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RFC 7604

Comparison of Different NAT Traversal Techniques for Media Controlled by the Real-Time Streaming Protocol (RTSP)

Pages: 46
Part 3 of 3 – Pages 29 to 46
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4.6.  Three-Way Latching

4.6.1.  Introduction

   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., a UDP packet with some session
   identifiers 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 message is added to the exchange.  The
   client acknowledges the reception of the Latching confirmation
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   message and echoes back the server's nonce, thus confirming that the
   Latched address goes to an RTSP 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.  ALG Considerations

   See Latching ALG considerations in Section 4.4.3.

4.6.4.  Deployment Considerations

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

   o  Only works for servers that are not behind a NAT.

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

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

4.6.5.  Security Considerations

   Three-Way Latching is significantly more secure than its simpler
   versions discussed above.  The client-to-server nonce, which is
   included in signaling and also can be bigger than the 32 bits of
   random data that the SSRC field supports, makes it very difficult for
   an off-path attacker to perform a DoS attack by diverting the media.

   The client-to-server nonce and its echoing back does not protect
   against on-path attackers, including malicious clients.  However, the
   server-to-client nonce and its echoing back prevents malicious
   clients to divert the media stream by spoofing the source address and
   port, as it can't echo back the nonce in these cases.  This is
   similar to the Mobile IPv6 return routability procedure
   (Section 5.2.5 of [RFC6275]).

   Three-Way Latching is really only vulnerable to an on-path attacker
   that is quite capable.  First, the attacker can learn the client-
   to-server nonce either by intercepting the signaling or by modifying
   the source information (target destination) of a client's Latching
   packet.  Second, it is also on-path between the server and target
   destination and can generate a response using the server's nonce.  An
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   adversary that has these capabilities is commonly capable of causing
   significantly worse damage than this using other methods.

   Three-Way Latching results in the server-to-client packet being
   bigger than the client-to-server packet, due to the inclusion of the
   server-to-client nonce in addition to the client-to-server nonce.
   Thus, an amplification effect does exist; however, to achieve this
   amplification effect, the attacker has to create a session state on
   the RTSP server.  The RTSP server can also limit the number of
   responses it will generate before considering the Latching to be

4.7.  Application Level Gateways

4.7.1.  Introduction

   An ALG reads the application level messages and performs necessary
   changes to allow the protocol to work through the middlebox.
   However, this behavior has some problems in regards to RTSP:

   1.  It does not work when RTSP 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 middlebox.

   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 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 SOHO environments.

4.7.2.  Outline on How ALGs for RTSP Work

   In this section, we provide a step-by-step outline on how one 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 change the address.  Ways to detect that
       these methods are used are:
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       *  For embedded STUN, it would be to watch for a feature tag,
          like "nat.stun", and to see if any of those exist in the
          "supported", "proxy-require", or "require" headers of the RTSP

       *  For stand-alone STUN and TURN-based solutions: This can 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, 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 NAT's own private address space.

       Otherwise, continue to the next step.

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

   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.

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

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

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

4.7.3.  Deployment Considerations


   o  No impact on either client or server.

   o  Can work for any type of NATs.
Top   ToC   Page 33

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

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

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


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

4.7.4.  Security Considerations

   An ALG will not work with deployment of end-to-end RTSP signaling
   security; however, it will work with the hop-by-hop security method
   defined in Section 19.3 of RTSP 2.0 [RTSP].  Therefore, deployment of
   ALG may result in clients located behind NATs not using end-to-end
   security, or more likely the selection of a NAT traversal solution
   that allows for security.

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

   A NAT with RTSP ALG that assigns mappings based on SETUP requests
   could potentially become the victim of a resource exhaustion attack.
   If an attacker creates a lot of RTSP sessions, even without starting
   media transmission, this 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 ALG.

4.8.  TCP Tunneling

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
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   transported over UDP requires the TCP connection to support some type
   of framing of the media data packets.  Using TCP also results in the
   client having to accept that real-time performance can be impacted.
   TCP's problem of ensuring timely delivery was one of the reasons why
   RTP was developed.  Problems that arise with TCP are: head-of-line
   blocking, delay introduced by retransmissions, and a highly varying
   rate due to the congestion control algorithm.  If a sufficient amount
   of buffering (several seconds) in the receiving client can be
   tolerated, then TCP will clearly work.

4.8.2.  Usage of TCP Tunneling in RTSP

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

4.8.3.  ALG Considerations

   An RTSP ALG will face a different issue with TCP tunneling, at least
   the interleaved version.  Now the full data stream can end up flowing
   through the ALG implementation.  Thus, it is important that the ALG
   is efficient in dealing with the interleaved media data frames to
   avoid consuming to many resources and thus creating performance

   The RTSP ALG can also affect the transport specifications that
   indicate that TCP tunneling can be done and its prioritization,
   including removing the transport specification, thus preventing TCP

4.8.4.  Deployment Considerations


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


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

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

4.8.5.  Security Considerations

   The TCP tunneling of RTP has no known significant security problems
   besides those already presented in the RTSP specification.  It is
   difficult to get any amplification effect for DoS attacks due to
   TCP's flow control.  The RTSP server's TCP socket, if independently
   used for media tunneling or only RTSP messages, can be used for a
   redirected syn attack.  By spoofing the source address of any TCP
   init packets, the TCP SYNs from the server can be directed towards a

   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

4.9.  Traversal Using Relays around NAT (TURN)

4.9.1.  Introduction

   TURN [RFC5766] is a protocol for setting up traffic relays that 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 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 allocation.  On the external
   side, it is limited to the source address/port pair that have been
   given permission by the TURN client creating the allocation.  Packets
   from any other source on this address will be discarded.

   Using a TURN server makes it possible for an 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 from.
   This means that TURN could only be used if the server knows and
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   accepts that the IP belongs to a TURN server, and the TURN server
   can't be targeted at an unknown address.  Alternatively, both the
   RTSP TCP connection as well as the RTP media is relayed through the
   same TURN server.

4.9.2.  Usage of TURN with RTSP

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

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

   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 contiguous ranges.

   3.  The RTSP client uses the acquired address and port allocations in
       the RTSP SETUP request using the destination header.

   4.  The RTSP server sends the SETUP reply, which must include the
       Transport header's "src_addr" parameter (source and port in RTSP
       1.0).  Note that the server is required to have a mechanism to
       verify that it is allowed to send media traffic to the given
       address unless TCP relaying of the RTSP messages also is

   5.  The RTSP client uses the RTSP server's response to create TURN
       permissions for the server's media traffic.

   6.  The client requests that the server starts playing.  The server
       starts sending media packets to the given destination address and

   7.  Media packets arrive at the TURN server on the external port; if
       the packets match an established permission, the TURN server
       forwards the media packets to the RTSP client.

   8.  If the client pauses and media is not sent for about 75% of the
       mapping timeout, the client should use TURN to refresh the
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4.9.3.  ALG Considerations

   As the RTSP client inserts the address information of the TURN
   relay's external allocations in the SETUP messages, the ALG that
   replaces the address, without considering that the address does not
   belong to the internal address realm of the NAT, will prevent this
   mechanism from working.  This can be prevented by securing the RTSP

4.9.4.  Deployment Considerations


   o  Does not require any server modifications given that the server
      includes the "src_addr" header in the SETUP response.

   o  Works for any type of NAT as long as the RTSP server has a
      reachable IP address that is not behind a NAT.


   o  Requires another network element, namely the TURN server.

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

   o  The TURN server becomes a single point of failure.

   o  Since TURN forwards media packets, as a necessity it introduces

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


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

4.9.5.  Security Considerations

   The TURN server can become part of a DoS 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 set up an RTSP session with
   media flows going through the TURN server.  The attacker is in fact
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   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 an RTSP session and 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.

   This attack requires that the attacker has access to a user account
   on the TURN server to be able to set up the TURN mappings.  To
   prevent this attack, the RTSP server needs to verify that the
   ultimate target destination accepts 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 an
   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 the media stream packets will 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.

   The above possibilities for firewalls to inspect and respond to the
   signaling are prevented if end-to-end confidentiality protection is
   used for the RTSP signaling, e.g., using the specified RTSP over TLS.
   As a result, firewalls can't be actively opening pinholes for the
   media streams based on the signaling.  To enable an RTSP ALG in the
   firewall to correctly function, the hop-by-hop signaling security in
   RTSP 2.0 can be used (see Section 19.3 of [RTSP]).  If not, other
   methods have to be used to enable the transport flows for the media.

   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
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   that no address translation is done.  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
   create the necessary filtering state.

6.  Comparison of NAT Traversal Techniques

   This section evaluates the techniques described above against the
   requirements listed in 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 3: must work for all flavors of NATs.
   The rows represent the different NAT/firewall traversal techniques.
   Latch is short for Latching, "V. Latch" is short for "variation of
   Latching" as described in Section 4.5, and "3-W Latch" is short for
   the Three-Way Latching described in Section 4.6.

   A summary of the requirements are:

   R1: Work for all flavors of NATs

   R2: Must work with firewalls, including those with ALGs

   R3: Should have minimal impact on clients not behind NATs, counted in
       minimal number of additional RTTs

   R4: Should be simple to use, implement, and administer

   R5: Should provide mitigation against DDoS attacks

   The following considerations are also added to the requirements:

   C1: Will the solution support both clients and servers behind NAT?

   C2: Is the solution robust as NAT behaviors change?
Top   ToC   Page 40
               |  R1  |  R2  |  R3  |  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  |
    V. Latch   | Yes  | Yes  |  1   | Yes  | No   | No   | Yes  |
    3-W Latch  | Yes  | Yes  | 1.5  | Maybe| Yes  | No   | Yes  |
    ALG        |(Yes) | Yes  |  0   | No   | Yes  | No   | Yes  |
    TCP Tunnel | Yes  | Yes  | 1.5  | Yes  | Yes  | No   | Yes  |
    TURN       | Yes  | Yes  |  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 are the solutions that best fulfill
   the requirements.  The different techniques 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 also handling
   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 was some discussion in the WG about if
   the increased implementation burden of ICE is sufficiently motivated
   compared to a the Three-Way Latching solution for this generality.
   In the end, the authors believed that the reuse of ICE, greater
   flexibility, and any way needed to deploy a new solution were 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)" [RTSP-NAT].
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7.  Security Considerations

   In the preceding sections, we have discussed security merits of the
   different NAT/firewall 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
   the 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 without transport-
   level security and source authentications, as long as the server does
   not allow media to be sent to a different IP address than the RTSP
   client request was sent from.

   Using Latching will have a higher risk of session hijacking or DoS
   than normal RTSP.  The reason is that there exists a probability that
   an attacker is able to guess the random bits that the client uses to
   prove its identity when creating the address bindings.  This can be
   solved in the variation of Latching (Section 4.5) with authentication
   features.  Still, both those variants of Latching are vulnerable
   against a 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.

   ICE resolves the binding vulnerability of Latching by using signed
   STUN messages, as well as requiring that both sides perform
   connectivity checks to verify that the target IP address in the
   candidate pair is both reachable and willing to respond.  ICE can,
   however, create a significant amount of traffic if the number of
   candidate pairs are large.  Thus, pacing is required and
   implementations should attempt to limit their number of candidates to
   reduce the number of packets.

   If the signaling between the ICE peers (RTSP client and server) is
   not confidentiality and integrity protected, ICE is vulnerable to
   attacks where the candidate list is manipulated.  The lack of
   signaling security will also simplify spoofing of STUN binding
   messages by revealing the secret used in signing.

   The usage of an RTSP ALG does not 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 end-
   to-end encrypted RTSP signaling.
Top   ToC   Page 42
   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 an interleaved RTSP
   signaling connection.

   The usage of TURN has severe risk of DoS 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

   Since Latching and the variants of Latching have such big security
   issues, they should not be used at all.  Three-Way Latching as well
   as ICE mitigates these security issues and performs the important
   return-routability checks that prevent spoofed source addresses, and
   they should be recommended for that reason.  RTP ALGs are a security
   risk as they can create an incitement against using secure RTSP
   signaling.  That can be avoided as ALGs require trust in the
   middlebox, and that trust becomes explicit if one uses the hop-by-hop
   security solution as specified in Section 19.3 of RTSP 2.0.  [RTSP].
   The remaining methods can be considered safe enough, assuming that
   the appropriate security mechanisms are used and not ignored.

8.  Informative References

   [NICE]      Libnice, "The GLib ICE implementation", June 2015,

   [PJNATH]    "PJNATH - Open Source ICE, STUN, and TURN Library", May
               2013, <>.

   [RFC768]    Postel, J., "User Datagram Protocol", STD 6, RFC 768,
               DOI 10.17487/RFC0768, August 1980,

   [RFC793]    Postel, J., "Transmission Control Protocol", STD 7,
               RFC 793, DOI 10.17487/RFC0793, September 1981,

   [RFC2326]   Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
               Streaming Protocol (RTSP)", RFC 2326,
               DOI 10.17487/RFC2326, April 1998,

   [RFC2588]   Finlayson, R., "IP Multicast and Firewalls", RFC 2588,
               DOI 10.17487/RFC2588, May 1999,
Top   ToC   Page 43
   [RFC2663]   Srisuresh, P. and M. Holdrege, "IP Network Address
               Translator (NAT) Terminology and Considerations",
               RFC 2663, DOI 10.17487/RFC2663, August 1999,

   [RFC3022]   Srisuresh, P. and K. Egevang, "Traditional IP Network
               Address Translator (Traditional NAT)", RFC 3022,
               DOI 10.17487/RFC3022, January 2001,

   [RFC3261]   Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
               A., Peterson, J., Sparks, R., Handley, M., and
               E. Schooler, "SIP: Session Initiation Protocol",
               RFC 3261, DOI 10.17487/RFC3261, June 2002,

   [RFC3424]   Daigle, L., Ed. and IAB, "IAB Considerations for
               UNilateral Self-Address Fixing (UNSAF) Across Network
               Address Translation", RFC 3424, DOI 10.17487/RFC3424,
               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,
               DOI 10.17487/RFC3489, March 2003,

   [RFC3550]   Schulzrinne, H., Casner, S., Frederick, R., and V.
               Jacobson, "RTP: A Transport Protocol for Real-Time
               Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
               July 2003, <>.

   [RFC4566]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
               Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
               July 2006, <>.

   [RFC4571]   Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
               and RTP Control Protocol (RTCP) Packets over Connection-
               Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July
               2006, <>.

   [RFC4787]   Audet, F., Ed. and C. Jennings, "Network Address
               Translation (NAT) Behavioral Requirements for Unicast
               UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
               2007, <>.
Top   ToC   Page 44
   [RFC4961]   Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
               BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,

   [RFC5245]   Rosenberg, J., "Interactive Connectivity Establishment
               (ICE): A Protocol for Network Address Translator (NAT)
               Traversal for Offer/Answer Protocols", RFC 5245,
               DOI 10.17487/RFC5245, April 2010,

   [RFC5382]   Guha, S., Ed., Biswas, K., Ford, B., Sivakumar, S., and
               P.  Srisuresh, "NAT Behavioral Requirements for TCP",
               BCP 142, RFC 5382, DOI 10.17487/RFC5382, October 2008,

   [RFC5389]   Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
               "Session Traversal Utilities for NAT (STUN)", RFC 5389,
               DOI 10.17487/RFC5389, October 2008,

   [RFC5764]   McGrew, D. and E. Rescorla, "Datagram Transport Layer
               Security (DTLS) Extension to Establish Keys for the
               Secure Real-time Transport Protocol (SRTP)", RFC 5764,
               DOI 10.17487/RFC5764, May 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,
               DOI 10.17487/RFC5766, April 2010,

   [RFC6062]   Perreault, S., Ed. and J. Rosenberg, "Traversal Using
               Relays around NAT (TURN) Extensions for TCP Allocations",
               RFC 6062, DOI 10.17487/RFC6062, 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,
               DOI 10.17487/RFC6263, June 2011,

   [RFC6275]   Perkins, C., Ed., Johnson, D., and J. Arkko, "Mobility
               Support in IPv6", RFC 6275, DOI 10.17487/RFC6275, July
               2011, <>.
Top   ToC   Page 45
   [RFC7362]   Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
               Traversal (HNT) for Media in Real-Time Communication",
               RFC 7362, DOI 10.17487/RFC7362, September 2014,

   [RTP-NO-OP] Andreasen, F., "A No-Op Payload Format for RTP", Work in
               Progress, draft-ietf-avt-rtp-no-op-04, May 2007.

   [RTSP]      Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
               and M. Stiemerling, "Real Time Streaming Protocol 2.0
               (RTSP)", Work in Progress,
               draft-ietf-mmusic-rfc2326bis-40, February 2014.

   [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)", Work
               in Progress, draft-ietf-mmusic-rtsp-nat-22, July 2014.

   [STUN-IMPL] "Open Source STUN Client and Server", May 2013,


   The authors would also like to thank all persons on the MMUSIC
   working group's mailing list that have commented on this document.
   Persons having contributed to this protocol, in no special order,
   are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon,
   Amir Wolf, Anders Klemets, Flemming Andreasen, Ari Keranen, Bill
   Atwood, Alissa Cooper, Colin Perkins, Sarah Banks, David Black, and
   Alvaro Retana.  Thomas Zeng would also like to give special thanks to
   Greg Sherwood of PacketVideo for his input into this memo.

   Section 1.1 contains text originally written for RFC 4787 by Francois
   Audet and Cullen Jennings.
Top   ToC   Page 46
Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   Stockholm  SE-164 80

   Phone: +46 8 719 0000

   Thomas Zeng
   PacketVideo Corp