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Network Working Group                                  Magnus Westerlund
INTERNET-DRAFT                                                  Ericsson
Expires: April 2006                                          Thomas Zeng
                                           PacketVideo Network Solutions
                                                        October 24, 2005





     How to Enable Real-Time Streaming Protocol (RTSP) Traverse Network
           Address Translators (NAT) and Interact with Firewalls.
                    <draft-ietf-mmusic-rtsp-nat-04.txt>


Status of this memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

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

   Internet-Drafts are draft documents valid for a maximum of six
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   at any time.  It is inappropriate to use Internet-Drafts as
   reference material or to cite them other than as "work in
   progress.".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/1id-abstracts.html

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

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

Abstract

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




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TABLE OF CONTENTS

1. Definitions.........................................................4
  1.1. Glossary........................................................4
  1.2. Terminology.....................................................4
2. Changes.............................................................4
3. Introduction........................................................5
  3.1. NATs............................................................5
  3.2. Firewalls.......................................................5
4. Requirements........................................................6
5. Detecting the loss of NAT mappings..................................7
6. NAT Traversal Techniques............................................8
  6.1. STUN............................................................8
   6.1.1. Introduction.................................................8
   6.1.2. Using STUN to traverse NAT without server modifications......9
   6.1.3. Embedding STUN in RTSP......................................11
   6.1.4. Discussion On Co-located STUN Server........................12
   6.1.5. ALG considerations..........................................12
   6.1.6. Deployment Considerations...................................12
   6.1.7. Security Considerations.....................................14
  6.2. ICE............................................................14
   6.2.1. Introduction................................................14
   6.2.2. Using ICE in RTSP...........................................15
   6.2.3. Implementation burden of ICE................................17
   6.2.4. Deployment Considerations...................................17
  6.3. Symmetric RTP..................................................18
   6.3.1. Introduction................................................18
   6.3.2. Necessary RTSP extensions...................................18
   6.3.3. Deployment Considerations...................................18
   6.3.4. Security Consideration......................................19
   6.3.5. A Variation to Symmetric RTP................................20
  6.4. Application Level Gateways.....................................21
   6.4.1. Introduction................................................21
   6.4.2. Guidelines On Writing ALGs for RTSP.........................22
   6.4.3. Deployment Considerations...................................23
   6.4.4. Security Considerations.....................................23
  6.5. TCP Tunneling..................................................23
   6.5.1. Introduction................................................23
   6.5.2. Usage of TCP tunneling in RTSP..............................24
   6.5.3. Deployment Considerations...................................24
   6.5.4. Security Considerations.....................................24
  6.6. TURN (Traversal Using Relay NAT)...............................25
   6.6.1. Introduction................................................25
   6.6.2. Usage of TURN with RTSP.....................................25
   6.6.3. Deployment Considerations...................................26



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   6.6.4. Security Considerations.....................................27
7. Firewalls..........................................................28
8. Comparison of Different NAT Traversal Techniques...................28
9. Open Issues........................................................29
10. Security Consideration............................................29
11. IANA Consideration................................................30
12. Acknowledgments...................................................30
13. Author's Addresses................................................30
14. References........................................................31
15. IPR Notice........................................................33
16. Copyright Notice........................Error! Bookmark not defined.









































Westerlund, Zeng                                                [Page 3]


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1. Definitions

1.1. Glossary

   ALG    - Application Level Gateway, an entity that can be embedded
            in a NAT or other middlebox to perform the application layer
            functions required for a particular protocol to traverse the
            NAT/middlebox [6]
   ICE    - Interactive Connectivity Establishment, see [9].
   DNS    - Domain Name Service
   DDOS   - Distributed Denial Of Service attacks
   MID    - Media Identifier from Grouping of media lines in SDP, see
            [10].
   NAT    - Network Address Translator, see [12].
   NAT-PT - Network Address Translator Protocol Translator, see [13]
   RTP    - Real-time Transport Protocol, see [5].
   RTSP   - Real-Time Streaming Protocol, see [1] and [7].
   SDP    - Session Description Protocol, see [2].
   SSRC   - Synchronization source in RTP, see [5].
   TBD    - To Be Decided


1.2. Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [4].


2. Changes

   The following changes have been done since draft-ietf-mmusic-rtsp-
   nat-03.txt:

   - A outline of a procedure for ICE is presneted.
   - Updated references
   - Replaced NAT classification by BEHAVE WG definitions.
















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3. Introduction

   Today there is a proliferate deployment of different flavors of
   Network Address Translator (NAT) boxes that in practice follow
   standards rather loosely [12][24][18].  NATs cause discontinuity in
   address realms [18], therefore a protocol, such as RTSP, needs to
   try to make sure that it can deal with such discontinuities caused
   by NATs. The problem with RTSP is that, being a media control
   protocol that manages one or more media streams; RTSP carries
   information about network addresses and ports inside itself. Because
   of this, even if RTSP itself, when carried over TCP for example, is
   not blocked by NATs, its media streams may be blocked by NATs,
   unless special provisions are added to support NAT-traversal.

   Like NATs, firewalls (FWs) are also middle boxes that need to be
   considered. They are deployed to prevent unwanted traffic to be able
   to get in or out of the protected network. RTSP is designed such
   that a firewall can be configured to let RTSP controlled media
   streams to go through with minimal implementation problems. However
   there is a need for more detailed information on how FWs should be
   configured to work with RTSP.

   This document describes several NAT-traversal mechanisms for RTSP
   based streaming. These NAT solutions fall into the category of
   ""UNilateral Self-Address Fixing (UNSAF)" as defined in [18] and
   quoted below:
          "UNSAF is a process whereby some originating process attempts
          to determine or fix the address (and port) by which it is
          known - e.g. to be able to use address data in the protocol
          exchange, or to advertise a public address from which it will
          receive connections."

   Following the guidelines spelled out in [18], we describe the
   required RTSP protocol extensions for each method, transition
   strategies, and security concerns.

   This document intends to recommend FW/NAT traversal methods for RTSP
   streaming servers based on RFC 2326 [1] as well as the updated RTSP
   core spec [7]. This document is intended to be updated to stay
   consistent with the RTSP core protocol [7].


3.1. NATs

   Today there exist a number of different NAT types and usage areas.
   These are described in the section 3 and 4 of [26].
3.2. Firewalls

   A firewall (FW) is a security gateway that enforces certain access
   control policies between two network administrative domains: a
   private domain (intranet) and a public domain (public internet).



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

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

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


4. Requirements

   This section considers the set of requirements when designing or
   evaluating RTSP NAT traversal solutions.

   RTSP is a client/server protocol, and as such the targeted
   applications in general deploy RTSP servers in the public address
   realm. However, there are use cases where the reverse is true: RTSP
   clients are connecting from public address realm to RTSP servers
   behind home NATs. This is the case for instance when home
   surveillance cameras running as RTSP servers intend to stream video
   to cell phone users in the public address realm through a home NAT.

   The first priority should be to solve the RTSP NAT traversal problem
   for RTSP servers deployed in the open.

   The list of feature requirements for RTSP NAT solutions are given
   below:
     1. MUST work for all flavors of NATs, including symmetric NATs.
     2. MUST work for firewalls (subject to pertinent firewall
        administrative policies), including those with ALGs.
     3. SHOULD have minimal impact on clients in the open and not dual-
        hosted:
          o For instance, no extra delay from RTSP connection till
             arrival of media.




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     4. SHOULD be simple to use/implement/administer that people
        actually turn them on
          o Otherwise people will resort to TCP tunneling through NATs
          o Address discovery for NAT traversal should take place
             behind the scene, if possible

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

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

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


5. Detecting the loss of NAT mappings

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

   First, since NATs may also dynamically reclaim or readjust
   address/port translations, "keep-alive" and periodic re-polling may
   be required [18]. Secondly, it is possible to detect and recover
   from the situation where the mapping has been changed or removed.



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   The possibility to detect a lost mapping is based on the fact that
   no traffic will arrive. Below we will give some recommendation on
   how to detect loss of NAT mappings when using RTP/RTCP under RTSP
   control.

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

   The loss of mapping for RTCP is simpler to detect. As RTCP is
   normally sent periodically in each direction, even during the RTSP
   ready state, if RTCP packets are missing for several RTCP intervals,
   the mapping is likely to be lost.  Note that if no RTCP packets are
   received by the RTSP server and nor RTSP messages for a while, the
   RTSP server has the option to delete the corresponding SSRC and RTSP
   session ID, because either the client can not get through a middle
   box NAT/FW, or that the client is mal-functioning.


6. NAT Traversal Techniques

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

   Not all of the techniques are yet described in the full details,
   because the intention is to refer to other documents, or some
   appendix to this document, for the full specification of a specific
   NAT traversal solution.  Note that some of the solutions make use of
   protocols (e.g., RTP-NOOP, TURN and ICE) in different stage of
   standardization and not yet completed.


6.1. STUN

6.1.1. Introduction

   STUN - "Simple Traversal of UDP Through Network Address Translators"
   [6][25] is a standardized protocol developed by the MIDCOM WG that
   allows a client to use secure means to discover the presence of a
   NAT between himself and the STUN server and the type of that NAT.



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   The client then uses the STUN server to discover the address
   bindings assigned by the NAT.

   STUN is a client-server protocol. STUN client sends a request to a
   STUN server and the server returns a response. There are two types
   of STUN requests - Binding Requests, sent over UDP, and Shared
   Secret Requests, sent over TLS over TCP.


6.1.2. Using STUN to traverse NAT without server modifications

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

   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 performing
     Endpoint Independent or Address Dependent Mappings. Clients behind
     NATs that do Address and Port Dependent Mappings cannot use this
     method.

   Method:

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

   1. Use STUN to try to discover the type of NAT, and the timeout
      period for any UDP mapping on the NAT. This is RECOMMENDED 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.

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

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





Westerlund, Zeng                                                [Page 9]


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   4. Perform the RTSP SETUP for each media. In the transport header
      the following parameter SHOULD be included with the given values:
      "dest_addr" [7] with the public/external IP address and port pair
      for both RTP and RTCP. To allow this to work servers MUST allow a
      client to setup the RTP stream on any port, not only even ports.
      This requires the new feature provided in the update to RFC2326
      ([7]). The server SHOULD respond with a transport header
      containing an "src_addr" parameter with the RTP and RTCP source
      IP address and port of the media stream.

   5. To keep the mappings alive, the client SHOULD periodically send
      UDP traffic over all mappings needed for the session. STUN MAY be
      used to determine the timeout period of the NAT(s) UDP mappings.
      For the mapping carrying RTCP traffic the periodic RTCP traffic
      may be enough. For mappings carrying RTP traffic and for mappings
      carrying RTCP packets at too low a frequency, keep-alive messages
      SHOULD be sent. As keep alive messages, one could use the RTP
      NOOP packet ([23]) to the streaming server's discard port (port
      number 9). The drawback of using RTP NOOP is that the payload
      type number must be dynamically assigned through RTSP first.

   If a UDP mapping is lost then 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 likely cause a
   glitch in media playback.

   To allow UDP packets to arrive from the server to a client behind a
   Address Dependent Filtering NAT, the client must send the very first
   UDP packet to pinch a hole in the NAT. The client, before sending a
   RTSP PLAY request, must send a so called FW packet (such as a RTP
   NOOP packet) on each mapping, to the IP address given as the servers
   source address. To create minimum problems for the server these UDP
   packets SHOULD be sent to the server's discard port (port number 9).
   Since UDP packets are inherently unreliable, to ensure that at least
   one UDP message passes the NAT, FW packets should be retransmitted
   in short intervals.

   For a Address and Port Dependent Filtering NAT the client must send
   messages to the exact ports used by the server to send UDP packets
   before sending a RTSP PLAY request. This makes it possible to use
   the above described process with the following additional
   restrictions: for each port mapping, FW packets need to be sent
   first to the server's source address/port. To minimize potential
   effects on the server from these messages the following type of FW
   packets MUST be sent. RTP: an empty or less than 12 bytes UDP
   packet.  RTCP: A correctly formatted RTCP RR or SR message.

   The above described adaptations for restricted NATs will not work
   unless the server includes the "src_addr" in the "Transport" header
   (which is the "source" transport parameter in RFC2326).




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6.1.3. Embedding STUN in RTSP

   This section outlines the adaptation and embedding of STUN within
   RTSP. This enables STUN to be used to traverse any type of NAT,
   including symmetric NATs.  Protocol changes are beyond the scope of
   this memo and are instead defined in TBD internet draft.

   Limitations:

   This NAT traversal solution has limitations:

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

   Deviations from STUN as defined in RFC 3489

   Specifically, we differ from RFC3489 in two aspects:
      1. We allow RTSP applications to have the option to perform STUN
         "Shared Secret Request" through RTSP, via extension to RTSP;
      2. We require STUN server to be co-located on RTSP server's media
         output ports.

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

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

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

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



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   can be done by using the "server_port" parameter in RFC2326, and the
   "src_addr" parameter in [7]. Both are in RTSP Transport header.

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


6.1.4. Discussion On Co-located STUN Server

   In order to use STUN to traverse port and address dependent mapping
   NATs the STUN server needs to be co-located with the streaming
   server media output ports. This creates a de-multiplexing problem:
   we must be able to differentiate a STUN packet from a media packet.
   This will be done based on heuristics. This works fine between STUN
   and RTP or RTCP where the first byte happens to be different, but
   may not work with other media transport protocols.


6.1.5. ALG considerations

   If a NAT supports RTSP ALG (Application Level Gateway) and is not
   aware of the STUN traversal option, service failure may happen,
   because a client discovers its public IP address and port numbers,
   and inserts them in its SETUP requests, when the RTSP ALG processes
   the SETUP request it may change the destination and port number,
   resulting in unpredictable behavior. In such cases a convenient way
   should be provided to turn off STUN-based NAT traversal.


6.1.6. Deployment Considerations

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

   Advantages:

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

   Disadvantages:

   - Requires a STUN server deployed in the public address space.
   - Only works with endpoint independent and address dependent
     mapping. Port and address dependent filtering NATs create some
     issues.
   - Does not work with port and address dependent mapping NATs without
     server modifications.
   - Will mostly not work if a NAT uses multiple IP addresses, since
     RTSP server generally requires all media streams to use the same
     IP as used in the RTSP connection.




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   - Interaction problems exist when a RTSP-aware ALG interferes with
     the use of STUN for NAT traversal.
   - 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
     RFC2326.

   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.


   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, RTSP application can re-use STUN as a
     NAT traversal solution, avoiding the pit-fall of solving a problem
     twice.
   - STUN has built-in message authentication features, which makes it
     more secure. See next section for an in-depth security discussion.
   - This solution works as long as there is only one RTSP end point in
     the private address realm, regardless of the NAT's type. There may
     even be multiple NATs (see figure 1 in [6]).
   - Compares to other UDP based NAT traversal methods in this
     document, STUN requires little new protocol development (since
     STUN is already a IETF standard), and most likely less
     implementation effort, since open source STUN server and client
     have become available [21]. There is the need to embed STUN in
     RTSP server and client, which require a de-multiplexer between
     STUN packets and RTP/RTCP packets. There is also a need to
     register the proper feature tags.

   Disadvantages:

   - Some extensions to the RTSP core protocol, signaled by RTSP
     feature tags, must be introduced.
   - Requires an embedded STUN server to co-locate on each of RTSP
     server's media protocol's ports (e.g. RTP and RTCP ports), which
     means more processing is required to de-multiplex STUN packets
     from media packets. For example, the de-multiplexer must be able
     to differentiate a RTCP RR packet from a STUN packet, and forward
     the former to the streaming server, the later to STUN server.
   - Even if the RTSP server is in the open, and the client is behind a
     multi-addressed NAT, it may still break if the RTSP server does



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     not allow RTP packets to be sent to an IP differs from the IP of
     the client's RTSP request.
   - Interaction problems exist when a RTSP ALG is not aware of STUN.
   - Using STUN requires that RTSP servers and clients support the
     updated RTSP specification, and they both agree to support the
     proper feature tag.
   - Increases the setup delay with at least the amount of time it
     takes to perform STUN message exchanges.


   Transition:

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

6.1.7. Security Considerations

   To prevent RTSP server being used as Denial of Service (DoS) attack
   tools the RTSP Transport header parameter "destination" and
   "dest_addr" are generally not allowed to point to any IP address
   other than the one that RTSP message originates from. The RTSP
   server is only prepared to make an exception of this rule when the
   client is trusted (e.g., through the use of a secure authentication
   process, or through some secure method of challenging the
   destination to verify its willingness to accept the RTP traffic).
   Such restriction means that STUN does not work for NATs that would
   assign different IP addresses to different UDP flows on its public
   side. Therefore the multi-addressed NATs will at times have trouble
   with STUN-based RTSP NAT traversals.

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

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


6.2. ICE

6.2.1. Introduction




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   ICE (Interactive Connectivity Establishment) [9] is a methodology
   for NAT traversal that is under development for SIP using SDP
   offer/answer. The basic idea is to try, in a parallel fashion, all
   possible connection addresses that an end point may have. This
   allows the end-point to use the best available UDP "connection"
   (meaning two UDP end-points capable of reaching each other). The
   methodology has very nice properties in that basically all NAT
   topologies are possible to traverse.

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

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

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

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


6.2.2. Using ICE in RTSP

   The usage of ICE for RTSP requires that both client and server be
   updated to include the ICE functionality. If both parties implement
   the necessary functionality the following step-by-step algorithm
   could be used to accomplish connectivity for the UDP traffic.

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



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

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

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

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

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

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

   Step 6a: Connectivity checks from the client's primary to the
   server's primary was successful. Thus no further SETUP requests are
   necessary. Go to 7.

   Step 6b: Connectivity checks for primary fails. If further
   candidates have been derived then those can be promoted in new
   candidate lines in SETUP request updating the list (Goto 5). If
   another address than the primary has been verified by the client to
   work, that address may then be promoted for usage in a SETUP request
   (Goto 7).




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   Step 7: Client issues PLAY request. If the server also has completed
   its connectivity checks for this primary addresses (based on
   username as it may be derived addresses if the client was behind
   NAT) then it can directly answer 200 ok (Goto 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. Client needs to go
   to 6b.

   Step 8: Process completed media can be delivered. ICE testing ports
   may be released.

   To keep media paths alive client must likely periodically send data
   to the server. This could be realized with either STUN or RTP No-op
   [23] packets. RTCP sent by client should be able to keep RTCP open.


6.2.3. Implementation burden of ICE

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

   Some RTSP server implementation requirements are:
    - STUN server features
    - limited STUN client features
    - SDP generation with more parameters.
    - RTSP error code for ICE extension

   Some client implantation requirements are:
    - Limited STUN server features
    - Limited STUN client features
    - RTSP error code and ICE extension

6.2.4. Deployment Considerations

   Advantages:
   - Solves NAT connectivity discovery for basically all cases as long
     as a TCP connection between them can be established. This includes
     servers behind NATs. (Note that a proxy between address domains
     may be required to get TCP through).
   - Improves defenses against DDOS attacks, as 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 de-multiplex between media packets
     and STUN packets.




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   - Has fairly high implementation burden put on both RTSP server and
     client. The precise implantation complexity needs to be assessed
     once ICE is fully defined as a standard. Currently ICE is still a
     protocol under development.

6.3. Symmetric RTP

6.3.1. Introduction

   Symmetric RTP is a NAT traversal solution that is based on requiring
   RTSP clients to send UDP packets to the server's media output ports.
   Conventionally, RTSP servers send RTP packets in one direction: from
   server to client. Symmetric RTP is similar to connection-oriented
   traffic, where one side (e.g., the RTSP client) first "connects" by
   sending a RTP packet to the other side's RTP port, the recipient
   then replies to the originating IP and port.

   Specifically, when the RTSP server receives the "connect" RTP packet
   (a.k.a. FW packet, since it is used to pinch a hole in the FW/NAT
   and to aid the server for port binding and address mapping) from its
   client, it copies the source IP and Port number and uses them as
   delivery address for media packets. By having the server send media
   traffic back the same way as the client's packet are sent to the
   server, address mappings will be honored. Therefore this technique
   works for all types of NATs. However, it does require server
   modifications. Unless there is built-in protection mechanism,
   symmetric RTP is very vulnerable to DDOS attacks, because attackers
   can simply forge the source IP & Port of the binding packet.


6.3.2. Necessary RTSP extensions

   To support symmetric RTP the RTSP signaling must be extended to
   allow the RTSP client to indicate that it will use symmetric RTP.
   The client also needs to be able to signal its RTP SSRC to the
   server in its SETUP request. The RTP SSRC is used to establish some
   basic level of security against hijacking attacks. Care must be
   taken in choosing client's RTP SSRC. First, it must be unique within
   all the RTP sessions belonging to the same RTSP session. Secondly,
   if the RTSP server is sending out media packets to multiple clients
   from the same send port, the RTP SSRC needs to be unique amongst
   those clients' RTP sessions. Recognizing that there is a potential
   that RTP SSRC collision may occur, the RTSP server must be able to
   signal to client that a collision has occurred and that it wants the
   client to use a different RTP SSRC carried in the SETUP response.

   Details of the RTSP extension are beyond the scope of this draft.


6.3.3. Deployment Considerations




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   Advantages:

   - Works for all types of NATs, including those using multiple IP
     addresses. (Requirement 1 in section 4).
   - Have no interaction problems with any RTSP ALG changing the
     client's information in the transport header.

   Disadvantages:

   - Requires modifications to both RTSP server and client.
   - The format of the RTP packet for "connection setup" (a.k.a FW
     packet) is yet to be defined. One possibility is to use RTP NOOP
     packet format in [23].
   - Has worse security situation than STUN when using address
     restrictions.
   - Would still require STUN to discover the timeout of NAT bindings.


6.3.4. Security Consideration

   Symmetric RTP's major security issue is that RTP streams can be
   hijacked and directed towards any target that the attacker desires.

   The most serious security problem is the deliberate attack with the
   use of a RTSP client and symmetric RTP. The attacker uses RTSP to
   setup a media session. Then it uses symmetric RTP with a spoofed
   source address of the intended target of the attack. There is no
   defense against this attack other than restricting the possible bind
   address to be the same as the RTSP connection arrived on. This
   prevents symmetric RTP to be used with multi-address NATs.

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

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

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





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

   An attacker not knowing the SSRC but aware of which port numbers
   that a server sends from can deploy a brute force attack on the
   server by testing a lot of different SSRCs until it finds a matching
   one. Therefore a server SHOULD implement functionality that blocks
   ports that receive multiple FW packets (i.e. the packet that is sent
   to the server for FW traversal) with different invalid SSRCs,
   especially when they are coming from the same IP/Port.

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


6.3.5. A Variation to Symmetric RTP

   Symmetric RTP requires a valid RTP format in the FW packet, which is
   the first packet that the client sends to the server to set up
   virtual RTP connection. There is currently no appropriate RTP packet
   format for this purpose, although the NOOP format is a proposal to
   fix the problem [23].

   Meanwhile, there has been FW traversal techniques deployed in the
   wireless streaming market place that use non-RTP messages as FW
   packets. This section attempts to summarize a subset of those
   solutions that happens to use a variation to the standard symmetric
   RTP solution.

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

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




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   server's SSRC, and record the NAT bindings accordingly. The server
   then sends a FW packet as the response to the client.

   The FW packet normally contains the SSRC used to identify the RTP
   stream, and can be made no bigger than 12 bytes, making it
   distinctively different from RTP packets, whose header size is 12
   bytes.

   RTSP signaling can be added to do the following:
      1. Enables or disables such FW message exchanges. When the FW/NAT
        has an RTSP-aware ALG, it is better to disable FW message
        exchange and let ALG works out the address and port mappings.
      2. Configures the number of re-tries and the re-try interval of
        the FW message exchanges.

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

   This approach has the following advantages when compared with the
   symmetric RTP approach:
      1. There is no need to define RTP payload format for FW traversal,
        therefore it is simple to use, implement and administer
        (Requirement 4 in section 4), although a binding protocol must
        be defined (which is out side of the scope of this memo).
      2. When properly defined, this kind of FW message exchange can
        also authenticate RTP receivers, so as to prevent DDoS attacks
        for dual-hosted RTSP client. By dual-hosted RTSP client we mean
        the kind that uses one "perceived" IP address for RTSP message
        exchange, and a different "perceived" IP address for RTP
        reception. (Requirement 5 in section 4).

   This approach has the following disadvantages when compared with the
   symmetric RTP approach:
      1. RTP traffic is normally accompanied by RTCP traffic. This
        approach still needs to rely on RTCP RRs and SRs to enable NAT
        traversal for RTCP endpoints, or use the same type of FW
        messages for RTCP endpoints.
      2. The server's sender SSRC for the RTP stream must be signaled in
        RTSP's SETUP response, in the Transport header of the RTSP
        SETUP response.


6.4. Application Level Gateways

6.4.1. Introduction

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



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

   2. ALGs need to be updated if extensions to the protocol are added.
   Due to deployment issues with changing 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 home or SOHO (small office/home office) environment.


6.4.2. Guidelines On Writing ALGs for RTSP


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

   1. Detect any SETUP request.

   2. Try to detect the usage of any of the NAT traversal methods that
      replace the address and port of the Transport header parameters
      "destination" or "dest_addr". If any of these methods are used,
      the ALG SHOULD NOT change the address. Ways to detect that these
      methods are used are:
      - For embedded STUN, it would be watch for a feature tag, like
      "nat.stun". If any of those exists in the "supported", "proxy-
      require", or "require" headers of the RTSP exchange.
      - For non-embedded STUN and TURN based solutions: This can in
      some case be detected by inspecting the "destination" or
      "dest_addr" parameter. If it contains either one of the NAT's
      external IP addresses or a public IP address. However if multiple
      NATs are used this detection may fail. Remapping should only be
      done for addresses belonging to the NATs own private address
      space.

      Otherwise continue to the next step.

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





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

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

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


6.4.3. Deployment Considerations

   Advantage:

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

   Disadvantage:

   - 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 type of traversal mechanisms, such as
     STUN.

   Transition:

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


6.4.4. Security Considerations

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


6.5. TCP Tunneling

6.5.1. Introduction



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   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 RTP packets.

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


6.5.2. Usage of TCP tunneling in RTSP

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

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


6.5.3. Deployment Considerations

   Advantage:

   - Works through all types of NATs where server is in the open.

   Disadvantage:

   - 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 price of loss of
   real-time properties.


6.5.4. Security Considerations

   The TCP tunneling of RTP has no known security problem besides those
   already present in RTSP. It is not possible to get any amplification



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   effect that is desired for denial of service attacks due to TCP's
   flow control.

   A possible security consideration, when session media data is
   interleaved with RTSP, would be the performance bottleneck when RTSP
   encryption is applied, since all session media data also needs to be
   encrypted.


6.6. TURN (Traversal Using Relay NAT)

6.6.1. Introduction

   Traversal Using Relay NAT (TURN) [8] is a protocol for setting up
   traffic relays that allows clients behind NATs and firewalls to
   receive incoming traffic for both UDP and TCP.  These relays are
   controlled and have limited resources. They need to be allocated
   before usage.

   TURN allows a client to temporarily bind an address/port pair on the
   relay (TURN server) to its local source address/port pair, which is
   used to contact the TURN server. The TURN server will then forward
   packets between the two sides of the relay. To prevent DOS attacks
   on either recipient, the packets forwarded are restricted to the
   specific source address. On the client side it is restricted to the
   source setting up the mapping. On the external side this is limited
   to the source address/port pair of the first packet arriving on the
   binding. After the first packet has arrived the mapping is "locked
   down" to that address. Packets from any other source on this address
   will be discarded.

   Using a TURN server makes it possible for a RTSP client to receive
   media streams from even an unmodified RTSP server. However the
   problem is those RTSP servers most likely restrict media
   destinations to no other IP address than the one RTSP message
   arrives. This means that TURN could only be used if the server knows
   and accepts that the IP belongs to a TURN server and the TURN server
   can't be targeted at an unknown address. Unfortunately TURN servers
   can be targeted at any host that has a public IP address by spoofing
   the source IP of TURN Allocation requests.


6.6.2. 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 RTSP server. The client retrieves
      the session description to determine the number of media streams.
      To avoid the issue with having RTSP connection and media traffic




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      from different addresses also the TCP connection must be done
      thru the same TURN server as the one in the next step.

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

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

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

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

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

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

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


6.6.3. Deployment Considerations

   Advantages:

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

   Disadvantage

   - TURN is not yet a standard.
   - Requires another network element, namely the TURN server.





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

   Transition:

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


6.6.4. Security Considerations

   An eavesdropper of RTSP messages between the RTSP client and RTSP
   server will be able to do a simple denial of service attack on the
   media streams by sending messages to the destination address and
   port present in the RTSP SETUP messages. If the attacker's message
   can reach the TURN server before the RTSP server's message, the lock
   down can be accomplished towards some other address. This will
   result in that the TURN server will drop all the media server's
   packets when they arrive. This can be accomplished with little risk
   for the attacker of being caught, as it can be performed with a
   spoofed source IP. The client may detect this attack when it
   receives the lock down packet sent by the attacker as being mal-
   formatted and not corresponding to the expected context. It will
   also notice the lack of incoming packets. See bullet 7 in section
   6.6.2.

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






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

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


7. Firewalls

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

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

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


8. Comparison of Different NAT Traversal Techniques

   This section evaluates the techniques described above against the
   requirements listed in section 4.

   In the following table, the columns correspond to the numbered
   requirements. For instance, the column under R1 corresponds to the
   first requirement in section 4: MUST work for all flavors of NATs.

   The rows represent the different FW traversal techniques. SymRTP is
   short for symmetric RTP, "V.SymRTP" is short for "variation of
   symmetric RTP" as described in section 6.3.5.



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   -----------------------------------------------+
               |  R1  |  R2  |  R3  |  R4  |  R5  |
   ------------+------+------+------+------+------+
     STUN      | Yes  | Yes  |  No  | Maybe|  No  |
   ------------+------+------+------+------+------+
     ICE       | Yes  | Yes  |  No  |  No  | Yes  |
   ------------+------+------+------+------+------+
     SymRTP    | Yes  | Yes  | Yes  |Maybe |  No  |
   ------------+------+------+------+------+------+
   V. SymRTP   | Yes  | Yes  | Yes  | Yes  |future|
   ------------+------+------+------+------+------+
     TURN      | Yes  | Yes  | No   | No   | Yes  |
   -----------------------------------------------+


9. Open Issues

   Some open issues with this draft:

   - At some point we need to recommend one RTSP NAT solution so as to
     ensure implementations can inter-operate. This decision will
     require that requirements, security and desired goals be evaluated
     against implementation cost and the probability to get the final
     solution deployed.
   - The ALG recommendations need to be improved and clarified.
   - The firewall RTSP ALG recommendations need to be written as they
     are different from the NAT ALG in some perspectives.
   - The ICE solution needs to be hammered out into all the details.


10. Security Consideration

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

   Each of the proposed solutions has security implications.

   Using STUN will provide the same level of security as RTSP with out
   transport level security and source authentications; as long as the
   server does not grant a client request to send media to different IP
   addresses.

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



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   solved in the variation of symmetric RTP (section 6.3.5) with
   authentication features.

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

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

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


11. IANA Consideration

   This specification does not define any protocol extensions hence no
   IANA action is requested.


12. Acknowledgments

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


13. Author's Addresses

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


   Thomas Zeng                     Tel: 1-858-320-3125
   PacketVideo Network Solutions   Email: zeng@pvnetsolutions.com
   9605 Scranton Rd., Suite 400
   San Diego, CA92121





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14. References

14.1. Normative references

   [1]  H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
        IETF RFC 2326, April 1998.
   [2]  M. Handley, V. Jacobson, "Session Description Protocol (SDP)",
        IETF RFC 2327, April 1998.
   [3]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", RFC 4234, October 2005.
   [4]  S. Bradner, "Key words for use in RFCs to Indicate Requirement
        Levels", RFC 2119, March 1997.
   [5]  H. Schulzrinne, et. al., "RTP: A Transport Protocol for Real-
        Time Applications", STD 64, RFC 3550, IETF, July 2003.
   [6]  J. Rosenberg, et. Al., " STUN - Simple Traversal of UDP Through
        Network Address Translators", IETF RFC 3489, March 2003
   [7]  H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
        draft-ietf-mmusic-rfc2326bis-11.txt, IETF draft, October 2005,
        work in progress.
   [8]  J. Rosenberg, et. Al., "Traversal Using Relay NAT (TURN)",
        draft-rosenberg-midcom-turn-08.txt, IETF draft, Sep 2005, work
        in progress.
   [9]  J. Rosenberg, "Interactive Connectivity Establishment (ICE): A
        Methodology for Network Address Translator (NAT) Traversal for
        the Session Initiation Protocol (SIP)," draft-ietf-mmusic-ice-
        06, IETF draft, October 2005, work in progress.
   [10] G. Camarillo, et. al., "Grouping of Media Lines in the Session
        Description Protocol (SDP)," IETF RFC 3388, December 2002.
   [11] Camarillo, G. and J. Rosenberg, "The Alternative Network
        Address Types (ANAT) Semantics for the Session Description
        Protocol (SDP) Grouping Framework", RFC 4091, June 2005.

14.2. Informative References

   [12] P. Srisuresh, K. Egevang, "Traditional IP Network Address
        Translator (Traditional NAT)," RFC 3022, Internet Engineering
        Task Force, January 2001.
   [13] Tsirtsis, G. and Srisuresh, P., "Network Address Translation -
        Protocol Translation (NAT-PT)", RFC 2766, Internet Engineering
        Task Force, February 2000.
   [14] S. Deering and R. Hinden, "Internet Protocol, Version 6 (IPv6)
        Specification", RFC 2460, Internet Engineering Task Force,
        December 1998.
   [15] J. Postel, "internet protocol", RFC 791, Internet Engineering
        Task Force, September 1981.




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   [16] Camarillo, G. and J. Rosenberg, "The Alternative Network
        Address Types (ANAT) Semantics for the Session Description
        Protocol (SDP) Grouping Framework", RFC 4091, June 2005.
   [17] John Lazzaro, "Framing RTP and RTCP Packets over Connection-
        Oriented Transport", IETF Draft, draft-ietf-avt-rtp-framing-
        contrans-06.txt, September 2005.
   [18] D. Daigle, "IAB Considerations for UNilateral Self-Address
        Fixing (UNSAF) Across Network Address Translation", RFC 3424,
        Internet Engineering Task Force, Nov. 2002
   [19] R. Finlayason, "IP Multicast and Firewalls", RFC 2588, Internet
        Engineering Task Force, May 1999
   [20] Krawczyk, H., Bellare, M., and Canetti, R.: "HMAC: Keyed-
        hashing for message authentication". IETF RFC 2104, February
        1997
   [21] Open Source STUN Server and Client,
        http://www.vovida.org/applications/downloads/stun/index.html
   [22]
   [23] Dan Wing, et.al. "RTP No-Op Payload Format", draft-wing-avt-
        rtp-noop-00.txt, March 2004
   [24] P. Srisuresh and M.Holdrege, "IP Network Address Translator
        (NAT) Terminology and Considerations", RFC2663, Internet
        Engineering Task Force, Aug. 1999
   [25] J. Rosenberg, C. Huitema and R. Mahy, "STUN - Simple Traversal
        of UDP Through Network Address Translators", draft-ietf-behave-
        rfc3489bis-02.txt, July 2005
   [26] F. Audet, "NAT Behavioral Requirements for Unicast UDP," draft-
        ietf-behave-nat-udp-04, September 6, 2005.























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15. IPR Notice

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16. Copyright Notice

   Copyright (C) The Internet Society (2005).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

   This document and the information contained herein are provided on
   an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
   INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
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   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


   This Internet-Draft expires in April 2006.











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