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Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                                   T. Zeng
Expires: September 15, 2011                               March 14, 2011


 The Evaluation of Different Network Addres Translator (NAT) Traversal
 Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
                draft-ietf-mmusic-rtsp-nat-evaluation-03

Abstract

   This document describes several Network Address Translator (NAT)
   traversal techniques that was considered to be used by Real-time
   Streaming Protocol (RTSP).  Each technique includes a description on
   how it would be used, the security implications of using it and any
   other deployment considerations it has.  There are also disussions on
   how NAT traversal techniques relates to firewalls and how each
   technique can be applied in different use cases.  These findings
   where used when selecting the NAT traversal for RTSP 2.0 standardized
   in the MMUSIC WG.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

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

   This Internet-Draft will expire on September 15, 2011.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
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   the Trust Legal Provisions and are provided without warranty as
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   This document may contain material from IETF Documents or IETF
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   it for publication as an RFC or to translate it into languages other
   than English.


































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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  5
     1.1.  Network Address Translators  . . . . . . . . . . . . . . .  6
     1.2.  Firewalls  . . . . . . . . . . . . . . . . . . . . . . . .  7
     1.3.  Glossary . . . . . . . . . . . . . . . . . . . . . . . . .  7
     1.4.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  8
   2.  Detecting the loss of NAT mappings . . . . . . . . . . . . . .  8
   3.  Requirements on NAT-Traversal  . . . . . . . . . . . . . . . .  9
   4.  NAT Traversal Techniques . . . . . . . . . . . . . . . . . . . 10
     4.1.  STUN . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
       4.1.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 11
       4.1.2.  Using STUN to traverse NAT without server
               modifications  . . . . . . . . . . . . . . . . . . . . 11
       4.1.3.  Embedding STUN in RTSP . . . . . . . . . . . . . . . . 13
       4.1.4.  Discussion On Co-located STUN Server . . . . . . . . . 14
       4.1.5.  ALG considerations . . . . . . . . . . . . . . . . . . 15
       4.1.6.  Deployment Considerations  . . . . . . . . . . . . . . 15
       4.1.7.  Security Considerations  . . . . . . . . . . . . . . . 17
     4.2.  ICE  . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
       4.2.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 18
       4.2.2.  Using ICE in RTSP  . . . . . . . . . . . . . . . . . . 19
       4.2.3.  Implementation burden of ICE . . . . . . . . . . . . . 20
       4.2.4.  Deployment Considerations  . . . . . . . . . . . . . . 21
       4.2.5.  Security Consideration . . . . . . . . . . . . . . . . 21
     4.3.  Symmetric RTP  . . . . . . . . . . . . . . . . . . . . . . 21
       4.3.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 22
       4.3.2.  Necessary RTSP extensions  . . . . . . . . . . . . . . 22
       4.3.3.  Deployment Considerations  . . . . . . . . . . . . . . 22
       4.3.4.  Security Consideration . . . . . . . . . . . . . . . . 23
       4.3.5.  A Variation to Symmetric RTP . . . . . . . . . . . . . 24
     4.4.  Application Level Gateways . . . . . . . . . . . . . . . . 26
       4.4.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 26
       4.4.2.  Outline On how ALGs for RTSP work  . . . . . . . . . . 27
       4.4.3.  Deployment Considerations  . . . . . . . . . . . . . . 28
       4.4.4.  Security Considerations  . . . . . . . . . . . . . . . 28
     4.5.  TCP Tunneling  . . . . . . . . . . . . . . . . . . . . . . 28
       4.5.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 28
       4.5.2.  Usage of TCP tunneling in RTSP . . . . . . . . . . . . 29
       4.5.3.  Deployment Considerations  . . . . . . . . . . . . . . 29
       4.5.4.  Security Considerations  . . . . . . . . . . . . . . . 29
     4.6.  TURN (Traversal Using Relay NAT) . . . . . . . . . . . . . 30
       4.6.1.  Introduction . . . . . . . . . . . . . . . . . . . . . 30
       4.6.2.  Usage of TURN with RTSP  . . . . . . . . . . . . . . . 30
       4.6.3.  Deployment Considerations  . . . . . . . . . . . . . . 31
       4.6.4.  Security Considerations  . . . . . . . . . . . . . . . 32
   5.  Firewalls  . . . . . . . . . . . . . . . . . . . . . . . . . . 33
   6.  Comparision of NAT traversal techniques  . . . . . . . . . . . 33



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   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 34
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 34
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35
   10. Informative References . . . . . . . . . . . . . . . . . . . . 35
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 37














































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1.  Introduction

   Today there is a proliferate deployment of different flavors of
   Network Address Translator (NAT) boxes that in many cases only
   loosely follows standards [RFC3022][RFC2663][RFC3424]].  NATs cause
   discontinuity in address realms [RFC3424], therefore an application
   protocol, such as Real-time Streaming Protocol (RTSP)
   [RFC2326][I-D.ietf-mmusic-rfc2326bis], needs to deal with such
   discontinuities caused by NATs.  The problem is that, being a media
   control protocol managing one or more media streams, RTSP carries
   network address and port information within its protocol messages.
   Because of this, even if RTSP itself, when carried over Transmission
   Control Protocol (TCP) [RFC0793] for example, may not be blocked by
   NATs, its media streams may be blocked by NATs.  This will occur
   unless special protocol provisions are added to support NAT-
   traversal.

   Like NATs, firewalls (FWs) are also middle boxes that need to be
   considered.  Firewalls helps prevent unwanted traffic from getting 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 effort.  The minimal effort is to
   implement an Application Level Gateway (ALG) to interpret RTSP
   parameters.  There is also a large class of firewalls, commonly home
   firewalls, that uses a similar filtering behavior to what NAT has.
   This type of firewalls can be handled using the same solution as
   employed for NAT traversal instead of relying on ALGs.

   This document describes several NAT-traversal mechanisms for RTSP
   controlled media streaming.  These NAT solutions fall into the
   category of "UNilateral Self-Address Fixing (UNSAF)" as defined in
   [RFC3424] 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 RFC 3424, we describe the
   required RTSP protocol extensions for each method, transition
   strategies, and security concerns.

   This document is capturing the evaluation done in the process to
   recommend FW/NAT traversal methods for RTSP streaming servers based
   on RFC 2326 [RFC2326] as well as the RTSP 2.0 core spec
   [I-D.ietf-mmusic-rfc2326bis].  The evaluation is focused on NAT
   traversal for the media streams carried over User Datagram Protocol
   (UDP) [RFC0768].  Where Real-time Transport Protocol (RTP) [RFC3550]



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   over UDP being the main case for such usage.  The findings should be
   applicable to other protocols as long as they have similar
   properties.

1.1.  Network Address Translators

   Readers are urged to refer to "IP Network Address Translator (NAT)
   Terminology and Considerations" [RFC2663] for information on NAT
   taxonomy and terminology.  Traditional NAT is the most common type of
   NAT device deployed.  Readers may refer to "Traditional IP Network
   Address Translator (Traditional NAT)" [RFC3022] for detailed
   information on traditional NAT.  Traditional NAT has two main
   varieties -- Basic NAT and Network Address/Port Translator (NAPT).

   NAPT is by far the most commonly deployed NAT device.  NAPT allows
   multiple internal hosts to share a single public IP address
   simultaneously.  When an internal host opens an outgoing TCP or UDP
   session through a NAPT, the NAPT assigns the session a public IP
   address and port number, so that subsequent response packets from the
   external endpoint can be received by the NAPT, translated, and
   forwarded to the internal host.  The effect is that the NAPT
   establishes a NAT mapping to translate the (private IP address,
   private port number) tuple to a (public IP address, public port
   number) tuple, and vice versa, for the duration of the session.  An
   issue of relevance to peer-to-peer applications is how the NAT
   behaves when an internal host initiates multiple simultaneous
   sessions from a single (private IP, private port) endpoint to
   multiple distinct endpoints on the external network.  In this
   specification, the term "NAT" refers to both "Basic NAT" and "Network
   Address/Port Translator (NAPT)".

   This document uses the term "address and port mapping" as the
   translation between an external address and port and an internal
   address and port.  Note that this is not the same as an "address
   binding" as defined in RFC 2663.  There exist a number of address and
   port mapping behaviors described in more detail in Section 4.1 of
   "Network Address Translation (NAT) Behavioral Requirements for
   Unicast UDP" [RFC4787].

   NATs also have a filtering behavior on traffic arriving on the
   external side.  Such behavior effects how well different methods for
   NAT traversal works through these NATs.  See Section 5 of "Network
   Address Translation (NAT) Behavioral Requirements for Unicast UDP"
   [RFC4787] for more information on the different types of filtering
   that have been identified.






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1.2.  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 external domain, e.g. public
   Internet.  Many organizations use firewalls to prevent privacy
   intrusions and malicious attacks to corporate computing resources in
   the private intranet [RFC2588].

   A comparison between NAT and FW is given below:

   1.  A firewall must sit between two network administrative domains,
       while NAT does not have to sit between two domains.

   2.  NAT does not in itself provide security, although some access
       control policies can be implemented using address translation
       schemes.  The inherent filtering behaviours are commonly mistaken
       for real security policies.

   It should be noted that many NAT devices intended for small office/
   home office (SOHO) include both NATs and firewall functionality.

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

1.3.  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.

   ICE:  Interactive Connectivity Establishment, see [RFC5245].

   DNS:  Domain Name Service

   DDOS: Distributed Denial Of Service attacks

   NAT:  Network Address Translator, see [RFC3022].

   NAPT: Network Address/Port Translator, see [RFC3022].

   RTP:  Real-time Transport Protocol, see [RFC3550].







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   RTSP: Real-Time Streaming Protocol, see [RFC2326] and
         [I-D.ietf-mmusic-rfc2326bis].

   SDP:  Session Description Protocol, see [RFC4566].

   SSRC: Synchronization source in RTP, see [RFC3550].

1.4.  Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in RFC
   2119 [RFC2119].


2.  Detecting the loss of NAT mappings

   Several 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 [RFC3424] 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 RFC3424.

   First, since NATs may also dynamically reclaim or readjust address/
   port translations, "keep-alive" and periodic re-polling may be
   required according to RFC 3424.  Secondly, it is possible to detect
   and recover from the situation where the mapping has been changed or
   removed.  The loss of a mapping can be detected when no traffic
   arrives for a while.  Below we will give some recommendation on how
   to detect loss of NAT mappings when using RTP/RTCP under RTSP
   control.

   A RTP session normally has both RTP and RTCP streams.  The loss of a
   RTP mapping can only be detected when expected traffic does not
   arrive.  If a client does not receive data within a few seconds after
   having received the "200 OK" response to a PLAY request, there are
   likely some middleboxes blocking the traffic.  However, for a
   receiver to be more 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.  RTCP is normally



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   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 neither RTCP packets nor
   RTSP messages are received by the RTSP server for a while, the RTSP
   server has the option to delete the corresponding RTP session, 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.


3.  Requirements on NAT-Traversal

   This section considers the set of requirements for the evaulation of
   RTSP NAT traversal solutions.

   RTSP is a client-server protocol.  Typically services providers
   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.  In terms of requirements, the
   first requirement should be to solve the RTSP NAT traversal problem
   for RTSP servers deployed in a public network, i.e. no NAT at the
   server side.

   The list of feature requirements for RTSP NAT solutions are given
   below:

   1.  MUST work for all flavors of NATs, including NATs with address
       and port restricted filtering.

   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.  RTSP dual-hosting means that RTSP protocol and the media
       protocol (e.g.  RTP) are implemented on different computers with
       different IP addresses.

       *  For instance, no extra delay from RTSP connection till arrival
          of media

   4.  SHOULD be simple to use/implement/administer that people actually
       turn them on

       *  Otherwise people will resort to TCP tunneling through NATs





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       *  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 determines the media
   destination (Address and port) for 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 target.example.com in order to degrade the services
   provided by target.example.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., centralized
   conferencing), 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.


4.  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.

   This section includes NAT traversal techniques that have not been
   formally specified anywhere else.  The overview section of this
   document may be the only publicly available specification of some of
   the NAT traversal techniques.  However that is no real barrier
   against doing an evaluation of the NAT traversal technique.  Some
   other techniques are currently (at the time of writing) in a state of



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   flux due to ongoing standardization work on these techniques, e.g.
   RTP No-Op [I-D.ietf-avt-rtp-no-op].

4.1.  STUN

4.1.1.  Introduction

   STUN - "Simple Traversal of UDP Through Network Address Translators"
   [RFC3489][RFC5389] is a standardized protocol that allows a client to
   use secure means to discover the presence of a NAT between himself
   and the STUN server.  The client uses the STUN server to discover the
   address mappings 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.

   The first version of STUN [RFC3489] included categorization and
   parameterization of NATs.  This was abandoned in the updated version
   due to it being unreliable.

4.1.2.  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.  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:

   o  The server must be located in either a public address realm or the
      next hop external address realm in regards to the client.

   o  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



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       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 [RFC4566].  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: client's local
       address/port <-> public address/port.

   4.  Perform the RTSP SETUP for each media.  In the transport header
       the following parameter SHOULD be included with the given values:
       "dest_addr" [I-D.ietf-mmusic-rfc2326bis] or "destination" +
       "client_port" [RFC2326] 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 setup the RTP stream on any port,
       not only even ports and with non-continuous port numbers for RTP
       and RTCP.  This requires the new feature provided in the update
       to RFC2326 [I-D.ietf-mmusic-rfc2326bis].  The server should
       respond with a transport header containing an "src_addr" or
       "source parameter" + "server_port" 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.  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
       No-Op packet [I-D.ietf-avt-rtp-no-op] to the streaming server's
       discard port (port number 9).  The drawback of using RTP No-Op is
       that the payload type number must be dynamically assigned through
       RTSP first.  Otherwise STUN could be used for the keep-alive as
       well as empty UDP packets.

   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 a
   "Address Dependent" filtering NAT, the client must first send a UDP
   packet to establish filtering state in the NAT.  The client, before



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   sending a RTSP PLAY request, must send a so called FW packet (such as
   a RTP No-Op 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 a reasonable number of times.

   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).

   This method is also brittle because it relies on that one can use
   STUN to classify the NAT behavior.  If the NAT changes the properties
   of the existing mapping and filtering state for example due to load,
   then the methods will fail.

4.1.3.  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.  This would require protocol changes.

   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.

   Deviations from STUN as defined in RFC 3489:

   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.



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   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 an RTSP server is behind a NAT, then an embedded STUN
   server must co-locate 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 that the server correctly can determine its
   NAT external source address for the media streams.  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
   can be done by using the "server_port" parameter in RFC2326, and the
   "src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis].  Both are in
   the RTSP Transport header.  But in general this strategy will require
   that one first do 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 end-point (server or
   client) will be required to send and receive RTCP packets from the
   same port.

4.1.4.  Discussion On Co-located STUN Server

   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 de-
   multiplexing problem: we must be able to differentiate a STUN packet
   from a media packet.  This will be done based on heuristics.  A
   common heuristics is the first byte in the packet, which works fine
   between STUN and RTP or RTCP where the first byte happens to be
   different, but may not work as well with other media transport



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   protocols.

4.1.5.  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.  An ALG should not update
   address fields which contains addresses other than the NATs internal
   address domain.  In cases where the ALG modifies fields unnecessary
   two alternatives exist:

   1.  The usage of TLS to encrypt the RTSP TCP connection to prevent
       the ALG from reading and modifying the RTSP messages.

   2.  To 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.

4.1.6.  Deployment Considerations

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

   Advantages:

   o  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.

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

   Disadvantages:

   o  Requires a STUN server deployed in the public address space.

   o  Only works with NATs that perform endpoint independent and address
      dependent mappings.  Port and address dependent filtering NATs
      create some issues.

   o  Brittle to NATs changing the properties of the NAT mapping and
      filtering.




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   o  Does not work with port and address dependent mapping NATs without
      server modifications.

   o  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 to prevent becoming a DDOS tool.

   o  Interaction problems exist when a RTSP-aware ALG interferes with
      the use of STUN for NAT traversal unless TLS secured RTSP message
      exchange is used.

   o  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:

   o  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.

   o  STUN has built-in message authentication features, which makes it
      more secure.  See next section for an in-depth security
      discussion.

   o  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 RFC3489).

   o  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 [STUN-IMPL].  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



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      to register the proper feature tags.

   Disadvantages:

   o  Some extensions to the RTSP core protocol, signaled by RTSP
      feature tags, must be introduced.

   o  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.

   o  Does not support use cases that requires the RTSP connection and
      the media reception to happen at different addresses, unless the
      servers sequrity policy is relaxed.

   o  Interaction problems exist when a RTSP ALG is not aware of STUN
      unless TLS is used to protect the RTSP messages.

   o  Using STUN requires that RTSP servers and clients support the
      updated RTSP specification, and they both agree to support the NAT
      traversal feature.

   o  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 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.

4.1.7.  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 use cases where RTSP



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   and media transport goes to different address.

   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.

4.2.  ICE

4.2.1.  Introduction

   ICE (Interactive Connectivity Establishment) [RFC5245] is a
   methodology for NAT traversal that has been developed 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 on a high level.  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 destinations to get the media
   through too A. Note that each of A's published address/port pairs has
   a STUN server co-located.  B, in its 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 have successfully completed
   are then indicated and 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 and therefore requests are



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

4.2.2.  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 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 been opened, or the deployment of proxies,
   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 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.

   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.

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

   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.

   4.  The server responds to the SETUP (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.




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   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.

   6.  Connectivity checks from the client's primary to the server's
       primary was successful.  Thus no further SETUP requests are
       necessary and processing can proceed with step 7.  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).  If the checks for the availble candidates failed and
       If further candidates have been derived during the connectivity
       checks, then those can be signalled in new candidate lines in
       SETUP request updating the list (Goto 5).

   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 back to 6.

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

   To keep media paths alive the client needs to periodically send data
   to the server.  This could be realized with either STUN or RTP No-op
   [I-D.ietf-avt-rtp-no-op] packets.  RTCP sent by client should be able
   to keep RTCP open.

4.2.3.  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.

   RTSP server implementation requirements are:

   o  STUN server features

   o  limited STUN client features

   o  SDP generation with more parameters.




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   o  RTSP error code for ICE extension

   RTSP client implantation requirements are:

   o  Limited STUN server features

   o  Limited STUN client features

   o  RTSP error code and ICE extension

4.2.4.  Deployment Considerations

   Advantages:

   o  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).

   o  Improves defenses against DDOS attacks, as media receiving client
      requires authentications, via STUN on its media reception ports.

   Disadvantages:

   o  Increases the setup delay with at least the amount of time it
      takes for the server to perform its STUN requests.

   o  Assumes that it is possible to de-multiplex between media packets
      and STUN packets.

   o  Has fairly high implementation burden put on both RTSP server and
      client.

4.2.5.  Security Consideration

   One should review the security consideration section of ICE and STUN
   to understand that ICE is contains some potential issues.  However
   these can be avoided by a correctly utilizing ICE in RTSP.  In fact
   ICE do help avoid the DDoS issue with RTSP substantially as it
   reduces the possibility for a DDoS using RTSP servers to attackers
   that are on-path between the RTSP server and the target and capable
   of intercepting the STUN connectivity check packets and correctly
   send a response to the server.

4.3.  Symmetric RTP






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4.3.1.  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 punch 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.  Using
   the rule for restriciting IP address to that one of the signalling
   connection will need to be applied here also.

4.3.2.  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 or use
   unique ports per RTSP session.  Using unique ports limits an RTSP
   server in the number of session it can simultaneously handle per
   interface IP addresses.

4.3.3.  Deployment Considerations

   Advantages:





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   o  Works for all types of NATs, including those using multiple IP
      addresses.  (Requirement 1 in Section 3).

   o  Have no interaction problems with any RTSP ALG changing the
      client's information in the transport header.

   Disadvantages:

   o  Requires modifications to both RTSP server and client.

   o  Limited to work with servers that have an public IP address.

   o  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 No-Op
      packet format in [I-D.ietf-avt-rtp-no-op].

   o  Has the same security situation as STUN and will need to use
      address restrictions.

4.3.4.  Security Consideration

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

   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 in use cases that require differet
   addresses for media destination and signalling.

   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 for a man in the middle to modify
   the RTP binding packet being sent by a client 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



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   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.

   Yet another issues is that these attacks also can be used to deny the
   client the service he desire from the RTSP server completely.  For a
   man in the middle capable of reading the signalling traffic or
   intercepting the binding packets can completely deny the client
   service by modifying or originating binding packets of itself.

   The random nonce used 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.

4.3.5.  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 No-Op format is a proposal to
   fix the problem [I-D.ietf-avt-rtp-no-op].  There exist a document
   that discusses the implication of different type of packets as keep-
   alives for RTP [I-D.ietf-avt-app-rtp-keepalive] and its findings are
   very relevant to the FW packet.

   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.



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

   The FW packet can contain the SSRC 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 possible 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 3), although a binding protocol must be
       defined.

   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 3).




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   This approach has the following disadvantages when compared with the
   symmetric RTP approach:

   1.  RTP traffic is normally accompanied by RTCP traffic.  This
       approach needs to rely on RTCP RRs and SRs to enable NAT
       traversal for RTCP endpoints, or use the same type of FW messages
       also 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.

   A solution with a 3-way handshaking and its own FW packets can be
   compared with ICE and have the following differencies:

   o  Only works for servers with public IP addresses compared to any
      type of server

   o  Is somewhat simpler to implement due to the avoidance of the ICE
      prioritization and checkboard mechanisms.

   However, a 3-way binding protocol is very similar to using STUN in
   both directions as binding protocol.  Using STUN would remove the
   need for implementing a new protocol.

4.4.  Application Level Gateways

4.4.1.  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:

   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.





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4.4.2.  Outline On how ALGs for RTSP work

   In this section, we provide a step-by-step outline 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.

   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



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       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.

4.4.3.  Deployment Considerations

   Advantage:

   o  No impact on either client or server

   o  Can work for any type of NATs

   Disadvantage:

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

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

   o  Can interfere with other 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.

4.4.4.  Security Considerations

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

4.5.  TCP Tunneling

4.5.1.  Introduction

   Using a TCP connection that is established from the client to the
   server ensures that the server can send data to the client.  The
   connection opened from the private domain ensures that the server can
   send data back to the client.  To send data originally intended to be
   transported over UDP requires the TCP connection to support some type
   of framing of the media data 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



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   why RTP was developed.  Problems that arise with TCP are: head-of-
   line blocking, delay introduced by retransmissions, highly varying
   rate due to the congestion control algorithm.

4.5.2.  Usage of TCP tunneling in RTSP

   The RTSP core specification [I-D.ietf-mmusic-rfc2326bis] supports
   interleaving of media data on the TCP connection that carries RTSP
   signaling.  See section 14 in [I-D.ietf-mmusic-rfc2326bis] for how to
   perform this type of TCP tunneling.  There also exist another way of
   transporting RTP over TCP defined in Appendix C.2.  For signaling and
   rules on how to establish the TCP connection in lieu of UDP, see
   appendix C.2 in [I-D.ietf-mmusic-rfc2326bis].  This is based on the
   framing of RTP over the TCP connection as described in RFC 4571
   [RFC4571].

4.5.3.  Deployment Considerations

   Advantage:

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

   Disadvantage:

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

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

   Transition:

   The tunneling over RTSP's TCP connection is not planned to be phased-
   out.  It is intended to be a fallback mechanism and for usage when
   total media reliability is desired, even at the price of loss of
   real-time properties.

4.5.4.  Security Considerations

   The TCP tunneling of RTP has no known security problem besides those
   already presented in the RTSP specification.  It is not possible to
   get any amplification effect that is desired for denial of service
   attacks due to TCP's flow control.  A possible security
   consideration, 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.







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4.6.  TURN (Traversal Using Relay NAT)

4.6.1.  Introduction

   Traversal Using Relay NAT (TURN) [RFC5766] 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 or also the RTSP
   connection is relayed through the same TURN server.

4.6.2.  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
       from different addresses also the TCP connection must be done
       through the same TURN server as the one in the next step.  This
       will require the usage of TURN for TCP [RFC6062].

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



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       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.

4.6.3.  Deployment Considerations

   Advantages:

   o  Does not require any server modifications.

   o  Works for any types of NAT as long as the server has public
      reachable IP address.

   Disadvantage:

   o  Requires another network element, namely the TURN server.

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

   o  TURN server becomes a single point of failure.

   o  Since TURN forwards media packets, it necessarily introduces
      delay.





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

   Transition:

   TURN is not intended to be phase-out completely, see chapter 11.2 of
   [RFC5766].  However the usage of TURN could be reduced when the
   demand for having NAT traversal is reduced.

4.6.4.  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 4.6.2.

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

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

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




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5.  Firewalls

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

   The firewall will need to allow the media streams associated with 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.


6.  Comparision of NAT traversal techniques

   This section evaluates the techniques described above against the
   requirements listed in section Section 3.

   In the following table, the columns correspond to the numbered
   requirements.  For instance, the column under R1 corresponds to the
   first requirement in section Section 3: 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 Section 4.3.5.

   A Summary of the requirements are:

   R1 Work for all flavors of NATs







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   R2 Most work with Firewalls, including them with ALGs

   R3 Should have minimal impact on clients not behind NATs

   R4 Should be simple to use, Implement and administrate.

   R5 Should provide a mitigation against DDoS attacks

   -----------------------------------------------+
               |  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|
   ------------+------+------+------+------+------+
    3-W SymRTP | Yes  | Yes  | Yes  | Maybe| Yes  |
   ------------+------+------+------+------+------+
    TURN       | Yes  | Yes  | No   | No   | Yes  |
   ------------------------------------------------

   The different techniques was discussed in the MMUSIC WG.  It was
   established that the WG would pursue an ICE based solution due to its
   generality and capability of handle also servers delivering media
   from behind NATs.  There has been some discussion if the increased
   implementation burden of ICE is motivated compared to a 3-W SymRTP
   solution for this generality.


7.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.


8.  Security Considerations

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



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   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 or
   denial of service 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 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.


9.  Acknowledgements

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

   Section Section 1.1 contains text originally written for RFC 4787 by
   Francois Audet and Cullen Jennings.


10.  Informative References

   [I-D.ietf-avt-app-rtp-keepalive]
              Marjou, X. and A. Sollaud, "Application Mechanism for
              keeping alive the Network Address Translator (NAT)
              mappings associated to RTP/RTCP flows.",
              draft-ietf-avt-app-rtp-keepalive-10 (work in progress),
              March 2011.

   [I-D.ietf-avt-rtp-no-op]
              Andreasen, F., "A No-Op Payload Format for RTP",
              draft-ietf-avt-rtp-no-op-04 (work in progress), May 2007.



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   [I-D.ietf-mmusic-rfc2326bis]
              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-27 (work in
              progress), March 2011.

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

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

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

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

   [RFC2588]  Finlayson, R., "IP Multicast and Firewalls", RFC 2588,
              May 1999.

   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
              Translator (NAT) Terminology and Considerations",
              RFC 2663, August 1999.

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

   [RFC3424]  Daigle, L. and IAB, "IAB Considerations for UNilateral
              Self-Address Fixing (UNSAF) Across Network Address
              Translation", RFC 3424, November 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

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

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

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



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   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

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

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

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations",
              RFC 6062, November 2010.

   [STUN-IMPL]
              "Open Source STUN Server and Client, http://
              www.vovida.org/applications/downloads/stun/index.html",
              June 2007.


Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   Stockholm,   SE-164 80
   Sweden

   Phone: +46 8 719 0000
   Fax:
   Email: magnus.westerlund@ericsson.com
   URI:












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   Thomas Zeng


   Phone:
   Fax:
   Email: thomas.zeng@gmail.com
   URI:












































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