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

There exists a number of potential NAT traversal techniques that can be used to allow RTSP to traverse NATs. They have different features and are applicable to different topologies; their costs are also different. They also vary in security levels. In the following sections, each technique is outlined with discussions on the corresponding advantages and disadvantages. The survey of traversal techniques was done prior to 2007 and is based on what was available then. This section includes NAT traversal techniques that have not been formally specified anywhere else. This document may be the only publicly available specification of some of the NAT traversal techniques. However, that is not a real barrier against doing an evaluation of the NAT traversal techniques. Some techniques used as part of some of the traversal solutions have been recommended against or are no longer possible due to the outcome of standardization work or their failure to progress within IETF after the initial evaluation in this document. For example, RTP No&nbhy;Op was a proposed RTP payload format that failed IETF approval; thus, it is not available for use today. In each such case, the missing parts will be noted and some basic reasons will be given.

Session Traversal Utilities for NAT (STUN) is a standardized protocol that allows a client to use secure means to discover the presence of a NAT between itself and the STUN server. The client uses the STUN server to discover the address and port mappings assigned by the NAT. Then using the knowledge of these NAT mappings, it uses the external addresses to directly connect to the independent RTSP server. However, this is only possible if the NAT address and port mapping behavior is such that the STUN server and RTSP server will see the same external address and port for the same internal address and port. STUN is a client-server protocol. The STUN client sends a request to a STUN server and the server returns a response. There are two types of STUN messages -- Binding Requests and Indications. Binding Requests are used when determining a client's external address and soliciting a response from the STUN server with the seen address. Indications are used by the client for keep-alive messages towards the server and requires no response from the server. The first version of STUN included categorization and parameterization of NATs. This was abandoned in the updated version due to it being unreliable and brittle. This particular traversal method uses the removed functionality described in RFC 3489 to detect the NAT type to give an early failure indication when the NAT is showing the behavior that this method can't support. This method also suggests using the RTP No-Op payload format for keep&nbhy;alives of the RTP traffic in the client-to-server direction. This can be replaced with another form of UDP packet as will be further discussed below.

This section describes how a client can use STUN to traverse NATs to RTSP servers without requiring server modifications. Note that this method has limited applicability and requires the server to be available in the external/public address realm in regards to the client located behind a NAT(s). Limitations: 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 perform Endpoint-Independent or Address-Dependent Mappings (the STUN server and RTSP server on the same IP address). Clients behind NATs that do Address and Port-Dependent Mappings cannot use this method. See for the full definition of these terms. Based on the discontinued middlebox classification of the replaced STUN specification; thus, it is brittle and unreliable. Method: An RTSP client using RTP transport over UDP can use STUN to traverse a NAT(s) in the following way: Use STUN to try to discover the type of NAT 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 delays in the session establishment will be reduced. If no NAT is detected, normal SETUP should be used. The RTSP client determines the number of UDP ports needed by counting the number of needed media transport protocols sessions in the multimedia 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. For each UDP port required, establish a mapping and discover the public/external IP address and port number with the help of the STUN server. A successful mapping looks like: client's local address/port <-> public address/port. Perform the RTSP SETUP for each media. In the Transport header, the following parameter should be included with the given values: "dest_addr" or "destination" + "client_port" with the public/external IP address and port pair for both RTP and RTCP. To be certain that this works, servers must allow a client to set up the RTP stream on any port, not only even ports and with non-contiguous port numbers for RTP and RTCP. This requires the new feature provided in RTSP 2.0. The server should respond with a Transport header containing an "src_addr" or "source" + "server_port" parameters with the RTP and RTCP source IP address and port of the media stream. To keep the mappings alive, the client should periodically send UDP traffic over all mappings needed for the session. For the mapping carrying RTCP traffic, the periodic RTCP traffic is likely enough. For mappings carrying RTP traffic and for mappings carrying RTCP packets at too low of a frequency, keep-alive messages should be sent. If a UDP mapping is lost, the above discovery process must be repeated. The media stream also needs to be SETUP again to change the transport parameters to the new ones. This will cause a glitch in media playback. To allow UDP packets to arrive from the server to a client behind an Address-Dependent or Address and Port-Dependent Filtering NAT, the client must first send a UDP packet to establish the filtering state in the NAT. The client, before sending an RTSP PLAY request, must send a so-called hole-punching packet on each mapping to the IP address and port given as the server's source address and port. For a NAT that only is Address-Dependent Filtering, the hole-punching packet could be sent to the server's discard port (port number 9). For Address and Port-Dependent Filtering NATs, the hole-punching packet must go to the port used for sending UDP packets to the client. To be able to do that, the server needs to include the "src_addr" in the Transport header (which is the "source" transport parameter in RFC2326). Since UDP packets are inherently unreliable, to ensure that at least one UDP message passes the NAT, hole-punching packets should be retransmitted a reasonable number of times. One could have used RTP No-Op packets as hole-punching and keep-alive messages had they been defined. That would have ensured that the traffic would look like RTP and thus would likely have the least risk of being dropped by any firewall. The drawback of using RTP No-Op is that the payload type number must be dynamically assigned through RTSP first. Other options are STUN, an RTP packet without any payload, or a UDP packet without any payload. For RTCP it is most suitable to use correctly generated RTCP packets. In general, sending unsolicited traffic to the RTSP server may trigger security functions resulting in the blocking of the keep-alive messages or termination of the RTSP session itself. This method is further brittle as it doesn't support Address and Port-Dependent Mappings. Thus, it proposes to use the old STUN methods to classify the NAT behavior, thus enabling early error indication. This is strictly not required but will lead to failures during setup when the NAT has the wrong behavior. This failure can also occur if the NAT changes the properties of the existing mapping and filtering state or between the classification message exchange and the actual RTSP session setup, for example, due to load.

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 NAT external 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. An ALG should not update address fields that contain addresses other than the NAT's internal address domain. In cases where the ALG modifies fields unnecessarily, two alternatives exist: Use Transport Layer Security (TLS) to encrypt the data over the RTSP TCP connection to prevent the ALG from reading and modifying the RTSP messages. Turn off the STUN-based NAT traversal mechanism. As it may be difficult to determine why the failure occurs, the usage of TLS-protected RTSP message exchange at all times would avoid this issue.

For the stand-alone usage of STUN, the following applies: Advantages: STUN is a solution first used by applications based on SIP (see Sections 1 and 2 of ). As shown above, with little or no changes, the RTSP application can reuse STUN as a NAT traversal solution, avoiding the pitfall of solving a problem twice. Using STUN does not require RTSP server modifications, assuming it is a server that is compliant with RTSP 2.0; it only affects the client implementation. Disadvantages: Requires a STUN server deployed in the same address domain as the server. Only works with NATs that perform Endpoint-Independent and Address-Dependent Mappings. Address and Port-Dependent Filtering NATs create some issues. Brittle to NATs changing the properties of the NAT mapping and filtering. Does not work with Address and Port-Dependent Mapping NATs without server modifications. Will not work if a NAT uses multiple IP addresses, since RTSP servers generally require all media streams to use the same IP as used in the RTSP connection to prevent becoming a DDoS tool. Interaction problems exist when an RTSP-aware ALG interferes with the use of STUN for NAT traversal unless TLS-secured RTSP message exchange is used. Using STUN requires that RTSP servers and clients support the updated RTSP specification, because it is no longer possible to guarantee that RTP and RTCP ports are adjacent to each other, as required by the "client_port" and "server_port" parameters in RFC 2326. Transition: The usage of STUN can be phased out gradually as the first step of a STUN-capable server or client should be to check the presence of NATs. The removal of STUN capability in the client implementations will have to wait until there is absolutely no need to use STUN.

To prevent the RTSP server from being used as Denial-of-Service (DoS) attack tools, the RTSP Transport header parameters "destination" and "dest_addr" are generally not allowed to point to any IP address other than the one the RTSP message originates from. The RTSP server is only prepared to make an exception to this rule when the client is trusted (e.g., through the use of a secure authentication process or through some secure method of challenging the destination to verify its willingness to accept the RTP traffic). Such a restriction means that STUN in general does not work for use cases where RTSP and media transport go to different addresses. STUN combined with RTSP that is restricted by destination address has the same security properties as the core RTSP. It is protected from being used as a DoS attack tool unless the attacker has the ability to spoof the TCP connection carrying RTSP messages. Using STUN's support for message authentication and the secure transport of RTSP messages, attackers cannot modify STUN responses or RTSP messages (TLS) to change the media destination. This protects against hijacking; however, as a client can be the initiator of an attack, these mechanisms cannot securely prevent RTSP servers from being used as DoS attack tools.

This section describes an alternative to the stand-alone STUN usage in the previous section that has quite significantly different behavior.

This section outlines the adaptation and embedding of STUN within RTSP. This enables STUN to be used to traverse any type of NAT, including Address and Port-Dependent Mapping NATs. This would require RTSP-level protocol changes. This NAT traversal solution has limitations: It does not work if both the RTSP client and RTSP server are behind separate NATs. The RTSP server may, for security reasons, refuse to send media streams to an IP that is different from the IP in the client RTSP requests. Deviations from STUN as defined in RFC 5389: The RTSP application must provision the client with an identity and shared secret to use in the STUN authentication; We require the STUN server to be co-located on the RTSP server's media source ports. If the STUN server is co-located with the RTSP server's media source port, an RTSP client using RTP transport over UDP can use STUN to traverse ALL types of NATs. In the case of Address and Port-Dependent Mapping NATs, the party on the inside of the NAT must initiate UDP traffic. The STUN Binding Request, being a UDP packet itself, can serve as the traffic initiating packet. Subsequently, both the STUN Binding Response packets and the RTP/RTCP packets can traverse the NAT, regardless of whether the RTSP server or the RTSP client is behind NAT (however, only one of them can be behind a NAT). Likewise, if an RTSP server is behind a NAT, then an embedded STUN server must be co-located on the RTSP client's RTCP port. Also, it will become the client that needs to disclose his destination address rather than the server, so the server can correctly determine its NAT external source address for the media streams. In this case, we assume that the client has some means of establishing a TCP connection to the RTSP server behind NAT so as to exchange RTSP messages with the RTSP server, potentially using a proxy or static rules. To minimize delay, we require that the RTSP server supporting this option must inform the client about the RTP and RTCP ports from where the server will send out RTP and RTCP packets, respectively. This can be done by using the "server_port" parameter in RFC 2326 and the "src_addr" parameter in . Both are in the RTSP Transport header. But in general, this strategy will require that one first does one SETUP request per media to learn the server ports, then perform the STUN checks, followed by a subsequent SETUP to change the client port and destination address to what was learned during the STUN checks. To be certain that RTCP works correctly, the RTSP endpoint (server or client) will be required to send and receive RTCP packets from the same port.

In order to use STUN to traverse Address and Port-Dependent Filtering or Mapping NATs, the STUN server needs to be co-located with the streaming server media output ports. This creates a demultiplexing problem: we must be able to differentiate a STUN packet from a media packet. This will be done based on heuristics. The existing STUN heuristics is the first byte in the packet and the Magic Cookie field (added in RFC 5389), which works fine between STUN and RTP or RTCP where the first byte happens to be different. Thanks to the Magic Cookie field, it is unlikely that other protocols would be mistaken for a STUN packet, but this is not assured. For more discussion of this, please see Section 5.1.2 of .

The same ALG traversal considerations as for stand-alone STUN applies.

For the "Embedded STUN" method the following applies: Advantages: STUN is a solution first used by SIP applications. As shown above, with little or no changes, the RTSP application can reuse STUN as a NAT traversal solution, avoiding the pitfall of solving a problem twice. STUN has built-in message authentication features, which makes it more secure against hijacking attacks. See the next section for an in-depth security discussion. This solution works as long as there is only one RTSP endpoint in the private address realm, regardless of the NAT's type. There may even be multiple NATs (see Figure 1 in ). Compared to other UDP-based NAT traversal methods in this document, STUN requires little new protocol development (since STUN is already an IETF standard), and most likely less implementation effort, since open source STUN server and client implementations are available . Disadvantages: Some extensions to the RTSP core protocol, likely signaled by RTSP feature tags, must be introduced. Requires an embedded STUN server to be co-located on each of the RTSP server's media protocol's ports (e.g., RTP and RTCP ports), which means more processing is required to demultiplex STUN packets from media packets. For example, the demultiplexer must be able to differentiate an RTCP RR packet from a STUN packet and forward the former to the streaming server and the latter to the STUN server. Does not support use cases that require the RTSP connection and the media reception to happen at different addresses, unless the server's security policy is relaxed. Interaction problems exist when an RTSP ALG is not aware of STUN unless TLS is used to protect the RTSP messages. Using STUN requires that RTSP servers and clients support the updated RTSP specification, and they both agree to support the NAT traversal feature. Increases the setup delay with at least the amount of time it takes to perform STUN message exchanges. Most likely an extra SETUP sequence will be required. Transition: The usage of STUN can be phased out gradually as the first step of a STUN-capable machine can be used to check the presence of NATs for the presently used network connection. The removal of STUN capability in the client implementations will have to wait until there is absolutely no need to use STUN, i.e., no NATs or firewalls.

See Stand-Alone STUN.

Interactive Connectivity Establishment (ICE) is a methodology for NAT traversal that has been developed for SIP using SDP offer/answer. The basic idea is to try, in a staggered parallel fashion, all possible connection addresses in which an endpoint may be reached. This allows the endpoint to use the best available UDP "connection" (meaning two UDP endpoints 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 at a high level. Endpoint A collects all possible addresses that can be used, including local IP addresses, STUN-derived addresses, Traversal Using Relay NAT (TURN) addresses, etc. On each local port that any of these address and port pairs lead to, a STUN server is installed. This STUN server only accepts STUN requests using the correct authentication through the use of a username and password. Endpoint A then sends a request to establish connectivity with endpoint B, which includes all possible "destinations" to get the media through to A. Note that each of A's local address/port pairs (host candidates and server reflexive base) has a co-located STUN server. B in turn provides A with all its possible destinations for the different media streams. A and B then uses a STUN client to try to reach all the address and port pairs specified by A from its corresponding destination ports. The destinations for which the STUN requests successfully complete are then indicated and one is selected. If B fails to get any STUN response from A, all hope is not lost. Certain NAT topologies require multiple tries from both ends before successful connectivity is accomplished; therefore, requests are retransmitted multiple times. The STUN requests may also result in more connectivity alternatives (destinations) being discovered and conveyed in the STUN responses.

The usage of ICE for RTSP requires that both client and server be updated to include the ICE functionality. If both parties implement the necessary functionality, the following steps could provide ICE support for RTSP. This assumes that it is possible to establish a TCP connection for the RTSP messages between the client and the server. This is not trivial in scenarios where the server is located behind a NAT, and may require some TCP ports be opened, or proxies are deployed, etc. The negotiation of ICE in RTSP of necessity will work different than in SIP with SDP offer/answer. The protocol interactions are different, and thus the possibilities for transfer of states are also somewhat different. The goal is also to avoid introducing extra delay in the setup process at least for when the server is not behind a NAT in regards to the client, and the client is either having an address in the same address domain or is behind the NAT(s), which can address the address domain of the server. This process is only intended to support PLAY mode, i.e., media traffic flows from server to client. ICE usage begins in the SDP. The SDP for the service indicates that ICE is supported at the server. No candidates can be given here as that would not work with on demand, DNS load balancing, etc., which have the SDP indicate a resource on a server park rather than a specific machine. The client gathers addresses and puts together its candidates for each media stream indicated in the session description. In each SETUP request, the client includes its candidates in an ICE-specific transport specification. For the server, this indicates the ICE support by the client. One candidate is the most prioritized candidate and here the prioritization for this address should be somewhat different compared to SIP. High-performance candidates are recommended rather than candidates with the highest likelihood of success, as it is more likely that a server is not behind a NAT compared to a SIP user agent. The server responds to the SETUP (200 OK) for each media stream with its candidates. A server not behind a NAT usually only provides a single ICE candidate. Also, here one candidate is the server primary address. The connectivity checks are performed. For the server, the connectivity checks from the server to the clients have an additional usage. They verify that there is someone willing to receive the media, thus preventing the server from unknowingly performing a DoS attack. Connectivity checks from the client promoting a candidate pair were successful. Thus, no further SETUP requests are necessary and processing can proceed with step 7. If an address other than the primary has been verified by the client to work, that address may then be promoted for usage in a SETUP request (go to step 7). If the checks for the available candidates failed and if further candidates have been derived during the connectivity checks, then those can be signaled in new candidate lines in a SETUP request updating the list (go to step 5). Client issues the PLAY request. If the server also has completed its connectivity checks for the promoted candidate pair (based on the username as it may be derived addresses if the client was behind NAT), then it can directly answer 200 OK (go to step 8). If the connectivity check has not yet completed, it responds with a 1xx code to indicate that it is verifying the connectivity. If that fails within the set timeout, an error is reported back. The client needs to go back to step 6. Process completed and media can be delivered. ICE candidates not used may be released. To keep media paths alive, the client needs to periodically send data to the server. This will be realized with STUN. RTCP sent by the client should be able to keep RTCP open, but STUN will also be used for SIP based on the same motivations as for 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 among all the NAT/firewall traversal methods in this document. RTSP server implementation requirements are: STUN server features Limited STUN client features SDP generation with more parameters RTSP error code for ICE extension RTSP client implementation requirements are: Limited STUN server features Limited STUN client features RTSP error code and ICE extension

If there is an RTSP ALG that doesn't support the NAT traversal method, it may interfere with the NAT traversal. As the usage of ICE for the traversal manifests itself in the RTSP message primarily as a new transport specification, an ALG that passes through unknown will not prevent the traversal. An ALG that discards unknown specifications will, however, prevent the NAT traversal. These issues can be avoided by preventing the ALG to interfere with the signaling by using TLS for the RTSP message transport. An ALG that supports this traversal method can, on the most basic level, just pass the transport specifications through. ALGs in NATs and firewalls could use the ICE candidates to establish a filtering state that would allow incoming STUN messages prior to any outgoing hole-punching packets, and in that way it could speed up the connectivity checks and reduce the risk of failures.

Advantages: Solves NAT connectivity discovery for basically all cases as long as a TCP connection between the client and server can be established. This includes servers behind NATs. (Note that a proxy between address domains may be required to get TCP through.) Improves defenses against DDoS attacks, since a media-receiving client requires authentications via STUN on its media reception ports. Disadvantages: Increases the setup delay with at least the amount of time it takes for the server to perform its STUN requests. Assumes that it is possible to demultiplex between the packets of the media protocol and STUN packets. This is possible for RTP as discussed, for example, in Section 5.1.2 of . Has a fairly high implementation burden put on both the RTSP server and client. However, several open source ICE implementations do exist, such as and .

One should review the Security Considerations section of ICE and STUN to understand that ICE contains some potential issues. However, these can be avoided by correctly using ICE in RTSP. An important factor is to secure the signaling, i.e., use TLS between the RTSP client and server. In fact ICE does help avoid the DDoS attack issue with RTSP substantially as it reduces the possibility for a DDoS using RTSP servers on attackers that are on path between the RTSP server and the target and capable of intercepting the STUN connectivity check packets and correctly sending a response to the server. The ICE connectivity checks with their random transaction IDs from the server to the client serves as a return-routability check and prevents off-path attackers to succeed with address spoofing. This is similar to Mobile IPv6's return routability procedure (Section 5.2.5 of ).

Latching is a NAT traversal solution that is based on requiring RTSP clients to send UDP packets to the server's media output ports. Conventionally, RTSP servers send RTP packets in one direction: from server to client. Latching is similar to connection-oriented traffic, where one side (e.g., the RTSP client) first "connects" by sending an RTP packet to the other side's RTP port; the recipient then replies to the originating IP and Port. This method is also referred to as "late binding". It requires that all RTP/RTCP transport be done symmetrically. This in effect requires Symmetric RTP . Refer to for a description of the Latching of SIP-negotiated media streams in Session Border Controllers. Specifically, when the RTSP server receives the Latching packet (a.k.a. hole-punching packet, since it is used to punch a hole in the firewall/NAT) from its client, it copies the source IP and Port number and uses them as the delivery address for media packets. By having the server send media traffic back the same way as the client's packets are sent to the server, address and port mappings will be honored. Therefore, this technique works for all types of NATs, given that the server is not behind a NAT. However, it does require server modifications. The format of the Latching packet will have to be defined. Latching is very vulnerable to both hijacking and becoming a tool in DDoS attacks (see Security Considerations in ) because attackers can simply forge the source IP and Port of the Latching packet. The rule for restricting IP addresses to one of the signaling connections will need to be applied here also. However, that does not protect against hijacking from another client behind the same NAT. This can become a serious issue in deployments with CGNs.

To support Latching, RTSP signaling must be extended to allow the RTSP client to indicate that it will use Latching. The client also needs to be able to signal its RTP SSRC to the server in its SETUP request. The RTP SSRC is used to establish some basic level of security against hijacking attacks or simply to avoid mis-association when multiple clients are behind the same NAT. Care must be taken in choosing clients´ RTP SSRC. First, it must be unique within all the RTP sessions belonging to the same RTSP session. Second, if the RTSP server is sending out media packets to multiple clients from the same send port, the RTP SSRC needs to be unique among those clients' RTP sessions. Recognizing that there is a potential that RTP SSRC collisions may occur, the RTSP server must be able to signal to a client that a collision has occurred and that it wants the client to use a different RTP SSRC carried in the SETUP response or use unique ports per RTSP session. Using unique ports limits an RTSP server in the number of sessions it can simultaneously handle per interface IP addresses. The Latching packet as discussed above should have a field that can contain a client and RTP session identifier to correctly associate the Latching packet with the correct context. If an RTP packet is to be used, there would be a benefit to using a well-defined RTP payload format for this purpose as the No-Op payload format proposed. However, in the absence of such a specification, an RTP packet without a payload could be used. Using SSRC is beneficial because RTP and RTCP both would work as is. However, other packet formats could be used that carry the necessary identification of the context, and such a solution is discussed in .

An RTSP ALG not supporting this method could interfere with the methods used to indicate that Latching is to be done, as well as the SSRC signaling, thus preventing the method from working. However, if the RTSP ALG instead opens the corresponding pinholes and creates the necessary mapping in the NAT, traversal will still work. Securing the RTSP message transport using TLS will avoid this issue. An RTSP ALG that supports this traversal method can for basic functionality simply pass the related signaling parameters transparently. Due to the security considerations for Latching, there might exist a benefit for an RTSP ALG that will enable NAT traversal to negotiate with the path and turn off the Latching procedures when the ALG handles this. However, this opens up to failure modes when there are multiple levels of NAT and only one supports an RTSP ALG.

Advantages: Works for all types of client-facing NATs (requirement 1 in ). Has little interaction problems with any RTSP ALG changing the client's information in the Transport header. Disadvantages: Requires modifications to both the RTSP server and client. Limited to working with servers that are not behind a NAT. The format of the packet for "connection setup" (a.k.a Latching packet) is not defined. SSRC management if RTP is used for Latching due to risk for mis-association of clients to RTSP sessions at the server if SSRC collision occurs. Has significant security considerations (See ), due to the lack of a strong authentication mechanism and will need to use address restrictions.

Latching's major security issue is that RTP streams can be hijacked and directed towards any target that the attacker desires unless address restrictions are used. In the case of NATs with multiple clients on the inside of them, hijacking can still occur. This becomes a significant threat in the context of CGNs. The most serious security problem is the deliberate attack with the use of an RTSP client and Latching. The attacker uses RTSP to set up a media session. Then it uses Latching with a spoofed source address of the intended target of the attack. There is no defense against this attack other than restricting the possible address a Latching packet can come from to the same address as the RTSP TCP connection is from. This prevents Latching to be used in use cases that require different addresses for media destination and signaling. Even allowing only a limited address range containing the signaling address from where Latching is allowed opens up a significant vulnerability as it is difficult to determine the address usage for the network the client connects from. A hijack attack can also be performed in various ways. The basic attack is based on the ability to read the RTSP signaling packets in order to learn the address and port the server will send from and also the SSRC the client will use. Having this information, the attacker can send its own Latching packets containing the correct RTP SSRC to the correct address and port on the server. The RTSP server will then use the source IP and Port from the Latching packet as the destination for the media packets it sends. Another variation of this attack is for a man in the middle to modify the RTP Latching packet being sent by a client to the server by simply changing the source IP and Port to the target one desires to attack. One can fend off the snooping-based attack by applying encryption to the RTSP signaling transport. However, if the attacker is a man in the middle modifying Latching packets, the attack is impossible to defend against other than through address restrictions. As a NAT rewrites the source IP and (possibly) port, this cannot be authenticated, but authentication is required in order to protect against this type of DoS attack. Yet another issue is that these attacks also can be used to deny the client the service it desires from the RTSP server completely. The attacker modifies or originates its own Latching packets with a port other than what the legit Latching packets use, which results in the media server sending the RTP/RTCP traffic to ports the client isn't listening for RTP/RTCP on. The amount of random non-guessable material in the Latching packet determines how well Latching can fend off stream hijacking performed by parties that are off the client-to-server network path, i.e., it lacks the capability to see the client's Latching packets. The proposal above uses the 32-bit RTP SSRC field to this effect. Therefore, it is important that this field is derived with a non-predictable random number generator. It should not be possible by knowing the algorithm used and a couple of basic facts to derive what random number a certain client will use. An attacker not knowing the SSRC but aware of which port numbers that a server sends from can deploy a brute-force attack on the server by testing a lot of different SSRCs until it finds a matching one. Therefore, a server could implement functionality that blocks packets to ports or from sources that receive or send multiple Latching packets with different invalid SSRCs, especially when they are coming from the same IP and Port. Note that this mitigation in itself opens up a new venue for DoS attacks against legit users trying to latch. To improve the security against attackers, the amount of random material could be increased. To achieve a longer random tag while still using RTP and RTCP, it will be necessary to develop RTP and RTCP payload formats for carrying the random material.

Latching as described above requires the usage of a valid RTP format as the Latching packet, i.e., the first packet that the client sends to the server to establish a bidirectional transport flow for RTP streams. There is currently no appropriate RTP packet format for this purpose, although the RTP No-Op format was a proposal to fix the problem ; however, that work was abandoned. discusses the implication of different types of packets as keep-alives for RTP, and its findings are very relevant to the format of the Latching packet. Meanwhile, there have been NAT/firewall traversal techniques deployed in the wireless streaming market place that use non-RTP messages as Latching packets. This section describes a variant based on a subset of those solutions that alters the previously described Latching solution.

In this variation of Latching, the Latching packet is a small UDP packet that does not contain an RTP header. In response to the client's Latching packet, the RTSP server sends back a similar Latching packet as a confirmation so the client can stop the so-called "connection phase" of this NAT traversal technique. Afterwards, the client only has to periodically send Latching packets as keep-alive messages for the NAT mappings. The server listens on its RTP-media output port and tries to decode any received UDP packet as the Latching packet. This is valid since an RTSP server is not expecting RTP traffic from the RTSP client. Then, it can correlate the Latching packet with the RTSP client's session ID or the client's SSRC and record the NAT bindings accordingly. The server then sends a Latching packet as the response to the client. The Latching packet can contain the SSRC to identify the RTP stream, and care must be taken if the packet is bigger than 12 bytes, ensuring that it is distinctively different from RTP packets, whose header size is 12 bytes. RTSP signaling can be added to do the following: Enable or disable such Latching message exchanges. When the firewall/NAT has an RTSP-aware ALG, it is possible to disable Latching message exchange and let the ALG work out the address and port mappings. Configure the number of retries and the retry interval of the Latching message exchanges.

See Latching ALG considerations in .

This approach has the following advantages when compared with the Latching approach: There is no need to define an RTP payload format for firewall traversal; therefore, it is more simple to use, implement, and administer (requirement 4 in ) than a Latching protocol, which must be defined. When properly defined, this kind of Latching packet exchange can also authenticate RTP receivers, to prevent hijacking attacks. This approach has the following disadvantage when compared with the Latching approach: The server's sender SSRC for the RTP stream or other session Identity information must be signaled in the RTSP's SETUP response, in the Transport header of the RTSP SETUP response.

Compared to the security properties of Latching, this variant is slightly improved. First of all it allows for a larger random field in the Latching packets, which makes it more unlikely for an off-path attacker to succeed in a hijack attack. Second, the confirmation allows the client to know when Latching works and when it doesn't and thus when to restart the Latching process by updating the SSRC. Still, the main security issue remaining is that the RTSP server can't know that the source address in the Latching packet was coming from an RTSP client wanting to receive media and not from one that likes to direct the media traffic to a DoS target.

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.

Uses the same RTSP extensions as the Alternative Latching method 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, 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.

See Latching ALG considerations in .

A solution with a three-way handshake and its own Latching packets can be compared with the ICE-based solution and have the following differences: Only works for servers that are not behind a NAT. 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.

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 ). Three-Way Latching is really only vulnerable to an on-path attacker that is quite capable. First, the attacker can learn the client-to&nbhy;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.

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: 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. 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.

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. Detect any SETUP request. 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. 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. 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. 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. 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. The ALG may remove a mapping as soon as a TEARDOWN response has been received for that media stream.

Advantages: No impact on either client or server. Can work for any type of NATs. Disadvantages: When deployed, they are hard to update to reflect protocol modifications and extensions. If not updated, they will break the functionality. When end-to-end security is used, the ALG functionality will fail. 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.

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. 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.

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.

The RTSP core specification supports interleaving of media data on the TCP connection that carries RTSP signaling. See Section 14 in 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 . For signaling and rules on how to establish the TCP connection in lieu of UDP, see Appendix C.2 in . This is based on the framing of RTP over the TCP connection as described in .

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.

Advantage: Works through all types of NATs where the RTSP server is not NATed or is at least reachable like it was not. Disadvantages: Functionality needs to be implemented on both server and client. 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.

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.

TURN 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.

To use a TURN server for NAT traversal, the following steps should be performed. 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. 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. The RTSP client uses the acquired address and port allocations in the RTSP SETUP request using the destination header. 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. The RTSP client uses the RTSP server's response to create TURN permissions for the server's media traffic. The client requests that the server starts playing. The server starts sending media packets to the given destination address and ports. 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. 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.

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.

Advantages: Does not require any server modifications given that the server includes the "src_addr" header in the SETUP response. Works for any type of NAT as long as the RTSP server has a reachable IP address that is not behind a NAT. Disadvantages: Requires another network element, namely the TURN server. 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. The TURN server becomes a single point of failure. Since TURN forwards media packets, as a necessity it introduces delay. 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 . However, the usage of TURN could be reduced when the demand for having NAT traversal is reduced.

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.