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
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
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 failed. 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:
* 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 exchange. * 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 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. 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 Advantages: o No impact on either client or server. o Can work for any type of NATs.
Disadvantages: o When deployed, they are hard to update to reflect protocol modifications and extensions. If not updated, they will break the functionality. o When end-to-end security is used, the ALG functionality will fail. o Can interfere with other types of traversal mechanisms, such as STUN. Transition: 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 response. 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
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 issues. 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 tunneling. 4.8.4. Deployment Considerations Advantage: o Works through all types of NATs where the RTSP server is not NATed or is at least reachable like it was not. Disadvantages: o Functionality needs to be implemented on both server and client. o Will not always meet multimedia stream's real-time requirements.
Transition: The tunneling over RTSP's TCP connection is not planned to be phased out. It is intended to be a fallback mechanism and for usage when total media reliability is desired, even at the potential price of loss of real-time properties. 4.8.5. 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 target. A possible security consideration, when session media data is interleaved with RTSP, would be the performance bottleneck when RTSP encryption is applied, since all session media data also needs to be encrypted. 4.9. 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
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 performed. 1. The RTSP client connects with the RTSP server. The client retrieves the session description to determine the number of media streams. To avoid the issue 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 [RFC6062]. 2. The client establishes the necessary bindings on the TURN server. It must choose the local RTP and RTCP ports that it desires to receive media packets. TURN supports requesting bindings of even port numbers and 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 performed. 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 ports. 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 bindings.
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 signaling. 4.9.4. Deployment Considerations Advantages: 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. Disadvantages: 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 sessions. o The TURN server becomes a single point of failure. o Since TURN forwards media packets, as a necessity it introduces delay. o An RTSP ALG may change the necessary destinations parameter. This will cause the media traffic to be sent to the wrong address. Transition: TURN is not intended to be 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
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
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?
------------+------+------+------+------+------+------+------+ | 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].
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.
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 bindings. 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, <http://nice.freedesktop.org/wiki/>. [PJNATH] "PJNATH - Open Source ICE, STUN, and TURN Library", May 2013, <http://www.pjsip.org/pjnath/docs/html/>. [RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI 10.17487/RFC0768, August 1980, <http://www.rfc-editor.org/info/rfc768>. [RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, DOI 10.17487/RFC0793, September 1981, <http://www.rfc-editor.org/info/rfc793>. [RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, DOI 10.17487/RFC2326, April 1998, <http://www.rfc-editor.org/info/rfc2326>. [RFC2588] Finlayson, R., "IP Multicast and Firewalls", RFC 2588, DOI 10.17487/RFC2588, May 1999, <http://www.rfc-editor.org/info/rfc2588>.
[RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address Translator (NAT) Terminology and Considerations", RFC 2663, DOI 10.17487/RFC2663, August 1999, <http://www.rfc-editor.org/info/rfc2663>. [RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network Address Translator (Traditional NAT)", RFC 3022, DOI 10.17487/RFC3022, January 2001, <http://www.rfc-editor.org/info/rfc3022>. [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, <http://www.rfc-editor.org/info/rfc3261>. [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, <http://www.rfc-editor.org/info/rfc3424>. [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, <http://www.rfc-editor.org/info/rfc3489>. [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, <http://www.rfc-editor.org/info/rfc3550>. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006, <http://www.rfc-editor.org/info/rfc4566>. [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, <http://www.rfc-editor.org/info/rfc4571>. [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, <http://www.rfc-editor.org/info/rfc4787>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007, <http://www.rfc-editor.org/info/rfc4961>. [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, <http://www.rfc-editor.org/info/rfc5245>. [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, <http://www.rfc-editor.org/info/rfc5382>. [RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, DOI 10.17487/RFC5389, October 2008, <http://www.rfc-editor.org/info/rfc5389>. [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, <http://www.rfc-editor.org/info/rfc5764>. [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, <http://www.rfc-editor.org/info/rfc5766>. [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, <http://www.rfc-editor.org/info/rfc6062>. [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, <http://www.rfc-editor.org/info/rfc6263>. [RFC6275] Perkins, C., Ed., Johnson, D., and J. Arkko, "Mobility Support in IPv6", RFC 6275, DOI 10.17487/RFC6275, July 2011, <http://www.rfc-editor.org/info/rfc6275>.
[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, <http://www.rfc-editor.org/info/rfc7362>. [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, <http://sourceforge.net/projects/stun/>. Acknowledgements 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.