Network Working Group                                  Magnus Westerlund
INTERNET-DRAFT                                                  Ericsson
Category: Standards Track                                    Thomas Zeng
Expires: December 2003                                       PacketVideo
                                                           June 30, 2003

     How to Enable Real-Time Streaming Protocol (RTSP) traverse Network
           Address Translators (NAT) and interact with Firewalls.

Status of this memo

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

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

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

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   This document is an individual submission to the IETF. Comments
   should be directed to the authors.

Copyright Notice

   Copyright (C) The Internet Society (2003). All Rights Reserved.


   This document describes six different types of NAT traversal
   techniques that can be used by RTSP. For each technique a description
   on how it shall be used, what security implications it has and other
   deployment considerations are given. Further a description on how
   RTSP relates to firewalls is given.

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1. Definitions.........................................................3
   1.1. Glossary.......................................................3
   1.2. Terminology....................................................3
2. Changes.............................................................3
3. Introduction........................................................4
   3.1. NATs...........................................................5
   3.2. Firewalls......................................................6
4. Detecting the loss of NAT mappings..................................6
5. NAT Traversal Techniques............................................7
   5.1. STUN...........................................................7
      5.1.1. Introduction..............................................7
      5.1.2. Using STUN to traverse NAT without server modifications...8
      5.1.3. Embedding STUN in RTSP...................................10
      5.1.4. Discussion On Co-located STUN Server.....................13
      5.1.5. ALG considerations.......................................13
      5.1.6. Deployment Considerations................................13
      5.1.7. Security Considerations..................................15
   5.2. ICE...........................................................16
      5.2.1. Introduction.............................................16
      5.2.2. Using ICE in RTSP........................................17
      5.2.3. Required Protocol Extensions.............................18
      5.2.4. Implementation burden of ICE.............................18
      5.2.5. Deployment Considerations................................18
      5.2.6. Security Considerations..................................19
   5.3. Symmetric RTP.................................................20
      5.3.1. Introduction.............................................20
      5.3.2. Necessary RTSP extensions................................20
      5.3.3. Using Symmetric RTP in RTSP..............................21
      5.3.4. Open Issues..............................................23
      5.3.5. Deployment Considerations................................24
      5.3.6. Security Consideration...................................24
   5.4. Application Level Gateways....................................25
      5.4.1. Introduction.............................................25
      5.4.2. Guidelines On Writing ALGs for RTSP......................26
      5.4.3. Deployment Considerations................................27
      5.4.4. Security Considerations..................................27
   5.5. TCP Tunneling.................................................27
      5.5.1. Introduction.............................................27
      5.5.2. Usage of TCP tunneling in RTSP...........................28
      5.5.3. Deployment Considerations................................28
      5.5.4. Security Considerations..................................28
   5.6. TURN (Traversal Using Relay NAT)..............................29
      5.6.1. Introduction.............................................29
      5.6.2. Usage of TURN with RTSP..................................29
      5.6.3. Deployment Considerations................................30
      5.6.4. Security Considerations..................................31
6. Firewalls..........................................................31

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7. Open Issues........................................................32
8. Security Consideration.............................................32
9. IANA Consideration.................................................33
10. Acknowledgments...................................................33
11. Author's Addresses................................................33
12. References........................................................35
   12.1. Normative references.........................................35
   12.2. Informative References.......................................35
13. IPR Notice........................................................36
14. Copyright Notice..................................................36

1. Definitions

1.1. Glossary

   ALG    – Application Level Gateway, an entity that can be embedded in
            a NAT to perform the application layer functions required
            for a particular protocol to traverse the NAT [6]
   ICE    - Interactive Connectivity Establishment, see [9].
   DNS    – Domain Name Service
   MID    - Media Identifier from Grouping of media lines in SDP, see
   NAT    - Network Address Translator, see [12].
   NAT-PT - Network Address Translator Protocol Translator, see [13]
   RTP    - Real-time Transport Protocol, see [5].
   RTSP   - Real-Time Streaming Protocol, see [1] and [7].
   SDP    - Session Description Protocol, see [2].
   SSRC   - Synchronization source in RTP, see [5].

1.2. Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [4].

2. Changes

   The following changes has been done since draft-ietf-mmusic-rtsp-nat-

   - New co-author Thomas Zeng.
   - Added a chapter on the usage of ICE in RTSP.
   - Added a definition for how to use STUN embedded to traverse
     symmetric NATs.
   - Added chapter on detecting loss of NAT mappings.
   - More text on Firewalls.
   - Symmetric RTP description has been extended with use case with a
     few well-known ports on the server side.

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   - Added text on transition strategies for the methods.
   - Improved language in the whole draft.
   - An Open Issues section has been created.

3. Introduction

   Today there is a proliferative deployment of different flavors of
   Network Address Translator (NAT) boxes that in practice follow no
   open standards [12][18].  NATs cause discontinuity in address realms
   [18], therefore a protocol, such as RTSP, needs to try to make sure
   that it can deal with such discontinuities caused by NATs. The
   problem with RTSP is that, being a media control protocol that
   manages one or more media streams, it carries information about
   network addresses and ports inside itself. Because of this, even if
   RTSP itself, when carried over TCP for example, is not blocked by
   NATs, its media streams are often blocked by NATs when RTSP based
   streaming servers are deployed as is and without special provisions
   to support NAT traversal.

   Like NATs, firewalls (FWs) are also middle boxes that need to be
   considered. They are deployed to prevent non-desired
   traffic/protocols to be able to get 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 problems. However there is a need for more detailed
   information on how FWs should be configured to work with RTSP.

   This document describes the usage of known NAT traversal mechanisms
   that can be used with RTSP. Following the guidelines spelled out in
   [18], we describe the required RTSP protocol extensions for each
   method, transition strategies, and we also discuss each method’s
   security concerns.

   Some of the NAT/FW traversal solutions are based on IETF internet
   drafts in their early stage of standardization (e.g., ICE and TURN).
   Given the current demand for NAT traversal solutions in the RTSP
   market place, it is foreseeable that a standard be created or
   adopted, in a timely fashion, by IETF MMUSIC WG to solve NAT
   traversal problem specifically for RTSP based streaming systems.

   This document is not based on RFC 2326 [1]. It is instead based and
   dependent on the updated RTSP specification [7], which is under
   development in IETF MMUSIC WG. The updated specification is a much-
   needed attempt to correct a number of shortcomings of RFC 2326. One
   important change is that the specification is split into several
   parts. So far only the updated core specification of RTSP is
   available in [7]. This document is one extension document to this
   core spec to document a special functionality that extends the RTSP
   protocol. This document is intended to be updated to stay consistent
   with the core protocol.

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3.1. NATs

   Today there exist a number of different NAT types and usage areas.
   The different NAT types are cited here from STUN [6]:

   Full Cone: A full cone NAT is one where all requests from the same
   internal IP address and port are mapped to the same external IP
   address and port. Furthermore, any external host can send a packet to
   the internal host, by sending a packet to the mapped external

   Restricted Cone: A restricted cone NAT is one where all requests from
   the same internal IP address and port are mapped to the same external
   IP address and port. Unlike a full cone NAT, an external host (with
   IP address X) can send a packet to the internal host only if the
   internal host had previously sent a packet to IP address X.

   Port Restricted Cone: A port restricted cone NAT is like a restricted
   cone NAT, but the restriction includes port numbers. Specifically, an
   external host can send a packet, with source IP address X and source
   port P, to the internal host only if the internal host had previously
   sent a packet to IP address X and port P.

   Symmetric: A symmetric NAT is one where all requests from the same
   internal IP address and port, to a specific destination IP address
   and port, are mapped to the same external IP address and port. If the
   same host sends a packet with the same source address and port, but
   to a different destination, a different mapping is used. Furthermore,
   only the external host that receives a packet can send a UDP packet
   back to the internal host.

   NATs are used on both small and large scale. The normal small-scale
   user is home user that has a NAT to allow multiple computers share
   the single IP address given by their Internet Service Provider (ISP).
   The large scale users are the ISP's themselves that give there users
   private addresses. This is done both for control and for lack of IP

   Native Address Translation and Protocol Translation (NAT-PT) [13] is
   a mechanism used for IPv4 to IPv6 transition. This device is used to
   allow devices only having connectivity using one of the IP versions
   to communicate with the other address domain. If the other address
   domain is addressable through the use of domain names, then a DNS ALG
   assigns temporary IP addresses in the requestor's domain. The NAT-PT
   device translates this temporary address to the receivers true IP
   address and at the same time modify all necessary fields to be
   correct in the receiver's address domain.

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3.2. 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 pulic domain (public internet).  Many
   organizations use firewalls to prevent privacy intrusions and
   malicious attacks to corporate computing resources in the private
   intranet [19].
   A comparison between NAT and FW are given below:

   1. FW must be a gateway between two network administrative domains,
      while NAT does not have to sit between two domains. In fact, in
      many corporations there are many NAT boxes within the intranet, in
      which case the NAT boxes sit between subnets.
   2. NAT does not in itself provide security, although some access
      control policies can be implemented using address translation
   3. NAT and FWs are similar in that they can both be configured to
      allow multiple network hosts to share a single public IP address.
      In other words, a host behind a NAT or FW can have a private IP
      address and a public one, so for NAT and FW there is the issue of
      address mapping which is important in order for RTSP protocol to
      work properly when there are NATs and FWs between the RTSP server
      and its clients.

4. Detecting the loss of NAT mappings

   Several of the described NAT traversal techniques in the next chapter
   use the fact that the NAT UDP mapping's external address and port can
   be discovered. This information is then utilized to direct the
   traffic intended for the local side's address to the external
   instead. However any such information is only valid while the
   mapping is intact. As the IAB's UNSAF document [18] points out the
   mapping can either timeout or change its properties. It is therefore
   important for the NAT/FW traversal solutions to handle the loss or
   change of NAT mappings, according to UNSAF.

   First, it is important to ensure that there exists the possibility to
   send keep-alive traffic to minimize the probability of timeout. The
   difficulty is that the timeout timer can have varying length between
   different NATs. That is the reason why that UNSAF recommends usage of
   STUN to determine this timeout.

   Secondly, it is possible to detect and recover from the situation
   where the mapping has been changed or removed. The possibility to
   detect a lost mapping is based on the fact that no traffic will
   arrive. Below we will give some recommendation on how to detect loss
   of NAT mappings when using RTP/RTCP under RTSP control.

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   For RTP session there is normally a need to have both RTP and RTCP
   functioning. The loss of a RTP mapping can only be detected when
   expected traffic does not arrive. If no data arrives after having
   issued a PLAY request and received the 200 response, one can normally
   expect to receive RTP packets within a few seconds. However, for a
   receiver to be 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 very likely the RTP mapping has been

   The loss of mapping carrying RTCP is simpler to detect. As 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 no RTCP packets are
   received by the RTSP server for a while, the RTSP server has the
   option to delete the corresponding SSRC and RTSP session ID, which
   means either the client could not get through a middle box NAT/FW, or
   that the client is mal-functioning.

5. NAT Traversal Techniques

   There exist a number of NAT traversal techniques that can be used to
   allow RTSP to traverse NATs. However they have different features,
   they are applicable to different topologies; and the cost is also
   different. They also differ in their security considerations. In the
   following sections, each technique is outlined in details in terms of
   its advantages and disadvantages.

   Not all of the techniques are yet described in the full details
   needed to actually use this document as a specification for how to
   use them. These sections are included to present comparison amongst
   the different methods in order for one to identify the most suitable
   method for a particular RTSP deployment scenario. There are methods
   that use protocols in early stage of standardization, such as TURN
   and ICE.

5.1. STUN

5.1.1. Introduction

   STUN – Simple Traversal of UDP Through Network Address Translators
   [6] is a standardized protocol developed by the MIDCOM WG that allows
   a client to use secure means to discover the presence of a NAT
   between himself and the STUN server and the type of that NAT. The
   client then uses the STUN server to discover the address bindings
   assigned by the NAT. The protocol also allows discovery of the
   mappings timeout period and can be used in any keep-alive mechanism.

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   STUN is a client-server protocol. STUN client sends a request to a
   STUN server and the server returns a response. There are two types of
   STUN requests – Binding Requests, sent over UDP, and Shared Secret
   Requests, sent over TLS over TCP. We note here that for RTSP clients
   running on embedded devices, it may not be practical to require TLS
   be implemented on the embedded device (such as a cell phone).
   Therefore in the next section we propose to adapt RFC 3489 ([6]) so
   as to let RTSP use a subset of STUN packets/features for NAT
   traversal, but without requiring full implementation of STUN in an
   RTSP server or RTSP client.  We note that RFC 3489 has provisions for
   STUN to be embedded in another application (see section 6 of [6]).

5.1.2. 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. However 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).


   - The server must be located in either a public address realm or the
     next hop external address realm in regards to the client.
   - The client may only be located behind NATs that are of the full
     cone, address restricted, or port restricted type. Clients behind
     symmetric NATs cannot use this method.


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

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

   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. For example, each RTP
      session will in general require two UDP ports, one for RTP, and
      one for RTCP.

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   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. If successful a mapping has been established:
      clients local address/port <-> public address/port.

   4. Perform the RTSP SETUP for each media. In the transport header the
      following parameter SHOULD be included with the given values:
      "dest_addr" with the public/external IP address and port pair for
      both RTP and RTCP. To allow this to work servers MUST allow a
      client to setup the RTP stream on any port, not only even ports.
      The server SHOULD respond with a transport header containing an
      "src_addr" parameter with the RTP and RTCP source IP address and
      port of the media stream.

   5. To keep the mappings alive, the client SHOULD periodically send
      UDP traffic over all mappings needed for the session. STUN MAY be
      used to determine the timeout period of the NAT(s) UDP mappings.
      For the mapping carrying RTCP traffic the periodic RTCP traffic
      may be enough. For mappings carrying RTP traffic and for mappings
      carrying RTCP packets not frequent enough, keep alive messages
      SHOULD be sent. As keep alive messages, empty IP/UDP messages
      SHOULD be sent to the streaming servers discard port (port number

   If a UDP mapping is lost then the above discovery process is required
   to be performed again. The media stream needs to be SETUP again to
   change the transport parameters to the new ones. This will likely
   cause a glitch in media playback.

   To allow UDP packets to arrive from the server to a client behind a
   restricted NAT, some UDP packets must first be sent to the server.
   The client, before sending a RTSP PLAY request, must send an empty or
   small UDP message, on each mapping, to the IP address given as the
   servers source address. To create minimum problems for the server
   these UDP packets SHOULD be sent to the server's discard port (port
   number 9) and contain no or very little data. To ensure that at least
   one UDP message passes the NAT, several messages are recommended to
   be sent.

   For a port restricted 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, UDP packets needs to be sent first to the servers source
   address/port. To minimize potential effects on the server from these
   messages the following type of messages MUST be sent. RTP: An empty
   or less than 12 bytes large UDP message. RTCP: A correctly formed
   RTCP message.

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   The above described adaptations for restricted NATs will not work
   unless the server includes the "src_addr" "Transport" header

5.1.3. Embedding STUN in RTSP

   This section describes the adaptation and embedding of STUN within
   RTSP. This enables STUN to be used to traverse any type of NAT,
   including symmetric NATs. This adaptation is an extension to the core
   RTSP protocol [7], and therefore is signaled by feature tag. As
   specified in [7], features are recommended to be negotiated using
   "supported" headers.

   We define the feature tag for embedded STUN with out authentication
   support as:


   and for embedded STUN supporting authentication as:


   If one side supports "nat.stun-auth" but the other side only supports
   "nat.stun", then both sides must go through negotiation and possibly
   downgrade to using "nat.stun". If one RTSP end system refuses to
   accept "nat.stun", then do not use STUN for RTSP.


   This NAT traversal solution (using STUN with RTSP) has limitations:

      1. It does not work if both RTSP client and RTSP server are behind
         separate NATs.
      2. In the case of "nat.stun", the RTSP server may, for security
         reasons, refuse to send media streams to an IP different from
         the IP in the client RTSP requests. Therefore, if the client is
         behind a NAT that has multiple public addresses, and the
         client’s RTSP port and UDP port are mapped to different IP
         addresses, RTSP SETUP will fail.

   Deviations from STUN as defined in RFC2389

   Specifically, we differ from RFC3489 in two aspects:
      1. We allow RTSP applications to have the option to perform
         "binding discovery" without authentication;
      2. We require STUN server be co-located on RTSP server’s media

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   In order to allow binding discovery without authentication, the STUN
   server embedded in RTSP application would ignore authentication tag,
   and the STUN client embedded in RTSP application would use dummy
   authentication tag, as well.

   In order to use STUN to solve NAT traversal when RTSP client is
   behind a symmetric NAT, STUN server must co-locate on RTSP server’s
   media ports. This can be done, for instance, by embedding STUN server
   in RTSP server.

   In fact, if STUN server is indeed co-located with RTSP server’s media
   port, then a RTSP client using RTP transport over UDP can use STUN to
   traverse ALL types of  NATs that have been defined in section 3.1. In
   the case of symmetric NAT, the party inside the NAT must initiate UDP
   traffic. The STUN Bind 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

   Likewise, if a RTSP server is behind a NAT, then an embedded STUN
   server must co-locate on the RTSP client’s RTCP port. In this case,
   we assume that the client has some means to establish TCP connection
   to the RTSP server behind NAT so as to exchange RTSP messages with
   the RTSP server.

   RTSP implementations supporting such features must use the feature
   tag, (nat.stun-auth or nat.stun) to indicate to each other the
   availability of such embedded, co-located STUN servers.

   To minimize delay, we require that the RTSP server supporting this
   option must inform its client the RTP and RTCP ports that the server
   intend to send RTP and RTCP packets, respectively.

   To minimize the keep-alive traffic for address mapping, we also
   require that the RTSP end-point (server or client) sends and receives
   RTCP packets from the same port.

RTSP NAT Traversal Algorithm Using STUN

   The actual NAT traversal algorithm contains six steps.

     Step 1:
            This first step is for both RTSP server and client to
          discover whether there is a NAT, and if yes, the timeout
          period for UDP mapping on the NAT. For the RTSP client, as
          soon as it has learnt that the RTSP server supports the
          "nat.stun" or "nat.stun-auth" feature, and that it has learnt
          the RTSP server’s RTP and RTCP ports, it should send STUN
          request packets to those ports, and also include the

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          appropriate feature tag (either nat.stun or nat.stun-auth) in
          all of its relevant RTSP requests and responses.

          On the other hand, a RTSP server can figure out whether it is
          in the public Internet at start up time. If it turns out that
          the RTSP server is in a private address realm, the RTSP server
          must be prepared to receive STUN Binding Request on its RTCP
          receive port (so as to help RTSP client’s RTCP RR reports to
          reach the right destination). Otherwise, if it turns out that
          RTSP server is in the public address realm, it must be
          prepared to do the following:
          - If "nat.stun" is the agreed-upon feature tag between server
            and client, the RTSP server must monitor its RTP and RTCP
            send ports for STUN Binding Requests;
          - If "nat.stun-auth" is the agreed-upon feature tag between
            server and client, the RTSP server must monitor its RTP and
            RTCP send ports for STUN Shared Secrete Requests and Binding

   Step 2:
            The RTSP client determines the number of UDP ports needed by
          counting the number of RTP sessions in the multi-media
          presentation. This information is available in the media
          description protocol, e.g. SDP [2], and according to the
          client’s media selection criteria. In general each RTP session
          will require two UDP ports, one for RTP, and one for RTCP. The
          RTSP client also obtains, for each RTP session, the media port
          from which RTSP server will send out the RTP packets.

   Step 3:
            This step applies if the client knows, from step 1, that it
          is behind NAT. For each UDP port required, the RTSP client
          must open a local socket using an available UDP port on the
          host computer, establish a mapping and discover the public IP
          address and port number with the help of the STUN server co-
          located at the RTSP server’s media ports.  Assume STUN
          protocol exchange is successful, an address mapping will be
          sent back to the RTSP client in a STUN response packet, then
          the RTSP client must record the mapping between client’s local
          address/port and the external address/port in its database.

   Step 4:
            RTSP client then performs the RTSP SETUP for each media. In
          the transport header the following parameter SHOULD be
          included with the given values: "dest_addr" with the external
          IP address and port pair for both RTP and RTCP. To allow this
          to work servers MUST allow a client to setup the RTP stream on
          any port, not only even ports.

   Step 5:

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            This step only applies if client is in the open, but RTSP
          server discovers, with the help of a public STUN server, that
          it is the one behind NAT. RTSP server obtains the client’s
          RTCP port number from the SETUP request, and immediately sends
          STUN request to that port to obtain the address mapping.
          Assume again the mapping is obtained successfully, then the
          server SHALL respond with a transport header containing a
          "src_addr" parameter with the mapped RTCP source IP address
          and port.

   Step 6:
            To keep the mappings alive, the party that is behind NAT
          SHOULD periodically send UDP traffic over all mappings needed
          for the session when no traffic is received. For the mapping
          carrying RTCP traffic the periodic RTCP traffic may be enough.
          For mappings carrying RTP traffic and for mappings carrying
          RTCP traffic infrequently, keep alive messages SHOULD be sent.
          STUN packets can serve as keep alive messages, given the
          requirement to have STUN server collocates on the RTSP
          server’s media ports.

   If a UDP mapping is lost then the above discovery process is required
   to be performed again. The media stream needs to be SETUP again to
   change the transport parameters to the new ones. This will likely
   cause a glitch in media playback.

5.1.4. Discussion On Co-located STUN Server

   In order to use STUN to traverse symmetric NATs the STUN server needs
   to be co-located with the streaming server media \ports, i.e., the
   port from which RTP packets will be sent. 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. This
   works fine between STUN and RTP or RTCP where the first byte has
   always present difference, but this can't be guaranteed to work with
   other media protocols.

5.1.5. ALG considerations

   If a NAT supports RTSP ALG (Application Level Gateway) and is not
   aware of the STUN traversal option, service failure may happen,
   because a client discovers its public IP address and port numbers,
   and inserts them in its SETUP requests, when the RTSP ALG processes
   the SETUP request it may change the destination and port number,
   resulting in unpredictable behavior.

5.1.6. Deployment Considerations

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   For the non-embedded usage of STUN the following applies:


   - Using STUN does not require RTSP server modifications, it only
     affects the client implementation.


   - Requires a STUN server deployed in the public address space.
   - Only works with Cone NATs. Restricted Cone NATs create some
     issues. Does not work with Symmetric NATs without server
   - Will mostly not work if a NAT uses multiple IP addresses, since
     RTSP server generally requires all media streams to use the same IP
     as used in the RTSP connection (for more on this subject, see next
     section, security considerations).
   - Interaction problems exist when a RTSP ALG is not aware of STUN.
   - Using STUN requires that RTSP servers and clients support the
     updated RTSP specification.


   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 however most probably wait until no
   need at all exists to use STUN.

   For the Embedded STUN method the following applies:


   - 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
   - STUN has built-in message authentication features, which makes it
     more secure. See next section for an in-depth security discussion.
   - This solution works as long as there is only one RTSP end point in
     the private address realm, regardless of the NAT’s type. There may
     even be multiple NATs (see figure 1 in [6]).
   - Compares 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 have become
     available [21]. There is the need to embed STUN in RTSP server and
     client, which require a de-multiplexer between STUN packets and

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     RTP/RTCP packets. There is also a need to register the proper
     feature tags.


   - Feature tags must be registered with IANA.
   - Requires an embedded STUN server to co-locate on each of 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, the later to STUN server.
   - It does not work if none of the RTSP server and client is in the
     public address realm, and each of them is behind a different NAT.
   - Even if the RTSP server is in the open, and the client is behind a
     multi-addressed NAT, it may still break if the RTSP server does not
     allow RTP packets to be sent to an IP differs from the IP of the
     client’s RTSP request.
   - Interaction problems exist when a RTSP ALG is not aware of STUN.
   - Using STUN requires that RTSP servers and clients support the
     updated RTSP specification, and they both agree to support the
     proper feature tag.


   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 however most probably wait until
   there is no need at all to use STUN. When there is no more need to
   use STUN, the feature tags, "nat.stun" and "nat.stun-auth", can be
   de-registered at IANA.

5.1.7. 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 UDP traffic). Such restriction means
   that STUN does not work for NATs that would assign different IP
   addresses to different UDP flows on its public side. Therefore most
   multi-addressed NATs will not work with STUN.

   In terms of security property, STUN combined with destination address
   restricted RTSP has the same security properties as the core RTSP. It

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   is protected from being used as a DoS attack tool unless the attacker
   has ability to hijack RTSP stream.

   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 can be the initiator of an attack,
   these mechanisms can't be used to protect servers against being DoS
   attack tools.

5.2. ICE

5.2.1. Introduction

   ICE (Interactive Connectivity Establishment) [9] is a methodology for
   NAT traversal that is under development for SIP. The basic idea is to
   try, in a parallel fashion, all possible connection addresses that an
   end point may have. This allows the end-point 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. End point A collects all possible address that
   can be used, including local IP addresses, STUN derived addresses,
   TURN addresses. On each local port that any of these address and port
   pairs leads to, a STUN server is installed. This STUN server only
   accepts STUN requests using the correct authentication through the
   use of username and password.

   End-point A then sends a request to establish connectivity with end-
   point B, which includes all possible ways to get the media through to
   A. Note that each of A’s published address/port pairs has a STUN
   server co-located. B, before responding to A, uses a STUN client to
   try to reach all the address and port pairs specified by A. The
   destinations for which the STUN requests have successfully completed
   are then indicated. If bi-directional communication is intended the
   end-point B must then in its turn offer A all its reachable address
   and port pairs, which then are tested by A.

   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. The STUN requests may also
   result in that more connectivity alternatives are discovered and
   conveyed in the STUN responses.

   This chapter is not yet a full technical solution. It is mostly a
   feasibility study on how ICE could be applied to RTSP and what
   properties it would have. One nice thing about ICE for RTSP is that
   it does make it possible to deploy RTSP server behind NAT/FIRWALL, a
   desirable option to some RTSP applications.

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5.2.2. 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 step-by-step algorithm
   could be used to accomplish connectivity for the UDP traffic.

   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 opened, or the deployment of proxies,

   1. The client retrieves the SDP from the ICE enabled RTSP server.
   This SDP contains indication that the RTSP server supports ICE and
   gives the address/ports for each media and its necessary UDP streams.
   This may require a SDP extension or possibly the "c=" lines in which
   port numbers can be used. This will however require the server to
   send media streams from well-known ports. This will result in that
   many sessions will go over the same ports for servers handling
   multiple users.

   2. The client analyzes the SDP and determines the number of local UDP
   ports it will need. For each port it also installs a STUN server with
   authentication requirement using an authentication tag. From these
   ports the client then tries a STUN request to the server's announced
   ports, which are intercepted by the co-located STUN servers. If
   successful, the client’s NAT bindings, as seen by the RTSP server,
   are discovered by these STUN servers and sent back to the RTSP
   client. Also, other addresses, including addresses from public STUN
   servers and TURN addresses, can be collected by the RTSP client.

   3. Client creates a SETUP request where he includes a number of
   transport header specifications. The client may offer more than one
   transport configurations, but for each configuration it will need to
   create multiple specifications of destination addresses that it has
   learned in descending priority order. The client also includes in the
   transport specification the ICE indicator carrying the user name and
   password required by the client's STUN servers.

   4. The server receives the SETUP request and selects which transport
   specification it would like to accept. Here all specifications are
   the same except for destination address/port. For all specifications
   in the configuration the server tries to "STUN" these
   addresses/ports. Depending on the answer, the following results may
    A. The RTSP server successfully connects to the client’s STUN
       server, and the RTSP server selects the specification with
       highest priority that yields a successful response and include

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       that address/port in the SETUP response's transport headers
       destination field. The media is ready to be played.

    B. The server fails to reach any of the client’s STUN servers. It
       uses a new error code to inform the client of this. At the same
       time it includes an updated SDP, which contains all addresses
       that it is reachable on. The server might have learned some new
       reachable addresses since the initial SDP. The client then tries
       again by going to step 2 above and repeat the entire process. If
       it fails multiple times the server and client eventually give up.

5.2.3. Required Protocol Extensions

    1. A SDP extension to indicates that the server supports ICE. It
       will also require that grouping of media lines [10] with the
       alternative semantics [11] be used in the SDP to indicate the
       different alternatives.

    2. A new Transport header parameter that indicates that ICE shall be
       used on these streams and a way to convey the authentication user
       name and password that the server shall use to contact the
       client’s STUN server.

    3. A RTSP error code for failed ICE setup. That error code will also
       need to include entity body in the response to carry the updated
       SDP description.

5.2.4. 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 traversal methods in this document.

   A RTSP server implementation requirements:
    - Full STUN server features
    - limited STUN client features
    - SDP extensions that includes MID [10] and ICE features
    - Dynamic SDP generation with more parameters.
    - RTSP error code for ICE extension

    - Limited STUN server features
    - Limited STUN client features
    - SDP extensions that include MID [10] and ICE features
    - RTSP error code and ICE extension

5.2.5. Deployment Considerations

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   - Solves NAT connectivity discovery for basically all cases as long
     as a TCP connection between them can be established in the first
     hand. This includes servers behind NATs. (Note that a proxy between
     address domains may be required to get TCP through).
   - Prevents DOS attacks as media receiving client is required to do
     STUN responses with authentications on its media reception ports,
     see 5.2.6.


   - Increases the setup delay with at least the amount of time it
     takes the server to perform its STUN requests.
   - Forces servers to use a few well-known media ports.
   - Assumes that it is possible to de-multiplex between media packets
     and STUN packets.
   - Has high implementation burden for both server and client. Given
     the complexity of ICE, it is foreseeable that practitioners may opt
     to use TCP tunneling to deploy RTSP based services. Note that TCP
     tunneling can result in loss of real-time properties for the media
   - ICE is not a standard yet. It is only an initial proposal in the
     SIPPING working group.
   - ICE has the same consideration regarding ALGs as STUN, see section


   The use of ICE enables a client to phase-out not needed methods of
   creating NAT bindings. However the usage of ICE to ensure that media
   delivery is not done to unwanted receiver is not intended to be
   removed as it strengthens security.

5.2.6. Security Considerations

   ICE has the advantage that it prevents RTSP servers from being used
   as DoS tools. The protection is achieved due to the STUN request sent
   from the server to the client. A client requesting media gives the
   destination address and port for the server to deliver the media too.
   The server tries these port using STUN requests. If the client does
   not have prior knowledge about the media stream no STUN server are
   present. The usage of user name and password ensures that only the
   server that the client has requested to deliver media can issue valid
   STUN request.

   This solution is only vulnerable to a man in the middle attack, where
   the attacker can redirect and answer the STUN request before it
   reaches the targeted host. If one utilizes a secure channel for the

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   RTSP messages a potential attacker can't eavesdrop the RTSP messages
   carrying the STUN username and password. However an eavesdropper of
   the STUN request can still learn them. There exist possibility to
   also fend off such attacks, by using HMAC [20] in the STUN request
   and send the shared secret in the protected RTSP messages. However
   this risk is considered small and the client can also refuse to
   answer STUN requests if these requests arrive undesirably frequent,
   which may be a sign that someone is trying to break the hash
   algorithm in the HMAC code.

   The simplest usage scenario of ICE will result in that the RTSP
   server utilize a few well known ports for sending media and having
   its STUN server available on. The solution does not force this usage
   onto the server, as sender ports can be created dynamically at the
   time of RTSP DESCRIBE request. However the amount of resources needed
   to maintain this usage will be significantly larger then for using a
   few well-known ports. The usage of well-known ports will simplify
   certain types of attacks on the server, like overload attacks using

5.3. Symmetric RTP

5.3.1. Introduction

   Symmetric RTP is a NAT traversal solution that is based on requiring
   NATed clients to send UDP packets to the server’s media send ports.
   In core RTSP, usage of RTP over UDP is uni-directional, where the
   server sends RTP packets to client’s RTP port. Symmetric RTP 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

   Specifically, when the RTSP server receives the "connect" RTP packet
   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 has the advantage of working for all types of NATs.
   However, it does require server modifications. Symmetric RTP is
   somewhat more vulnerable to hijacking attacks, which will be
   explained in more details in the section discussing security

5.3.2. Necessary RTSP extensions

   To support symmetric RTP the RTSP signaling must be extended to allow
   the RTSP client to indicate that it will use symmetric RTP. The
   client also needs to be able to signal its RTP SSRC to the server in

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   its SETUP request. The RTP SSRC is used to establish some basic level
   of security against hijacking attacks. Care must be taken in choosing
   client’s 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 may occur, the RTSP server must be able to signal to client
   that a collision has occurred and that it wants the client to use a
   different RTP SSRC carried in the SETUP response.

   A RTP specific "Transport" header parameter is defined to indicate
   that symmetric RTP shall be used for the media transport. The
   parameter is included in each "Transport" header specification where
   the client will use symmetric RTP. A Server SHALL NOT accept the
   transport specification unless it supports symmetric RTP. If the
   client has requested to use symmetric RTP for a session the server
   MUST include this parameter ("sym_rtp") in the response.

   The parameter is defined in ABNF [3] as:

     symm-usage = "sym_rtp" "=" "1"

   Which follows the definition in [7] for transport parameter

   It is also necessary to define a "Transport" header parameter,
   "client_ssrc", to carry the SSRC that the client will use. In RTP[5],
   SSRC parameter is only valid for uni-cast transmission. It identifies
   the synchronization source to be associated with the media stream,
   and is expressed as an eight-digit hexadecimal value. In cases where
   a client will use multiple SSRCs it SHOULD NOT use this parameter.

   The parameter is defined in ABNF [3] as:

     client_ssrc_def = "client_ssrc" "=" ssrc

   Where "ssrc" is defined in [7].

   Further, a RTSP options tag that can be used to indicate support of
   symmetric RTP according to this specification is defined below:


   This tag SHALL be included in the supported header by both clients
   and servers supporting symmetric RTP according to this specification.

5.3.3. Using Symmetric RTP in RTSP

   The server and client uses Symmetric RTP in the following way:

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   1. The client can optionally determine through the use of STUN that
       it is located behind a NAT. If the client uses STUN it should
       also determine the timeout of NAT it is behind.

   2. The client MAY investigate if the server supports symmetric RTP
       by including the supported header with the tag "nat.sym-rtp" in
       an OPTIONS or DESCRIBE request to the server. A server supporting
       symmetric RTP will include the tag in its response.

   3. The client determines that it will use symmetric RTP to traverse
       the NAT. This decision is based on the knowledge about the NAT
       type and the necessary support from the server. If there is no
       NAT the client SHOULD NOT use symmetric RTP due to the higher
       risk of session hijacking.

   4. The client sends a SETUP request which has the parameter
       "sym_rtp=1" in the transport line. It MUST also include the
       parameter "client_ssrc" in each transport specification
       containing "sym_rtp=1". The "client_ssrc" contains the random
       SSRC it is going to use for that RTP session, unless in SETUP
       response the server over-ride "client_ssrc", in which case the
       client must use the server assigned SSRC. The SSRC MUST be
       generated in a random way, as the randomness of the SSRC is the
       basic security mechanism that prevents hijacking. Symmetric RTP
       MUST only be requested for unicast transport.

   5. The server chooses the transport specification that it will use
       to transport the media. When this transport specification is the
       one declaring the use of symmetric RTP the server performs the
       - The server MUST include the transport parameters "sym_rtp=1",
         and "src_dest" in the response.
       - The server MUST both send and receive data on the indicated
       - The server SHALL ignore any of the transport header parameters
         "destination", and client_port.
       - If the server is using the same UDP send port to send media
         packets to multiple RTSP clients, it must also check for client
         RTP SSRC collisions. If in a SETUP request, the "client_ssrc"
         is already in use, the server must assign a different SSRC that
         is unique, and signal it in SETUP response.

   6. The Server starts listening on the declared server ports for
       RTP/UDP packets containing valid client SSRCs. Any received
       RTP/UDP packet not containing a valid client SSRC SHOULD be
       ignored. When a RTP/UDP packet containing valid client SSRC is
       received, the server looks up the id of the client media session
       using the unique client SSRC, stores the source IP and Port as
       being the destination address and port for that media session
       (i.e., RTP session). It performs the corresponding actions for

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       the RTCP port to establish the destination of the RTCP
       transmissions as well.

   7. The client establishes the address binding at the server by
       sending RTP or RTCP to the servers declared media address and
       port from the port it desires to receive RTP/RTCP on. For the RTP
       channel it sends a RTP/UDP packet containing the client SSRC. The
       RTP/UDP packet SHALL NOT contain any payload data and use payload
       type 0. To the servers RTCP port it sends a normal RTCP packet.

   8. Upon reception of a "binding packet" the server SHALL respond. On
       the RTP port it SHALL respond with a single RTP/UDP packet using
       payload type 0 and having a 0 byte payload. For each received
       client packet that contains the correct SSRC the server SHALL
       respond with a single packet. For RTCP the client starts
       transmitting RTCP packets according to the normal RTCP timing
       rules. The server SHALL also send RTCP as soon as it receives a
       RTCP packet creating the binding.

   9. To ensure that the clients binding packets are not lost the
       client SHOULD retransmit the binding RTP packet every 250 ms
       until it receives a response with an empty RTP packet or it has
       retransmitted 20 times. For RTCP it is enough to transmit RTCP
       packet according to the normal rules. However a client MAY send
       one RTCP packet every 500 ms until it receives an answer, or it
       has tried for 10 seconds.

   10. When the client has received answers for both RTP and RTCP it can
       safely progress the session and send a PLAY request.

   11. To ensure that the binding is not lost the client SHOULD send a
       empty RTP/UDP packet with PT=0 to the server every tenth of the
       mapping timeout time that has discovered for the NAT. The
       discovery can be performed by using STUN. The client SHOULD NOT
       send these packets as long as the server transmit RTP packets to
       the client. Unless the NAT mappings has very short timeouts or
       the RTCP bandwidth is severely restricted the RTCP traffic should
       automatically keep its connection open.

5.3.4. Open Issues

   The proposal for symmetric RTP contains some open issues that needs
   to be addressed.

   - Should it be allowed to change a binding once it has been
   established? Probably not as it would be security weakness, however
   this result in that RTSP SETUP must be used to update the server
   destination once a binding has been lost and restored.

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   - What RTP payload type(s) shall the client use. Should it use one of
   the types that the server has declared is going to use in the server
   -> client direction or a static one?

   - Should the security be improved by having a RTP challenge that can
   contain longer random identifiers? If that is the case it should have
   a static payload type as the client can't establish dynamic payload
   type declarations.

5.3.5. Deployment Considerations


   - Works for all types of NATs, including those using multiple IP
   - Have no interaction problems with any RTSP ALG changing the
     client's information in the transport header.


   - Requires Server support.
   - Has somewhat worse security situation then STUN when using address
   - Still requires STUN to discover the timeout of NAT bindings.


   The usage of symmetric RTP can be phased out as long as the client
   has a way of detecting that it does not need it any more. Possible
   ways of detecting this is the use of STUN as proposed in the optional
   first step. Another way is that it simply is replaced with something

5.3.6. Security Consideration

   Symmetric RTP's major security issue is that RTP streams can be
   hijacked and directed towards any target that the attacker desires.
   The method has also no protection if client desires to initiate media
   streams to a target it desires to do a DOS attack on.

   The most serious security problem is the deliberate attack with the
   use of a RTSP client and symmetric RTP. The attacker uses RTSP to
   setup a media session. Then it uses symmetric RTP 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 to be the same as the RTSP connection arrived on. This
   prevents symmetric RTP to be used with multi-address NATs.

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   The hijack attack can 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 RTP packets containing the correct RTP SSRC to the
   correct address and port on the server. The destination of the
   packets is set as the source IP and port in these RTP packets.

   Another variation of this attack is to modify the RTP binding packet
   being sent to the server by simply changing the source IP to the
   target one desires to attack.

   One can protect oneself against 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, which is required in
   order to protect against this type of DOS attack.

   The random SSRC tag in the binding packet determines how well
   symmetric RTP can fend off streaming hijacking performed by parties
   that are not "men-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-
   predictive randomizer. 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 implement functionality that blocks ports
   that receive multiple binding packets with different invalid SSRCs,
   especially when they are coming from the same IP/Port.

   To improve the security against attackers the random tags length
   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.

5.4. Application Level Gateways

5.4.1. 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:

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   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 ALG's this may also break the
   end-to-end functionality of RTSP.

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

5.4.2. Guidelines On Writing ALGs for RTSP

   In this section, we provide a step-by-step guideline on how one
   should 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,
      the ALG SHOULD NOT change the address. Ways to detect that these
      methods are used are:
      - For embedded STUN, watch for the feature tag "nat.stun". If any
      of those exists in the "supported", "proxy-require", or "require"
      headers of the RTSP exchange.
      - For non-embedded 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. However if multiple NATs are
      used this detection may fail.

      Otherwise continue to the next step.

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

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

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

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

5.4.3. Deployment Considerations


   - No impact on either client or server
   - Can work for any type of NATs


   - When deployed they are hard to update to reflect protocol
     modifications and extensions. If not updated they will break the
   - When end-to-end security is used the ALG functionality will fail.
   - Can interfere with other type of traversal mechanisms, such as


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

5.4.4. Security Considerations

   An ALG will not work when deployment of end-to-end RTSP signaling
   security. Therefore deployment of ALG will result in that end-to-end
   security will not be used by clients located behind NATs.

5.5. TCP Tunneling

5.5.1. 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

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   transported over UDP requires the TCP connection to support some type
   of framing of the RTP packets.

   Using TCP also results in that the client has to accept that real-
   time performance may no longer be possible. TCP's problem of ensuring
   timely deliver was the reasons why RTP was developed. Problems that
   arise with TCP are: head-of-line blocking, delay introduced by
   retransmissions, highly varying congestion control.

5.5.2. Usage of TCP tunneling in RTSP

   The RTSP core specification [7] supports interleaving of media data
   on the TCP connection that carries RTSP signaling. See section 10.13
   in [7] for how to perform this type of TCP tunneling.

   There is currently new work on one more way of transporting RTP over
   TCP in AVT and MMUSIC. For signaling and rules on how to establish
   the TCP connection in lieu of UDP, see [16]. Another draft describes
   how to frame RTP over the TCP connection is described in [17].

5.5.3. Deployment Considerations


   - Works through all types of NATs.


   - Functionality needs to be implemented on both server and client.
   - May not give real-time performance.


   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 price of loss of
   real-time properties.

5.5.4. Security Considerations

   The TCP tunneling of RTP has no known security problem besides those
   already present in RTSP. 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 would be the performance bottleneck
   when RTSP encryption is applied, since all session media data also
   needs to be encrypted.

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5.6. TURN (Traversal Using Relay NAT)

5.6.1. Introduction

   Traversal Using Relay NAT (TURN) [8] is a protocol for setting up
   traffic relays that allows 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 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 that RTSP server may restrict that destinations other than
   the IP address that the RTSP message arrives from shall not be
   accepted. 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. Unfortunately TURN servers
   can be targeted at any host that has a public IP address by spoofing
   the source IP of TURN Allocation requests.

5.6.2. Usage of TURN with RTSP

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

   1. The RTSP client connects with RTSP server. The client retrieves
      the session description to determine the number of media streams.

   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

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      allowed to send media traffic to the given address. The server
      SHOULD include its RTP SSRC in the SETUP response.

   4. Client requests that the Server starts playing. The server starts
      sending media packet 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 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

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

5.6.3. Deployment Considerations


   - Does not require any server modifications.
   - Works for any types of NAT as long as the server has public
     reachable IP address.


   - TURN is not yet a standard.
   - Requires another network element, namely the TURN server.
   - Such a TURN server for RTSP is not scalable since the number of
     sessions it must forward is proportional to the number of client
     media sessions.
   - TURN server becomes a single point of failure.
   - Since TURN forwards media packets, it necessarily introduces
   - Requires that the server can verify that the given destination
     address is valid to be used by the client.
   - An RTSP ALG MAY change the necessary destinations parameter. This
     will cause the media traffic to be sent to the wrong address.


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

5.6.4. 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 attackers 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 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 and not
   corresponding to the expected context. It will also notice the lack
   of incoming packets. See bullet 7 in section 5.6.2.

   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 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 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 access to a user
   account on the TURN server to be able set up the TURN mappings. To
   prevent this attack the server shall verify that the target
   destination accept this media stream.

6. 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

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   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 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 the media streams will use. 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 full cone NAT or a Symmetric 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 includes sending packets to the
   server to open up the mappings.

7. Open Issues

   The below list the current open issues with this draft:

   - The lost mappings text needs better text.
   - Their is need to decide on one of the server modifying schemes and
     ensure that a stable specification of that method exist. This
     decision process will require that requirements, security and
     desired goals are evaluated against implementation cost and
     probability to get it deployed.
   - The ALG recommendations needs to be improved and clearer.
   - The firewall RTSP ALG recommendations need to be written as they
     are different from the NAT ALG in some perspectives.

8. Security Consideration

   In preceding sessions we have discussed security merits of each and
   every NAT/FW traversal methods for RTSP. 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 these has security implications.

   Using STUN will provide the same level of security as RTSP with out
   transport level security and source authentications, as long as the

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   server do not grant a client request to send media to different IP

   Using symmetric RTP will have a slightly higher risk of session
   hijacking than normal RTSP. The reason is that there exists a
   probability that an attacker is able to guess the random tag that the
   client uses to prove its identity when creating the address bindings.

   The usage of an RTSP ALG does not increase in itself the risk for
   session hijacking. However the deployment of ALGs as sole mechanism
   for RTSP NAT traversal will prevent deployment of encrypted end-to-
   end RTSP signaling.

   The usage of TCP tunneling has no known security problems. However it
   might provide a bottleneck when it comes to end-to-end RTSP signaling
   security if TCP tunneling is used on a interleaved RTSP signaling

   The usage of TURN has high risk of denial of service attacks against
   a client. The TURN server can also be used as a redirect point in a
   DDOS attack unless the server has strict enough rules for who may
   create bindings.

9. IANA Consideration

   This specification would like to register 2 new Transport header
   parameters "sym_rtp" and "client_ssrc" as defined in section 5.3.2.

   It does also register one more RTSP feature tag "nat.sym-rtp" as
   defined in section 5.3.2.

10. Acknowledgments

   The author would also like to thank all persons on the MMUSIC working
   group's mailing list that has commented on this specification.
   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 protocol.

11. Author's Addresses

    Magnus Westerlund       Tel: +46 8 4048287
    Ericsson Research       Email:
    Ericsson AB
    Torshamnsgatan 23
    SE-164 80 Stockholm, SWEDEN

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    Thomas Zeng             Tel: 1-858-731-5465
    PacketVideo Corp.       Email:
    10350 Science Center Dr.
    San Diego, CA92121

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12. References

12.1. Normative references

   [1]  H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
        IETF RFC 2326, April 1998.
   [2]  M. Handley, V. Jacobson, "Session Description Protocol (SDP)",
        IETF RFC 2327, April 1998.
   [3]  D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
        tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov.
   [4]  S. Bradner, "Key words for use in RFCs to Indicate Requirement
        Levels", RFC 2119, March 1997.
   [5]  H. Schulzrinne, et. al., "RTP: A Transport Protocol for Real-
        Time Applications", IETF RFC 1889, January 1996.
   [6]  J. Rosenberg, et. Al., " STUN - Simple Traversal of UDP Through
        Network Address Translators", IETF RFC 3489, March 2003
   [7]  H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
        draft-ietf-mmusic-rfc2326bis-04.txt, IETF draft, June 2003, work
        in progress.
   [8]  J. Rosenberg, et. Al., "Traversal Using Relay NAT (TURN)",
        draft-rosenberg-midcom-turn-01.txt, IETF draft, March 2003, work
        in progress.
   [9]  J. Rosenberg, "Interactive Connectivity Establishment (ICE): A
        Methodology for Network Address Translator (NAT) Traversal for
        the Session Initiation Protocol (SIP)," draft-rosenberg-sipping-
        ice-00, IETF draft, February 2003, work in progress.
   [10] G. Camarillo, et. al., "Grouping of Media Lines in the Session
        Description Protocol (SDP)," IETF RFC 3388, December 2002.
   [11] G. Camarillo, J. Rosenberg, " The Alternative Semantics for the
        Session Description Protocol Grouping Framework," draft-
        camarillo-mmusic-alt-01.txt, IETF draft, June 2002, work in

12.2. Informative References

   [12] P. Srisuresh, K. Egevang, "Traditional IP Network Address
        Translator (Traditional NAT)," RFC 3022, Internet Engineering
        Task Force, January 2001.
   [13] Tsirtsis, G. and Srisuresh, P., "Network Address Translation -
        Protocol Translation (NAT-PT)", RFC 2766, Internet Engineering
        Task Force, February 2000.
   [14] S. Deering and R. Hinden, "Internet Protocol, Version 6 (IPv6)
        Specification", RFC 2460, Internet Engineering Task Force,
        December 1998.

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   [15] J. Postel, "internet protocol", RFC 791, Internet Engineering
        Task Force, September 1981.
   [16] D. Yon, "Connection-Oriented Media Transport in SDP", IETF
        draft, draft-ietf-mmusic-sdp-comedia-04.txt, July 2002.
   [17] John Lazzaro, "Framing RTP and RTCP Packets over Connection-
        Oriented Transport", IETF Draft, draft-lazzaro-avt-rtp-framing-
        contrans-00.txt, January 2003.
   [18] D. Daigle, "IAB Considerations for UNilateral Self-Address
        Fixing (UNSAF) Across Network Address Translation", RFC 3424,
        Internet Engineering Task Force, Nov. 2002
   [19] R. Finlayason, "IP Multicast and Firewalls", RFC 2588, Internet
        Engineering Task Force, May 1999
   [20] Krawczyk, H., Bellare, M., and Canetti, R.: "HMAC: Keyed-hashing
        for message authentication". IETF RFC 2104, February 1997
   [21] Open Source STUN Server and Client,

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   part, without restriction of any kind, provided that the above
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