), and most likely less
      implementation effort, since open source STUN server and client
      implementations have become available [STUN-IMPL].  There is the
      need to embed STUN in RTSP server and client, which require a de-multiplexer de-
      multiplexer between STUN packets and RTP/RTCP packets.  There is
      also a need to register the proper feature tags.

   Disadvantages:

   o  Some extensions to the RTSP core protocol, likely signaled by RTSP
      feature tags, must be introduced.

   o  Requires an embedded STUN server to co-locate be co-located on each of the
      RTSP server's media protocol's ports (e.g.  RTP and RTCP ports),
      which means more processing is required to de-multiplex STUN
      packets from media packets.  For example, the de-multiplexer must
      be able to differentiate a RTCP RR packet from a STUN packet, and
      forward the former to the streaming server, and the later latter to the
      STUN server.

   o  Does not support use cases that requires require the RTSP connection and
      the media reception to happen at different addresses, unless the
      servers sequrity
      server's security policy is relaxed.

   o  Interaction problems exist when a RTSP ALG is not aware of STUN
      unless TLS is used to protect the RTSP messages.

   o  Using STUN requires that RTSP servers and clients support the
      updated RTSP specification, specification [I-D.ietf-mmusic-rfc2326bis], and they
      both agree to support the NAT traversal feature.

   o  Increases the setup delay with at least the amount of time it
      takes to perform STUN message exchanges.  Most likely an extra
      SETUP sequence will be required.

   Transition:

   The usage of STUN can be phased out gradually as the first step of a
   STUN capable machine can be to check the presence of NATs for the
   presently used network connection.  The removal of STUN capability in
   the client implementations will have to wait until there is
   absolutely no need to use STUN.

4.1.7.  Security Considerations

   To prevent RTSP server being used as Denial of Service (DoS) attack
   tools the RTSP Transport header parameter "destination" and
   "dest_addr" are generally not allowed to point to any IP address
   other than the one that RTSP message originates from.  The RTSP
   server is only prepared to make an exception of this rule when the
   client is trusted (e.g., through the use of a secure authentication
   process, or through some secure method of challenging the destination
   to verify its willingness to accept the RTP traffic).  Such
   restriction means that STUN does not work for use cases where RTSP
   and media transport goes to different address.

   In terms of security property, STUN combined with destination address
   restricted RTSP has the same security properties as the core RTSP.
   It is protected from being used as a DoS attack tool unless the
   attacker has ability the to spoof the TCP connection carrying RTSP
   messages.

   Using STUN's support for message authentication and secure transport
   of RTSP messages, attackers cannot modify STUN responses or RTSP
   messages to change media destination.  This protects against
   hijacking, however as a client
   STUN capable machine can be to check the initiator presence of an attack,
   these mechanisms cannot securely prevent RTSP servers being NATs for the
   presently used as
   DoS attack tools.

4.2. network connection.  The removal of STUN capability in
   the client implementations will have to wait until there is
   absolutely no need to use STUN.

4.2.6.  Security Considerations

   See Stand-Alone STUN (Section 4.1.5).

4.3.  ICE

4.2.1.

4.3.1.  Introduction

   ICE (Interactive Connectivity Establishment) [RFC5245] is a
   methodology for NAT traversal that has been developed for SIP using
   SDP offer/answer.  The basic idea is to try, in a staggered parallel
   fashion, all possible connection addresses that an end point endpoint may have. be
   reachable by.  This allows the end-point endpoint to use the best available UDP
   "connection" (meaning two UDP end-points capable of reaching each
   other).  The methodology has very nice properties in that basically
   all NAT topologies are possible to traverse.

   Here is how ICE works on at a high level.  End point A collects all
   possible address addresses that can be used, including local IP addresses,
   STUN derived addresses, TURN addresses, etc.  On each local port that
   any of these address and port pairs leads lead to, a STUN server is
   installed.  This STUN server only accepts STUN requests using the
   correct authentication through the use of a username and password.

   End-point A then sends a request to establish connectivity with end-
   point B, which includes all possible destinations "destinations" [RFC5245] to get
   the media through too to A. Note that each of A's published local address/port
   pairs (host candidates and server reflexive base) has a STUN server
   co-located.  B,  B in its turn provides A with all its possible destinations
   for the different media streams.  A and B then uses a STUN client to
   try to reach all the address and port pairs specified by A from its
   corresponding destination ports.  The destinations for which the STUN
   requests have successfully completed complete are then indicated and one is
   selected.

   If B fails to get any STUN response from A, all hope is not lost.
   Certain NAT topologies require multiple tries from both ends before
   successful connectivity is accomplished and therefore requests are
   retransmitted multiple times.  The STUN requests may also result in
   that more connectivity alternatives (destinations) are discovered and
   conveyed in the STUN responses.

4.2.2.

4.3.2.  Using ICE in RTSP

   The usage of ICE for RTSP requires that both client and server be
   updated to include the ICE functionality.  If both parties implement
   the necessary functionality the following steps could provide ICE
   support for RTSP.

   This assumes that it is possible to establish a TCP connection for
   the RTSP messages between the client and the server.  This is not
   trivial in scenarios where the server is located behind a NAT, and
   may require some TCP ports been be opened, or the deployment of proxies,
   etc.

   The negotiation of ICE in RTSP of necessity will work different than
   in SIP with SDP offer/answer.  The protocol interactions are
   different and thus the possibilities for transfer of states are also
   somewhat different.  The goal is also to avoid introducing extra
   delay in the setup process at least for when the server is using a
   public address and the client is either having a public address or is
   behind NAT(s).  This process is only intended to support PLAY mode,
   i.e. media traffic flows from server to client.

   1.  The ICE usage begins in the SDP.  The SDP for the service
       indicates that ICE is supported at the server.  No candidates can
       be given here as that would not work with the on demand, DNS load
       balancing, etc., that make a have the SDP indicate a resource on a
       server park rather than a specific machine.

   2.  The client gathers addresses and puts together its candidate candidates for
       each media stream indicated in the session description.

   3.  In each SETUP request the client includes its candidates in a an
       ICE specific transport specification.  This indicates for the
       server the ICE support by the client.  One candidate is the most
       prioritized candidate and here the prioritization for this
       address should be somewhat different compared to SIP.  High
       performance rather than always successful is to recommended recommended, as it
       is most likely to be a server in the public.

   4.  The server responds to the SETUP (200 OK) for each media stream
       with its candidates.  A server with a public address usually only
       provides a single ICE candidate.  Also here one candidate is the
       server primary address.

   5.  The connectivity checks are performed.  For the server the
       connectivity checks from the server to the clients have an
       additional usage.  They verify that there is someone willingly to
       receive the media, thus protecting itself from performing
       unknowingly an DoS attack.

   6.  Connectivity checks from the client promoting a candiadate candidate pair
       was
       were successful.  Thus no further SETUP requests are necessary
       and processing can proceed with step 7.  If another address than
       the primary has been verified by the client to work, that address
       may then be promoted for usage in a SETUP request (Goto (Go to 7).  If
       the checks for the availble available candidates failed and If if further
       candidates have been derived during the connectivity checks, then
       those can be signalled in new candidate lines in a SETUP request
       updating the list (Goto (Go to 5).

   7.  Client issues PLAY request.  If the server also has completed its
       connectivity checks for the promoted candidate pair (based on
       username as it may be derived addresses if the client was behind
       NAT) then it can directly answer 200 OK (Goto (Go to 8).  If the
       connectivity check has not yet completed it responds with a 1xx
       code to indicate that it is verifying the connectivity.  If that
       fails within the set timeout timeout, an error is reported back.  Client
       needs to go back to 6.

   8.  Process completed and media can be delivered.  ICE candidates not
       used may be released.

   To keep media paths alive the client needs to periodically send data
   to the server.  This will be realized with STUN.  RTCP sent by client
   should be able to keep RTCP open but STUN will also be used based on
   the same motivations as for ICE for SIP.

4.2.3.

4.3.3.  Implementation burden of ICE

   The usage of ICE will require that a number of new protocols and new
   RTSP/SDP features be implemented.  This makes ICE the solution that
   has the largest impact on client and server implementations amongst
   all the NAT/FW NAT/Firewall traversal methods in this document.

   RTSP server implementation requirements are:

   o  STUN server features

   o  limited STUN client features

   o  SDP generation with more parameters.

   o  RTSP error code for ICE extension

   RTSP client implantation implementation requirements are:

   o  Limited STUN server features

   o  Limited STUN client features

   o  RTSP error code and ICE extension

4.2.4.

4.3.4.  Deployment Considerations

   Advantages:

   o  Solves NAT connectivity discovery for basically all cases as long
      as a TCP connection between them can be established.  This
      includes servers behind NATs.  (Note that a proxy between address
      domains may be required to get TCP through).

   o  Improves defenses against DDOS DDoS attacks, as media receiving client
      requires authentications, via STUN on its media reception ports.

   Disadvantages:

   o  Increases the setup delay with at least the amount of time it
      takes for the server to perform its STUN requests.

   o  Assumes that it is possible to de-multiplex between the packets of
      the media protocol and STUN packets.

   o  Has fairly high implementation burden put on both RTSP server and
      client.

4.2.5.

4.3.5.  Security Consideration

   One should review the security consideration section of ICE and STUN
   to understand that ICE contains some potential issues.  However these
   can be avoided by a correctly utilizing using ICE in RTSP.  In fact ICE do does help
   avoid the DDoS attack issue with RTSP substantially as it reduces the
   possibility for a DDoS using RTSP servers to attackers that are on-
   path between the RTSP server and the target and capable of
   intercepting the STUN connectivity check packets and correctly send a
   response to the server.

4.3.  Symmetric RTP

4.3.1.

4.4.  Latching

4.4.1.  Introduction

   Symmetric RTP

   Latching is a NAT traversal solution that is based on requiring RTSP
   clients to send UDP packets to the server's media output ports.
   Conventionally, RTSP servers send RTP packets in one direction: from
   server to client.  Symmetric RTP  Latching is similar to connection-oriented
   traffic, where one side (e.g., the RTSP client) first "connects" by
   sending a RTP packet to the other side's RTP port, the recipient then
   replies to the originating IP and port.  This method is also referred
   to as "Late binding".  It requires that all RTP/RTCP transport is
   done symmetrical, i.e.  Symmetric RTP [RFC4961].

   Specifically, when the RTSP server receives the "connect" RTP latching packet
   (a.k.a.  FW hole-punching packet, since it is used to punch a hole in the FW/NAT
   Firewall/NAT and to aid the server for port binding and address
   mapping) from its client, it copies the source IP and Port number and
   uses them as delivery address for media packets.  By having the
   server send media traffic back the same way as the client's packet
   are sent to the server, address mappings will be honored.  Therefore
   this technique works for all types of NATs. NATs, given that the server is
   not behind a NAT.  However, it does require server modifications.
   Unless there is built-in protection mechanism,
   symmetric RTP latching is very
   vulnerable to DDOS DDoS attacks, because attackers can simply forge the
   source IP & Port of the binding latching packet.  Using the rule for restriciting
   restricting IP address to that the one of the signalling signaling connection will
   need to be applied here also.

4.3.2.  However, that does not protect against
   hijacking from another client behind the same NAT.  This can become a
   serious issue in deployments with CGNs.

4.4.2.  Necessary RTSP extensions

   To support symmetric RTP Latching, the RTSP signaling must be extended to allow the
   RTSP client to indicate that it will use symmetric RTP. Latching.  The client also
   needs to be able to signal its RTP SSRC to the server in its SETUP
   request.  The RTP SSRC is used to establish some basic level of
   security against hijacking attacks. attacks or simply avoid mis-association
   when multiple clients are behind the same NAT.  Care must be taken in
   choosing client's clients' RTP SSRC.  First, it must be unique within all the
   RTP sessions belonging to the same RTSP session.  Secondly, if the
   RTSP server is sending out media packets to multiple clients from the
   same send port, the RTP SSRC needs to be unique amongst those
   clients' RTP sessions.  Recognizing that there is a potential that
   RTP SSRC collision collisions may occur, the RTSP server must be able to signal
   to a client that a collision has occurred and that it wants the
   client to use a different RTP SSRC carried in the SETUP response or
   use unique ports per RTSP session.  Using unique ports limits an RTSP
   server in the number of session sessions it can simultaneously handle per
   interface IP addresses.

4.3.3.

4.4.3.  Deployment Considerations

   Advantages:

   o  Works for all types of NATs, including those using multiple IP
      addresses. client-facing NATs.  (Requirement 1 in
      Section 3).

   o  Have  Has no interaction problems with any RTSP ALG changing the
      client's information in the transport header.

   Disadvantages:

   o  Requires modifications to both RTSP server and client.

   o  Limited to work with servers that have an public IP address.

   o  The format of the RTP packet for "connection setup" (a.k.a FW
      Latching packet) is yet to be defined.  One possibility is to use
      RTP No-Op packet format in [I-D.ietf-avt-rtp-no-op].

   o  Has  SSRC management if RTP is used for latching due to risk for mis-
      association of clients to RTSP sessions at the same server if SSRC
      collision occurs.

   o  Has worse security situation as than STUN due to lack of STUN message
      authentication and will need to use address restrictions.

4.3.4.

4.4.4.  Security Consideration

   Symmetric RTP's

   Latching's major security issue is that RTP streams can be hijacked
   and directed towards any target that the attacker desires unless
   address restricitons restrictions are used.  In the case of NATs with multiple
   clients on the inside of them, hijacking can still occur.  This
   becomes a significant threat in the context of carrier grade NATs
   (CGN).

   The most serious security problem is the deliberate attack with the
   use of a RTSP client and symmetric RTP. Latching.  The attacker uses RTSP to setup a
   media session.  Then it uses symmetric RTP Latching with a spoofed source address
   of the intended target of the attack.  There is no defense against
   this attack other than restricting the possible bind address a latching
   packet can come from to be the same as the RTSP TCP connection arrived on. are from.
   This prevents symmetric RTP Latching to be used in use cases that require differet different
   addresses for media destination and signalling.

   A hijack attack can also be performed in various ways.  The basic
   attack is based on the ability to read the RTSP signaling packets in
   order to learn the address and port the server will send from and
   also the SSRC the client will use.  Having this information the
   attacker can send its own NAT-traversal RTP Latching packets containing the correct RTP
   SSRC to the correct address and port on the server.  The
   destination of the packets is set as RTSP server
   will then use the source IP and port in these
   RTP packets. from the Latching packet as the
   destination for the media packets it sends.

   Another variation of this attack is for a man in the middle to modify
   the RTP binding latching packet being sent by a client to the server by
   simply changing the source IP to the target one desires to attack.

   One can fend off the first attack by applying encryption to the RTSP
   signaling transport.  However, the second variation is impossible to
   defend against.  As a NAT re-writes the source IP and port this
   cannot be authenticated, but authentication is required in order to
   protect against this type of DOS DoS attack.

   Yet another issues is that these attacks also can be used to deny the
   client the service he desire it desires from the RTSP server completely.  For a
   man in the middle capable of reading the signalling signaling traffic or
   intercepting the binding latching packets can completely deny the client
   service by modifying or originating binding latching packets of itself.

   The amount of random nonce used non-guessable material in the binding latching packet
   determines how well
   symmetric RTP Latching can fend off stream-hijacking performed
   by parties that are not "man-in-the-middle".  This proposal uses the
   32-bit RTP SSRC field to this effect.  Therefore it is important that
   this field is derived with a non-predictable randomizer. random number generator.
   It should not be possible by knowing the algorithm used and a couple
   of basic facts, to derive what random number a certain client will
   use.

   An attacker not knowing the SSRC but aware of which port numbers that
   a server sends from can deploy a brute force attack on the server by
   testing a lot of different SSRCs until it finds a matching one.
   Therefore a server SHOULD could implement functionality that blocks packets
   to ports or from sources that receive or send multiple FW Latching
   packets (i.e. the packet that is sent to the
   server for FW traversal) with different invalid SSRCs, especially when they are coming
   from the same IP/Port.  Note that this mitigation in itself opens up
   a new venue for DoS attacks against legit users trying to latch.

   To improve the security against attackers the amount of random tag's length
   material could be increased.  To achieve a longer random tag while
   still using RTP and RTCP, it will be necessary to develop RTP and
   RTCP payload formats for carrying the random tag.

4.3.5. material.

4.5.  A Variation to Symmetric RTP

   Symmetric RTP Latching

4.5.1.  Introduction

   Latching as described above requires the usage of a valid RTP format in
   as the FW Latching packet, which is i.e. the first packet that the client sends
   to the server to set up virtual RTP connection.  There is currently existed no
   appropriate RTP packet format for this purpose, although the No-Op
   format was a proposal to fix the problem [I-D.ietf-avt-rtp-no-op].
   However, that work has
   been was abandoned.  There exists a RFC that discusses
   the implication of different type of packets as keep-alives for RTP
   [RFC6263] and its findings are very relevant to the FW format of the
   Latching packet.

   Meanwhile, there has been FW NAT/Firewall traversal techniques deployed
   in the wireless streaming market place that use non-RTP messages as FW
   Latching packets.  This section attempts to summarize describes a variant based on a subset
   of those solutions that happens to use a variation to alters the standard symmetric
   RTP previously described Latching
   solution.

4.5.2.  Necessary RTSP extensions

   In this variation of symmetric RTP, Latching, the FW Latching packet is a small UDP
   packet that does not contain an RTP header.  Hence the solution can no
   longer be called symmetric RTP, yet it employs the same technique for
   FW traversal.  In response to client's FW
   Latching packet, the RTSP server sends back a similar FW Latching packet
   as a confirmation so that the client can stop the so called "connection
   phase" of this NAT traversal technique.  Afterwards, the client only
   has to periodically send FW Latching packets as keep-alive messages for
   the NAT mappings.

   The server listens on its RTP-media output port, and tries to decode
   any received UDP packet as FW Latching packet.  This is valid since an
   RTSP server is not expecting RTP traffic from the RTSP client.  Then,
   it can correlate the FW Latching packet with the RTSP client's session
   ID or the client's SSRC, and record the NAT bindings accordingly.
   The server then sends a FW Latching packet as the response to the
   client.

   The FW Latching packet can contain the SSRC to identify the RTP stream,
   and care must be taken if the packet is bigger than 12 bytes,
   ensuring that it is distinctively different from RTP packets, whose
   header size is 12 bytes.

   RTSP signaling can be added to do the following:

   1.  Enables  Enable or disables disable such FW Latching message exchanges.  When the FW/NAT
       Firewall/NAT has an RTSP-aware ALG, it is possible to disable FW
       Latching message exchange and let the ALG works work out the address
       and port mappings.

   2.  Configures  Configure the number of re-tries and the re-try interval of the
       FW
       Latching message exchanges.

   Such FW packets may also contain digital signatures to support three-
   way handshake based receiver authentications, so as to prevent DDoS
   attacks described before.

4.5.3.  Deployment Considerations

   This approach has the following advantages when compared with the
   symmetric RTP approach:
   Latching approach (Section 4.4):

   1.  There is no need to define RTP payload format for FW Firewall
       traversal, therefore it is simple to use, implement and
       administer (Requirement 4 in Section 3), although instead a binding Latching
       protocol must be defined.

   2.  When properly defined, this kind of FW message Latching packet exchange can
       also authenticate RTP receivers, so as to prevent DDoS attacks for
       dual-hosted RTSP client.  By dual-hosted RTSP client we mean the
       kind that uses one "perceived" IP address for RTSP message
       exchange, and a different "perceived" IP address for RTP
       reception.  (Requirement 5 in Section 3). receivers, to prevent hijacking attacks.

   This approach has the following disadvantages when compared with the
   symmetric RTP
   Latching approach:

   1.  RTP traffic is normally accompanied by RTCP traffic.  This
       approach needs to rely on RTCP RRs and SRs to enable NAT
       traversal for RTCP endpoints, use RTP/RTCP Multiplexing
       [RFC5761], or use the same type of FW messages Latching packets also for RTCP
       endpoints.

   2.  The server's sender SSRC for the RTP stream must be signaled in
       RTSP's SETUP response, in RTP stream or other session
       Identity information must be signaled in RTSP's SETUP response,
       in the Transport header of the RTSP SETUP response.

4.5.4.  Security Considerations

   Compared to the security properties of Latching this variant is
   slightly improved.  First of all it allows for a larger random field
   in the Latching packets which makes it more unlikelier for an off-
   path attacker to succeed in a hi-jack attack.  Secondly the
   confirmation allows the client to know when Latching works and when
   it didn't and thus restart the Latching process by updating the SSRC.
   Thirdly if an authentication mechanism are included in the latching
   packet hijacking attacks can be prevented.

   Still the main security issue remain that the RTSP server can't know
   that the source address in the latching packet was coming from a RTSP
   client wanting to receive media and not one that likes to direct the
   media traffic to an DoS target.

4.6.  Three Way Latching
4.6.1.  Introduction

   The three way Latching is an attempt to try to resolve the most
   significant security issues for both previously discussed variants of
   Latching.  By adding a server request response exchange directly
   after the initial latching the server can verify that the target
   address present in the latching packet is an active listener and
   confirm its desire to establish a media flow.

4.6.2.  Necessary RTSP extensions

   Uses the same RTSP extensions as the alternative latching method
   (Section 4.5) uses.  The extensions for this variant are only in the
   format and transmission of the Latching packets.

   The client to server latching packet is similar to the Alternative
   Latching (Section 4.5), i.e. an UDP packet with some session
   identifier and a random value.  When the server responds to the
   Latching packet with a Latching confirmation, it includes a random
   value (Nonce) of its own in addition to echoing back the one the
   client sent.  Then a third messages is added to the exchange.  The
   client acknowledges the Transport header reception of the Latching confirmation
   message and echoes back the server's nonce.  Thus confirming that the
   Latched address goes to a RTSP SETUP
       response. client that initiated the latching and
   is actually present at that address.  The RTSP server will refuse to
   send any media until the Latching Acknowledgement has been received
   with a valid nonce.

4.6.3.  Deployment Considerations

   A solution with a 3-way handshaking handshake and its own FW Latching packets can be
   compared with ICE the ICE-based solution (Section 4.3) and have the
   following differencies: differences:

   o  Only works for servers with public IP addresses compared to any
      type of server

   o  Is somewhat  May be simpler to implement due to the avoidance of the ICE
      prioritization and checkboard check-board mechanisms.

   However, a 3-way binding Latching protocol is very similar to using STUN in
   both directions as binding Latching and verification protocol.  Using STUN
   would remove the need for implementing a new protocol.

4.4.

4.7.  Application Level Gateways

4.4.1.
4.7.1.  Introduction

   An Application Level Gateway (ALG) reads the application level
   messages and performs necessary changes to allow the protocol to work
   through the middle box.  However this behavior has some problems in
   regards to RTSP:

   1.  It does not work when the RTSP protocol is used with end-to-end
       security.  As the ALG can't inspect and change the application
       level messages the protocol will fail due to the middle box.

   2.  ALGs need to be updated if extensions to the protocol are added.
       Due to deployment issues with changing ALGs this may also break
       the end-to-end functionality of RTSP.

   Due to the above reasons it is NOT RECOMMENDED not recommended to use an RTSP ALG in
   NATs.  This is especially important for NATs targeted to home users
   and small office environments, since it is very hard to upgrade NATs
   deployed in home or SOHO (small office/home office) environment.

4.4.2.

4.7.2.  Outline On how ALGs for RTSP work

   In this section, we provide a step-by-step outline on how one should could
   go about writing an ALG to enable RTSP to traverse a NAT.

   1.  Detect any SETUP request.

   2.  Try to detect the usage of any of the NAT traversal methods that
       replace the address and port of the Transport header parameters
       "destination" or "dest_addr".  If any of these methods are used,
       then the ALG SHOULD NOT should not change the address.  Ways to detect that
       these methods are used are:

       *  For embedded STUN, it would be to watch for a feature tag,
          like "nat.stun".  If any of those exists in the "supported",
          "proxy-require", or "require" headers of the RTSP exchange.

       *  For non-embedded stand alone STUN and TURN based solutions: This can in
          some case be
          detected by inspecting the "destination" or "dest_addr"
          parameter.  If it contains either one of the NAT's external IP
          addresses or a public IP address. address then such a solution is in
          use.  However if multiple NATs are used this detection may
          fail.  Remapping should only be done for addresses belonging
          to the NATs NAT's own private address space.

       Otherwise continue to the next step.

   3.  Create UDP mappings (client given IP/port <-> external IP/port)
       where needed for all possible transport specification specifications in the
       transport header of the request found in (1).  Enter the public external
       address and port(s) of these mappings in transport header.
       Mappings SHALL shall be created with consecutive public port number numbers
       starting on an even number for RTP for each media stream.
       Mappings SHOULD should also be given a long timeout period, at least 5
       minutes.

   4.  When the SETUP response is received from the server server, the ALG MAY may
       remove the unused UDP mappings, i.e. the ones not present in the
       transport header.  The session ID SHOULD should also be bound to the UDP
       mappings part of that session.

   5.  If SETUP response settles on RTP over TCP or RTP over RTSP as
       lower transport, do nothing: let TCP tunneling to take care of NAT
       traversal.  Otherwise go to next step.

   6.  The ALG SHOULD should keep alive the UDP mappings belonging to the an RTSP
       session as long as: an RTSP messages with the session's ID has
       been sent in the last timeout interval, or a UDP messages are has
       been sent on any of the UDP mappings during the last timeout
       interval.

   7.  The ALG MAY may remove a mapping as soon a TEARDOWN response has been
       received for that media stream.

4.4.3.

4.7.3.  Deployment Considerations

   Advantage:

   o  No impact on either client or server

   o  Can work for any type of NATs

   Disadvantage:

   o  When deployed they are hard to update to reflect protocol
      modifications and extensions.  If not updated they will break the
      functionality.

   o  When end-to-end security is used used, the ALG functionality will fail.

   o  Can interfere with other type types of traversal mechanisms, such as
      STUN.

   Transition:

   An RTSP ALG will not be phased out in any automatically automatic way.  It must be
   removed, probably through the removal of the NAT it is associated
   with.

4.4.4.

4.7.4.  Security Considerations

   An ALG will not work when whit deployment of end-to-end RTSP signaling
   security.  Therefore deployment of ALG will likely result in clients
   located behind NATs not using end-to-end security.

   The creation of an UDP mapping based on the signalling message has
   some potential security implications.  First of all if the RTSP
   client releases its ports and another application are assigned these
   instead it could receive RTP media as long as the mappings exist and
   the RTSP server has failed to be signalled or notice the lack of
   client response.

   An NAT with RTSP ALG that assigns mappings based on SETUP requests
   could potentially become victim of an resource exhaustion attack.  If
   an attacker creates a lot of RTSP sessions, even without starting
   media transmission could exhaust the pool of available UDP ports on
   the NAT.  Thus only a limited number of UDP mappings should be
   allowed to be created by the RTSP signaling
   security.  Therefore deployment of ALG will likely result in that
   clients located behind NATs will not use end-to-end security.

4.5. ALG.

4.8.  TCP Tunneling

4.5.1.

4.8.1.  Introduction

   Using a TCP connection that is established from the client to the
   server ensures that the server can send data to the client.  The
   connection opened from the private domain ensures that the server can
   send data back to the client.  To send data originally intended to be
   transported over UDP requires the TCP connection to support some type
   of framing of the media data packets.  Using TCP also results in that the
   client has having to accept that real-time performance may no longer can be
   possible. impacted.
   TCP's problem of ensuring timely deliver delivery was one of the reasons why
   RTP was developed.  Problems that arise with TCP are: head-of-
   line head-of-line
   blocking, delay introduced by retransmissions, highly varying rate
   due to the congestion control algorithm.

4.5.2.  If sufficient amount of
   buffering (several seconds) in the receiving client can be tolerated
   then TCP clearly can work.

4.8.2.  Usage of TCP tunneling in RTSP

   The RTSP core specification [I-D.ietf-mmusic-rfc2326bis] supports
   interleaving of media data on the TCP connection that carries RTSP
   signaling.  See section 14 in [I-D.ietf-mmusic-rfc2326bis] for how to
   perform this type of TCP tunneling.  There also exist exists another way of
   transporting RTP over TCP defined in Appendix C.2. C.2 in
   [I-D.ietf-mmusic-rfc2326bis].  For signaling and rules on how to
   establish the TCP connection in lieu of UDP, see appendix C.2 in
   [I-D.ietf-mmusic-rfc2326bis].  This is based on the framing of RTP
   over the TCP connection as described in RFC 4571 [RFC4571].

4.5.3.

4.8.3.  Deployment Considerations

   Advantage:

   o  Works through all types of NATs where the RTSP server is in the open. not NATed
      or at least reachable like it was not.

   Disadvantage:

   o  Functionality needs to be implemented on both server and client.

   o  Will not always meet multimedia stream's real-time requirements.

   Transition:

   The tunneling over RTSP's TCP connection is not planned to be phased-
   out.  It is intended to be a fallback mechanism and for usage when
   total media reliability is desired, even at the potential price of
   loss of real-time properties.

4.5.4.

4.8.4.  Security Considerations

   The TCP tunneling of RTP has no known security problem problems besides those
   already presented in the RTSP specification.  It is not possible to
   get any amplification effect that is desired for denial of service attacks due to
   TCP's flow control.  A possible security consideration, when session
   media data is interleaved with RTSP, would be the performance
   bottleneck when RTSP encryption is applied, since all session media
   data also needs to be encrypted.

4.6.

4.9.  TURN (Traversal Using Relay NAT)

4.6.1.

4.9.1.  Introduction

   Traversal Using Relay NAT (TURN) [RFC5766] is a protocol for setting
   up traffic relays that allows allow clients behind NATs and firewalls to
   receive incoming traffic for both UDP and TCP.  These relays are
   controlled and have limited resources.  They need to be allocated
   before usage.  TURN allows a client to temporarily bind an address/
   port pair on the relay (TURN server) to its local source address/port
   pair, which is used to contact the TURN server.  The TURN server will
   then forward packets between the two sides of the relay.  To prevent
   DOS
   DoS attacks on either recipient, the packets forwarded are restricted
   to the specific source address.  On the client side it is restricted
   to the source setting up the mapping.  On the external side this is
   limited to the source address/port pair of the first packet arriving
   on the binding.  After the first packet has arrived the mapping is
   "locked down" to that address.  Packets from any other source on this
   address will be discarded.  Using a TURN server makes it possible for
   a RTSP client to receive media streams from even an unmodified RTSP
   server.  However the problem is those RTSP servers most likely
   restrict media destinations to no other IP address than the one the
   RTSP message arrives. arrives from.  This means that TURN could only be used
   if the server knows and accepts that the IP belongs to a TURN server
   and the TURN server can't be targeted at an unknown address or also
   the RTSP connection is relayed through the same TURN server.

4.6.2.

4.9.2.  Usage of TURN with RTSP

   To use a TURN server for NAT traversal, the following steps should be
   performed.

   1.  The RTSP client connects with the RTSP server.  The client
       retrieves the session description to determine the number of
       media streams.  To avoid the issue with having RTSP connection
       and media traffic from different addresses also the TCP
       connection must be done through the same TURN server as the one
       in the next step.  This will require the usage of TURN for TCP
       [RFC6062].

   2.  The client establishes the necessary bindings on the TURN server.
       It must choose the local RTP and RTCP ports that it desires to
       receive media packets.  TURN supports requesting bindings of even
       port numbers and continuous ranges.

   3.  The RTSP client uses the acquired address and port mappings in
       the RTSP SETUP request using the destination header.  Note that
       the server is required to have a mechanism to verify that it is
       allowed to send media traffic to the given address.  The server
       SHOULD
       should include its RTP SSRC in the SETUP response.

   4.  Client  The client requests that the Server server starts playing.  The server
       starts sending media packet packets to the given destination address and
       ports.

   5.  The first media packet to arrive at the TURN server on the
       external port causes "lock down"; then TURN server forwards the
       media packets to the RTSP client.

   6.  When media arrives at the client, the client should try to verify
       that the media packets are from the correct RTSP server, by
       matching the RTP SSRC of the packet.  Source  The source IP address of
       this packet will be that of the TURN server and can therefore not
       be used to verify that the correct source has caused lock down.

   7.  If the client notices that some other source has caused lock down
       on the TURN server, the client should create new bindings and
       change the session transport parameters to reflect the new
       bindings.

   8.  If the client pauses and media are is not sent for about 75% of the
       mapping timeout the client should use TURN to refresh the
       bindings.

4.6.3.

4.9.3.  Deployment Considerations

   Advantages:

   o  Does not require any server modifications.

   o  Works for any types type of NAT as long as the RTSP server has public
      reachable IP address.

   Disadvantage:

   o  Requires another network element, namely the TURN server.

   o  A TURN server for RTSP is may not scale since the number of sessions
      it must forward is proportional to the number of client media
      sessions.

   o  TURN server becomes a single point of failure.

   o  Since TURN forwards media packets, it necessarily introduces
      delay.

   o  An RTSP ALG MAY may change the necessary destinations parameter.  This
      will cause the media traffic to be sent to the wrong address.

   Transition:

   TURN is not intended to be phase-out phased-out completely, see chapter 11.2 Section 19 of
   [RFC5766].  However the usage of TURN could be reduced when the
   demand for having NAT traversal is reduced.

4.6.4.

4.9.4.  Security Considerations

   An eavesdropper of RTSP messages between the RTSP client and RTSP
   server will be able to do a simple denial of service attack on the
   media streams by sending messages to the destination address and port
   present in the RTSP SETUP messages.  If the attacker's message can
   reach the TURN server before the RTSP server's message, the lock down
   can be accomplished towards some other address.  This will result in
   that
   the TURN server will drop dropping all the media server's packets when they
   arrive.  This can be accomplished with little risk for the attacker
   of being caught, as it can be performed with a spoofed source IP.
   The client may detect this attack when it receives the lock down
   packet sent by the attacker as being mal-formatted mal-formed and not corresponding
   to the expected context.  It will also notice the lack of further
   incoming packets.  See bullet 7 in Section 4.6.2. 4.9.2.

   The TURN server can also become part of a denial of service attack
   towards any victim.  To perform this attack the attacker must be able
   to eavesdrop on the packets from the TURN server towards a target for
   the DOS DoS attack.  The attacker uses the TURN server to setup a RTSP
   session with media flows going through the TURN server.  The attacker
   is in fact creating TURN mappings towards a target by spoofing the
   source address of TURN requests.  As the attacker will need the
   address of these mappings he must be able to eavesdrop or intercept
   the TURN responses going from the TURN server to the target.  Having
   these addresses, he can set up a RTSP session and starts start delivery of
   the media.  The attacker must be able to create these mappings.  The
   attacker in this case may be traced by the TURN username in the
   mapping requests.

   The first attack can be made very hard by applying transport security
   for the RTSP messages, which will hide the TURN servers address and
   port numbers from any eavesdropper.

   The second attack requires that the attacker have has access to a user
   account on the TURN server to be able set up the TURN mappings.  To
   prevent this attack the RTSP server shall needs to verify that the ultimate
   target destination accept this media stream.  Which would require
   something like ICE's connectivity checks being run between the RTSP
   server and the RTSP client.

5.  Firewalls

   Firewalls exist for the purpose of protecting a network from traffic
   not desired by the firewall owner.  Therefore it is a policy decision
   if a firewall will let RTSP and its media streams through or not.
   RTSP is designed to be firewall friendly in that it should be easy to
   design firewall policies to permit passage of RTSP traffic and its
   media streams.

   The firewall will need to allow the media streams associated with a
   RTSP session to pass through it.  Therefore the firewall will need an
   ALG that reads RTSP SETUP and TEARDOWN messages.  By reading the
   SETUP message the firewall can determine what type of transport and
   from where where, the media streams stream packets will use. be sent.  Commonly there
   will be the need to open UDP ports for RTP/RTCP.  By looking at the
   source and destination addresses and ports the opening in the
   firewall can be minimized to the least necessary.  The opening in the
   firewall can be closed after a TEARDOWN message for that session or
   the session itself times out.

   Simpler firewalls do allow a client to receive media as long as it
   has sent packets to the target.  Depending on the security level this
   can have the same behavior as a NAT.  The only difference is that no
   address translation is done.  To be able to use such a firewall a client would
   need to implement one of the above described NAT traversal methods
   that include sending packets to the server to open up the mappings.

6.  Comparision  Comparison of NAT traversal techniques

   This section evaluates the techniques described above against the
   requirements listed in section Section 3.

   In the following table, the columns correspond to the numbered
   requirements.  For instance, the column under R1 corresponds to the
   first requirement in section Section 3: MUST must work for all flavors of NATs.
   The rows represent the different FW NAT/Firewall traversal techniques.
   SymRTP
   Latch is short for symmetric RTP, "V.SymRTP" Latching, "V. Latch" is short for "variation of symmetric RTP"
   Latching" as described in section Section 4.3.5. 4.5. "3-W Latch" is short for the
   Three Way Latching described in Section 4.6.

   A Summary of the requirements are:

   R1

   R1:  Work for all flavors of NATs
   R2 Most

   R2:  Must work with Firewalls, including them those with ALGs

   R3

   R3:  Should have minimal impact on clients not behind NATs

   R4 NATs, counted
      in minimal number of additional RTTs

   R4:  Should be simple to use, Implement and administrate.

   R5 administer.

   R5:  Should provide a mitigation against DDoS attacks

   -----------------------------------------------+

   The following considerations are also added to requirements:

   C1:  Will solution support both Clients and Servers behind NAT

   C2:  How Robust is the solution to changing NAT behaviors

   ------------+------+------+------+------+------+------+------+
               |  R1  |  R2  |  R3  |  R4  R4  |  R5  |  C1  |  C2  |
   ------------+------+------+------+------+------+------+------+
    STUN       | No   | Yes  |  1   | Maybe| No   | No   | No   |
   ------------+------+------+------+------+------+------+------+
    Emb. STUN  | Yes  | Yes  |  2   | Maybe| No   | No   | Yes  |
   ------------+------+------+------+------+------+------+------+
    ICE        | Yes  | Yes  | 2.5  | No   | Yes  | Yes  | Yes  |
   ------------+------+------+------+------+------+------+------+
    Latch      | Yes  | Yes  |  1   | Maybe| No   | No   | Yes  |  R5
   ------------+------+------+------+------+------+------+------+
    V. Latch   |
   ------------+------+------+------+------+------+
    STUN Yes  | Yes  |  1   | Yes  | No   | Maybe| No   |
   ------------+------+------+------+------+------+
    ICE Yes  |
   ------------+------+------+------+------+------+------+------+
    3-W Latch  | Yes  | Yes  |  No 1.5  | Maybe| Yes  | No   | Yes  |
   ------------+------+------+------+------+------+
    SymRTP
   ------------+------+------+------+------+------+------+------+
    ALG        |(Yes) | Yes  | Yes  0   | No   | Yes  |Maybe  | No   |
   ------------+------+------+------+------+------+
    V. SymRTP  | Yes  | Yes
   ------------+------+------+------+------+------+------+------+
    TCP Tunnel | Yes  | Yes  |future|
   ------------+------+------+------+------+------+
    3-W SymRTP  | Yes 1.5  | Yes  | Yes  | Maybe| No   | Yes  |
   ------------+------+------+------+------+------+
   ------------+------+------+------+------+------+------+------+
    TURN       | Yes  | Yes  | No  1   | No   | Yes  |(Yes) |
   ------------------------------------------------ Yes  |
   ------------+------+------+------+------+------+------+------+

            Figure 1: Comparison of fulfillment of requirements

   Looking at Figure 1 one would draw the conclusion that using TCP
   Tunneling or Three-Way Latching is the solutions that best fulfill
   the requirements.  The different techniques was were discussed in the
   MMUSIC WG.  It was established that the WG would pursue an ICE based
   solution due to its generality and capability of handle handling also
   servers delivering media from behind NATs.  TCP Tunneling is likely
   to be available as an alternative, due to its specification in the
   main RTSP specification.  Thus it can be used if desired and the
   potential downsides of using TCP is acceptable in particular
   deployments.  When it comes to Three-Way Latching it is a very
   competitive technique given that you don't need support for RTSP
   servers behind NATs.  There has been were some discussion in the WG if the
   increased implementation burden of ICE is sufficiently motivated
   compared to a 3-W SymRTP the Three-Way Latching solution for this generality.
   In the end the authors believe that reuse of ICE, the greater
   flexibility and anyway need to deploy a new solution was the decisive
   factors.

   The ICE based RTSP NAT traversal solution is specified in "A Network
   Address Translator (NAT) Traversal mechanism for media controlled by
   Real-Time Streaming Protocol (RTSP)" [I-D.ietf-mmusic-rtsp-nat].

7.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.

8.  Security Considerations

   In the preceding sessions sections we have discussed security merits of each and
   every NAT/FW the
   different NAT/Firewall traversal methods for RTSP discussed here.  In
   summary, the presence of NAT(s) is a security risk, as a client
   cannot perform source authentication of its IP address.  This
   prevents the deployment of any future RTSP extensions providing
   security against hijacking of sessions by a man-in-the-middle.

   Each of the proposed solutions has security implications.  Using STUN
   will provide the same level of security as RTSP with out transport
   level security and source authentications; authentications, as long as the server does
   not grant a client request to send allow media to be sent to a different IP addresses. IP-address than the RTSP
   client request was sent from.  Using symmetric RTP Latching will have a higher risk
   of session hijacking or denial of service than normal RTSP.  The
   reason is that there exists a probability that an attacker is able to
   guess the random tag bits that the client uses to prove its identity when
   creating the address bindings.  This can be solved in the variation
   of symmetric RTP
   (section 6.3.5) Latching (Section 4.5) with authentication features.  Still both
   those variants of Latching is vulnerable against deliberate attack
   from the RTSP client to redirect the media stream requested to any
   target assuming it can spoof the source address.  This security
   vulnerability is solved by performing a Three-way Latching procedure
   as discussed in Section 4.6.  The usage of an RTSP ALG does not increase in
   itself increase the risk for session hijacking.  However the
   deployment of ALGs as the sole mechanism for RTSP NAT traversal will
   prevent deployment of encrypted end-to-end encrypted RTSP signaling.  The usage
   of TCP tunneling has no known security problems.  However  However, it might
   provide a bottleneck when it comes to end-to-end RTSP signaling
   security if TCP tunneling is used on an interleaved RTSP signaling
   connection.  The usage of TURN has severe risk of denial of service
   attacks against a client.  The TURN server can also be used as a
   redirect point in a DDOS DDoS attack unless the server has strict enough
   rules for who may create bindings.

9.  Acknowledgements

   The author would also like to thank all persons on the MMUSIC working
   group's mailing list that has commented on this document.  Persons
   having contributed in such way in no special order to this protocol
   are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon,
   Amir Wolf, Anders Klemets, and Colin Perkins.  Thomas Zeng would also
   like to give special thanks to Greg Sherwood of PacketVideo for his
   input into this memo.

   Section Section 1.1 contains text originally written for RFC 4787 by Francois
   Audet and Cullen Jennings.

10.  Informative References

   [I-D.ietf-avt-rtp-no-op]
              Andreasen, F., "A No-Op Payload Format for RTP",
              draft-ietf-avt-rtp-no-op-04 (work in progress), May 2007.

   [I-D.ietf-mmusic-rfc2326bis]
              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-29 draft-ietf-mmusic-rfc2326bis-30 (work in
              progress), March July 2012.

   [I-D.ietf-mmusic-rtsp-nat]
              Goldberg, J., Westerlund, M., and T. Zeng, "A Network
              Address Translator (NAT) Traversal mechanism for media
              controlled by Real-Time Streaming Protocol (RTSP)",
              draft-ietf-mmusic-rtsp-nat-12
              draft-ietf-mmusic-rtsp-nat-14 (work in progress),
              May
              November 2012.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, September 1981.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2588]  Finlayson, R., "IP Multicast and Firewalls", RFC 2588,
              May 1999.

   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
              Translator (NAT) Terminology and Considerations",
              RFC 2663, August 1999.

   [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
              Address Translator (Traditional NAT)", RFC 3022,
              January 2001.

   [RFC3424]  Daigle, L. and IAB, "IAB Considerations for UNilateral
              Self-Address Fixing (UNSAF) Across Network Address
              Translation", RFC 3424, November 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5382]  Guha, S., Biswas, K., Ford, B., Sivakumar, S., and P.
              Srisuresh, "NAT Behavioral Requirements for TCP", BCP 142,
              RFC 5382, October 2008.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations",
              RFC 6062, November 2010.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263, June 2011.

   [STUN-IMPL]
              "Open Source STUN Server and Client, http://
              www.vovida.org/applications/downloads/stun/index.html",
              June 2007.

Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   Stockholm,   SE-164 80
   Sweden

   Phone: +46 8 719 0000
   Fax:
   Email: magnus.westerlund@ericsson.com
   URI:

   Thomas Zeng

   Phone:
   Fax:
   Email: thomas.zeng@gmail.com
   URI: