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Versions: 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 RFC 6314

SIPPING Working Group                                         C. Boulton
Internet-Draft                             Ubiquity Software Corporation
Expires: September 3, 2006                                  J. Rosenberg
                                                           Cisco Systems
                                                            G. Camarillo
                                                                Ericsson
                                                           March 2, 2006


            Best Current Practices for NAT Traversal for SIP
                  draft-ietf-sipping-nat-scenarios-04

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Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   Traversal of the Session Initiation Protocol (SIP) and the sessions
   it establishes through Network Address Translators (NAT) is a complex
   problem.  Currently there are many deployment scenarios and traversal
   mechanisms for media traffic.  This document aims to provide concrete
   recommendations and a unified method for NAT traversal as well as



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   documenting corresponding call flows.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Problem Statement  . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Solution Technology Outline Description  . . . . . . . . . . .  6
     3.1.  SIP Signaling  . . . . . . . . . . . . . . . . . . . . . .  7
       3.1.1.  Symmetric Response . . . . . . . . . . . . . . . . . .  7
       3.1.2.  Connection Re-use  . . . . . . . . . . . . . . . . . .  8
     3.2.  Media Traversal  . . . . . . . . . . . . . . . . . . . . .  8
       3.2.1.  Symmetric RTP  . . . . . . . . . . . . . . . . . . . .  8
       3.2.2.  STUN . . . . . . . . . . . . . . . . . . . . . . . . .  9
       3.2.3.  TURN . . . . . . . . . . . . . . . . . . . . . . . . .  9
       3.2.4.  ICE  . . . . . . . . . . . . . . . . . . . . . . . . .  9
       3.2.5.  RTCP Attribute . . . . . . . . . . . . . . . . . . . . 10
       3.2.6.  Solution Profiles  . . . . . . . . . . . . . . . . . . 10
   4.  NAT Traversal Scenarios  . . . . . . . . . . . . . . . . . . . 11
     4.1.  Basic NAT SIP Signaling Traversal  . . . . . . . . . . . . 11
       4.1.1.  Registration (Registrar/Proxy Co-Located)  . . . . . . 11
       4.1.2.  Registration(Registrar/Proxy not Co-Located) . . . . . 15
       4.1.3.  Initiating a Session . . . . . . . . . . . . . . . . . 16
       4.1.4.  Receiving an Invitation to a Session . . . . . . . . . 18
     4.2.  Basic NAT Media Traversal  . . . . . . . . . . . . . . . . 21
       4.2.1.  Port Restricted Cone NAT . . . . . . . . . . . . . . . 21
       4.2.2.  Symmetric NAT  . . . . . . . . . . . . . . . . . . . . 33
     4.3.  Advanced NAT media Traversal Using ICE . . . . . . . . . . 38
       4.3.1.  Full Cone --> Full Cone traversal  . . . . . . . . . . 38
       4.3.2.  Port Restricted Cone --> Port Restricted Cone
               traversal  . . . . . . . . . . . . . . . . . . . . . . 38
       4.3.3.  Internal TURN Server (Enterprise Deployment) . . . . . 38
     4.4.  Intercepting Intermediary (B2BUA)  . . . . . . . . . . . . 38
     4.5.  IPv4-IPv6 Transition . . . . . . . . . . . . . . . . . . . 38
       4.5.1.  IPv4-IPv6 Transition for SIP Signalling  . . . . . . . 39
       4.5.2.  IPv4-IPv6 Transition for Media . . . . . . . . . . . . 39
     4.6.  ICE with RTP/TCP . . . . . . . . . . . . . . . . . . . . . 41
   5.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 41
   6.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 42
     6.1.  Normative References . . . . . . . . . . . . . . . . . . . 42
     6.2.  Informative References . . . . . . . . . . . . . . . . . . 43
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 44
   Intellectual Property and Copyright Statements . . . . . . . . . . 45








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1.  Introduction

   NAT (Network Address Translators) traversal has long been identified
   as a large problem when considered in the context of the Session
   Initiation Protocol (SIP)[1] and it's associated media such as Real
   Time Protocol (RTP)[2].  The problem is further confused by the
   variety of NATs that are available in the market place today and the
   large number of potential deployment scenarios.  Detail of different
   NAT types can be found in RFC 3489bis [16].

   The IETF has produced many specifications for the traversal of NAT,
   including STUN, ICE, rport, symmetric RTP, TURN, connection reuse,
   SDP attribute for RTCP, and others.  These each represent a part of
   the solution, but none of them gives the overall context for how the
   NAT traversal problem is decomposed and solved through this
   collection of specifications.  This document serves to meet that
   need.

   This document attempts to provide a definitive set of 'Best Common
   Practices' to demonstrate the traversal of SIP and its associated
   media through NAT devices.  The document does not propose any new
   functionality but does draw on existing solutions for both core SIP
   signaling and media traversal (as defined in Section 3).

   The draft will be split into distinct sections as follows:
   1.  A clear definition of the problem statement
   2.  Description of proposed solutions for both SIP protocol signaling
       and media signaling
   3.  A set of basic and advanced call flow scenarios


2.  Problem Statement

   The traversal of SIP through NAT can be split into two categories
   that both require attention - The core SIP signaling and associated
   media traversal.

   The core SIP signaling has a number of issues when traversing through
   NATs.

   Firstly, the default operation for SIP response generation using
   unreliable protocols such as the Unicast Datagram Protocol (UDP)
   results in responses being generated at the User Agent Server (UAS)
   being sent to the source address, as specified in either the SIP
   'Via' header or the 'received' parameter (as defined in RFC 3261
   [1]).  The port is extracted from the SIP 'Via' header to complete
   the IP address/port combination for returning the SIP response.
   While the destination is correct, the port contained in the SIP 'Via'



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   header represents the listening port of the originating client and
   not the port representing the open pin hole on the NAT.  This results
   in responses being sent back to the NAT but to a port that is likely
   not open for SIP traffic.  The SIP response will then be dropped at
   the NAT.  This is illustrated in Figure 1 which depicts a SIP
   response being returned to port 5060.


     Private                       NAT                         Public
     Network                        |                          Network
                                    |
                                    |
     --------     SIP Request       |open port 5650            --------
    |        |-------------------->--->-----------------------|        |
    |        |                      |                         |        |
    | Client |                      |port 5060   SIP Response | Proxy  |
    |        |                      x<------------------------|        |
    |        |                      |                         |        |
     --------                       |                          --------
                                    |
                                    |
                                    |


   Figure 1

   Secondly, when using a reliable, connection orientated transport
   protocol such as TCP, SIP has an inherent mechanism that results in
   SIP responses reusing the connection that was created/used for the
   corresponding transactional request.  The SIP protocol does not
   provide a mechanism that allows new requests generated in the reverse
   direction of the originating client to use the existing TCP
   connection created between the client and the server during
   registration.  This results in the registered contact address not
   being bound to the "connection" in the case of TCP.  Requests are
   then blocked at the NAT, as illustrated in Figure 2.  This problem
   also exists for unreliable transport protocols such as UDP where
   external NAT mappings need to be re-used to reach a SIP entity on the
   private side of the network.












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     Private                       NAT                         Public
     Network                        |                          Network
                                    |
                                    |
     -------- (UAC 8023)    REGISTER/Response       (UAS 5060) --------
    |        |-------------------->---<-----------------------|        |
    |        |                      |                         |        |
    | Client |                      |5060  INVITE   (UAC 8015)| Proxy  |
    |        |                      x<------------------------|        |
    |        |                      |                         |        |
     --------                       |                          --------
                                    |
                                    |
                                    |

   Figure 2

   In Figure 2 the original REGISTER request is sent from the client on
   port 8023 and received on port 5060, establishing a reliable
   connection and opening a pin-hole in the NAT.  The generation of a
   new request from the proxy results in a request destined for the
   registered entity (Contact IP address) which is not reachable from
   the public network.  This results in the new SIP request attempting
   to create a connection to a private network address.  This problem
   would be solved if the original connection was re-used.  While this
   problem has been discussed in the context of connection orientated
   protocols such as TCP, the problem exists for SIP signaling using any
   transport protocol.  The solution proposed for this problem in
   section 3 of this document is relevant for all SIP signaling,
   regardless of the transport protocol.

   NAT policy can dictate that connections should be closed after a
   period of inactivity.  This period of inactivity can range
   drastically from a number seconds to hours.  Pure SIP signaling can
   not be relied upon to keep alive connections for a number of reasons.
   Firstly, SIP entities can sometimes have no signaling traffic for
   long periods of time which has the potential to exceed the inactivity
   timer, and this can lead to problems where endpoints are not
   available to receive incoming requests as the connection has been
   closed.  Secondly, if a low inactivity timer is specified, SIP
   signaling is not appropriate as a keep-alive mechanism as it has the
   potential to add a large amount of traffic to the network which uses
   up valuable resource and also requires processing at a SIP stack,
   which is also a waste of processing resources.

   Media associated with SIP calls also has problems traversing NAT.
   RTP [2]] is one of the most common media transport types used in SIP
   signaling.  Negotiation of RTP occurs with a SIP session



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   establishment using the Session Description Protocol(SDP) [3] and a
   SIP offer/answer exchange[5].  During a SIP offer/answer exchange an
   IP address and port combination are specified by each client in a
   session as a means of receiving media such as RTP.  The problem
   arises when a client advertises its address to receive media and it
   exists in a private network that is not accessible from outside the
   NAT.  Figure 3 illustrates this problem.


                 NAT             Public Network           NAT
                  |                                        |
                  |                                        |
                  |                                        |
     --------     |            SIP Signaling Session       |   --------
    |        |----------------------->---<--------------------|        |
    |        |    |                                        |  |        |
    | Client |    |                                        |  | Client |
    |   A    |>=====>RTP>==Unknown Address==>X             |  |   B    |
    |        |    |             X<==Unknown Address==<RTP<===<|        |
     --------     |                                        |   --------
                  |                                        |
                  |                                        |
                  |                                        |


   Figure 3

   The connection address representing both clients are not available on
   the public internet and traffic can be sent from both clients through
   their NATs.  The problem occurs when the traffic reaches the public
   internet and is not resolveable.  The media traffic fails.  The
   connection address extracted from the SDP payload is that of an
   internal address, and so not resolvable from the public side of the
   NAT.  To complicate the problem further, a number of different NAT
   topologies with different default behaviors increase the difficulty
   of proposing a single solution.


3.  Solution Technology Outline Description

   When analyzing issues associated with traversal of SIP through
   existing NAT, it has been identified that the problem can be split
   into two clear solution areas as defined in section 2 of this
   document.  The traversal of the core protocol signaling and the
   traversal of the associated media as specified in the Session
   Description Payload (SDP) of a SIP offer/answer exchange[5].  The
   following sub-sections outline solutions that enable core SIP
   signaling and its associated media to traverse NATs.



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3.1.  SIP Signaling

   SIP signaling has two areas that result in transactional failure when
   traversing through NAT, as described in section 2 of this document.
   The remaining sub-sections describe appropriate solutions that result
   in SIP signalling traversal through NAT, regardless of transport
   protocol.  IT is RECOMMEDED that SIP compliant entities follow the
   guidelines presented in this section to enable traversal of SIP
   signaling through NATs.

3.1.1.  Symmetric Response

   As described in section 2 of this document, when using an unreliable
   transport protocol such as UDP, SIP responses are sent to the IP
   address and port combination contained in the SIP 'Via' header field
   (or default port for the appropriate transport protocol if not
   present).  This can result in responses being blocked at a NAT.  In
   such circumstances, SIP signaling requires a mechanism that will
   allow entities to override the basic response generation mechanism in
   RFC 3261 [1].  Once the SIP response is constructed, the destination
   is still derived using the mechanisms described in RFC 3261 [1].  The
   port (to which the response will be sent), however, will not equal
   that specified in the SIP 'Via' header field but will be the port
   from which the original request was sent.  This results in the pin-
   hole opened for the requests traversal of the NAT being reused, in a
   similar manner to that of reliable connection orientated transport
   protocols such as TCP.  Figure 4 illustrates the response traversal
   through the open pin hole using this method.


     Private                        NAT                       Public
     Network                         |                        Network
                                     |
                                     |
     --------                        |                        --------
    |        |                       |                       |        |
    |        |send/receive           |           send/receive|        |
    | Client |port 5060-----<<->>---------<<->>-----port 5060| Client |
    |   A    |                       |                       |   B    |
    |        |                       |                       |        |
     --------                        |                        --------
                                     |
                                     |
                                     |

   Figure 4

   The exact functionality for this method of response traversal is



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   called 'Symmetric Response' and the details are documented in RFC
   3581 [6].  Additional requirements are imposed on SIP entities in
   this specification such as listening and sending SIP requests/
   responses from the same port.

3.1.2.  Connection Re-use

   The second problem with sip signaling, as defined in Section 2 and
   illustrated in Figure 2, is to allow incoming requests to be properly
   routed.

   Guidelines for devices such as User Agents that can only generate
   outbound connections through a NAT are documented in 'SIP Conventions
   for UAs with Outbound Only Connections'[11].  The document provides
   techniques for the reuse of a TCP connection or UDP 5-tuple for
   incoming requests.  It also provides a keepalive mechanism based on
   using STUN to the SIP server.  Usage of this specification is
   RECOMMENDED.  This mechanism is not transport specific and should be
   used for any transport protocol.

   Even if the previous draft is not used, clients SHOULD use the same
   IP address and port (i.e., socket) for both transmission and receipt
   of SIP messages.  Doing so allows for the vast majority of industry
   provided solutions to properly function.

3.2.  Media Traversal

   This document has already provided guidelines that recommend using
   extensions to the core SIP protocol to enable traversal of NATs.
   While ultimately not desirable, the additions are relatively straight
   forward and provide a simple, universal solution for varying types of
   NAT deployment.  The issues of media traversal through NATs is not
   straight forward and requires the combination of a number of
   traversal methodologies.  The technologies outlined in the remainder
   of this section provide the required solution set.

3.2.1.  Symmetric RTP

   The primary problem identified in section 2 of this document is that
   internal IP address/port combinations can not be reached from the
   public side of a NAT.  In the case of media such as RTP, this will
   result in no audio traversing a NAT(as illustrated in Figure 3).  To
   overcome this problem, a technique called 'Symmetric' RTP can be
   used.  This involves an SIP endpoint both sending and receiving RTP
   traffic from the same IP Address/Port combination.  This technique
   also requires intelligence by a client on the public internet as it
   identifies that incoming media for a particular session does not
   match the information that was conveyed in the SDP.  In this case the



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   client will ignore the SDP address/port combination and return RTP to
   the IP address/port combination identified as the source of the
   incoming media.  This technique is known as 'Symmetric RTP' and is
   documented in [14].  'Symmetric RTP' SHOULD only be used for
   traversal of RTP through NAT when one of the participants in a media
   session definitively knows that it is on the public network.

3.2.2.  STUN

   Simple Traversal of User Datagram Protocol(UDP) through Network
   Address Translators(NAT) or STUN is defined in RFC 3489 [10].  It
   provides a lightweight protocol that allows entities to probe and
   discover the type of NAT that exists between itself and external
   entities.  It also provides details of the external IP address/port
   combination used by the NAT device to represent the internal entity
   on the public facing side of a NAT.  On learning of such an external
   representation, a client can use it accordingly as the connection
   address in SDP to provide NAT traversal.  STUN only works with Full
   Cone, Restricted Cone and Port Restricted Cone type NATs.  STUN does
   not work with Symmetric NATs as the technique used to probe for the
   external IP address port representation using a STUN server will
   provide a different result to that required for traversal by an
   alternative SIP entity.  The IP address/port combination deduced for
   the STUN server would be blocked for incoming packets from an
   alterative SIP entity.

3.2.3.  TURN

   As mentioned in the previous section, the STUN protocol does not work
   for UDP traversal through a Symmetric style NAT.  Traversal Using
   Relay NAT (TURN) provides the solution for UDP traversal of symmetric
   NAT.  TURN is extremely similar to STUN in both syntax and operation.
   It provides an external address at a TURN server that will act as a
   relay and guarantee traffic will reach the associated internal
   address.  The full details of the TURN specification are defined in
   [13].  A TURN service will almost always provide media traffic to a
   SIP entity but it is RECOMMENDED that this method only be used as a
   last resort and not as a general mechanism for NAT traversal.  This
   is because using TURN has high performance costs when relaying media
   traffic and can lead to unwanted latency.

3.2.4.  ICE

   Interactive Connectivity Establishment (ICE) is the RECOMMENDED
   method for traversal of existing NAT if Symmetric RTP is not
   appropriate.  ICE is a methodology for using existing technologies
   such as STUN, TURN and any other UNSAF[9] compliant protocol to
   provide a unified solution.  This is achieved by obtaining as many



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   representative IP address/port combinations as possible using
   technologies such as STUN/TURN etc.  Once the addresses are
   accumulated, they are all included in the SDP exchange in a new media
   attribute called 'candidate'.  Each 'candidate' SDP attribute entry
   has detailed connection information including a media addresses
   (including optional RTCP information), priority, username, password
   and a unique session ID.  The appropriate IP address/port
   combinations are used in the correct order depending on the specified
   priority.  A client compliant to the ICE specification will then
   locally run instances of STUN servers on all addresses being
   advertised using ICE.  Each instance will undertake connectivity
   checks to ensure that a client can successfully receive media on the
   advertised address.  Only connections that pass the relevant
   connectivity checks are used for media exchange.  The full details of
   the ICE methodology are contained in [15].

3.2.5.  RTCP Attribute

   Normal practice when selecting a port for defining Real Time Control
   Protocol(RTCP) [2] is for consecutive order numbering (i.e select an
   incremented port for RTCP from that used for RTP).  This assumption
   causes RTCP traffic to break when traversing many NATs due to blocked
   ports.  To combat this problem a specific address and port need to be
   specified in the SDP rather than relying on such assumptions.  RFC
   3605 [6] defines an SDP attribute that is included to explicitly
   specify transport connection information for RTCP.  The address
   details can be obtained using any appropriate method including those
   detailed previously in this section (e.g.  STUN, TURN).

3.2.6.  Solution Profiles

   This draft has documented a number of technology solutions for the
   traversal of media through differing NAT deployments.  A number of
   'profiles' will now be defined that categorize varying levels of
   support for the technologies described.

3.2.6.1.  Primary Profile

   A client falling into the 'Primary' profile supports ICE in
   conjunction with STUN, TURN and RFC 3605 [6] for RTCP.  ICE is used
   in all cases and falls back to standard operation when dealing with
   non-ICE clients.  A client which falls into the 'Primary' profile
   will be maximally interoperable and function in a rich variety of
   environments including enterprise, consumer and behind all varieties
   of NAT.






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3.2.6.2.  Consumer Profile

   A client falling into the 'Consumer' profile supports STUN and RFC
   3605 [6] for RTCP.  It uses STUN to allocate bindings, and can also
   detect when it is in the unfortunate situation of being behind a
   'Symmetric' NAT, although it simply cannot function in this case.
   These clients will only work in deployment situations where the
   access is sufficiently controlled to know definitively that there
   won't be Symmetric NAT.  This is hard to guarantee as users can
   always pick up their client and connect via a different access
   network.

3.2.6.3.  Minimal Profile

   A client falling into the 'Minimal' profile will send/receive RTP
   form the same IP/port combination.  This client requires proprietary
   network based solutions to function in any NAT traversal scenario.

   All clients SHOULD support the 'Primary Profile', MUST support the
   'Minimal Profile' and MAY support the 'Consumer Profile'.


4.  NAT Traversal Scenarios

   This section of the document includes detailed NAT traversal
   scenarios for both SIP signaling and the associated media.

4.1.  Basic NAT SIP Signaling Traversal

   The following sub-sections concentrate on SIP signaling traversal of
   NAT.  The scenarios include traversal for both reliable and un-
   reliable transport protocols.

   [Editors Note: The scenarios are still in early construction and a
   couple have been included as a hint of direction - All comments
   welcome for next release]

4.1.1.  Registration (Registrar/Proxy Co-Located)

   The set of scenarios in this section document basic signaling
   traversal of a SIP REGISTER method through a NAT.

4.1.1.1.  UDP








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           Client              NAT               Proxy
             |                  |                  |
             |(1) REGISTER      |                  |
             |----------------->|                  |
             |                  |(1) REGISTER      |
             |                  |----------------->|
             |                  |(2) 401 Unauth    |
             |                  |<-----------------|
             |(2) 401 Unauth    |                  |
             |<-----------------|                  |
             |(3) REGISTER      |                  |
             |----------------->|                  |
             |                  |(3) REGISTER      |
             |                  |----------------->|
             |*************************************|
             |    Create Connection Re-use Tuple   |
             |*************************************|
             |                  |(4) 200 OK        |
             |                  |<-----------------|
             |(4) 200 OK        |                  |
             |<-----------------|                  |
             |                  |                  |

     Figure 5.

   Figure 5

   In this example the client sends a SIP REGISTER request through a NAT
   which is challenged using the Digest authentication scheme.  The
   client will include an 'rport' parameter as described in section
   3.1.1 of this document for allowing traversal of UDP responses.  The
   original request as illustrated in (1) in Figure 5 is a standard
   REGISTER message:

    REGISTER sip:proxy.example.com SIP/2.0
    Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK
    Max-Forwards: 70
    Supported: gruu
    From: Client <sip:client@example.com>;tag=djks8732
    To: Client <sip:client@example.com>
    Call-ID: 763hdc73y7dkb37@example.com
    CSeq: 1 REGISTER
    Contact: <sip:client@client.example.com>; connectioId=1
         ;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E120>"
    Content-Length: 0

   This proxy now generates a SIP 401 response to challenge for
   authentication, as depicted in (2) from Figure 5:



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    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP client.example.com:5060
       ;rport=8050;branch=z9hG4bK;received=192.0.1.2
    From: Client <sip:client@example.com>;tag=djks8732
    To: Client <sip:client@example.com>;tag=876877
    Call-ID: 763hdc73y7dkb37@example.com
    CSeq: 1 REGISTER
    WWW-Authenticate: [not shown]
    Content-Length: 0

   The response will be sent to the address appearing in the 'received'
   parameter of the SIP 'Via' header (address 192.0.1.2).  The response
   will not be sent to the port deduced from the SIP 'Via' header, as
   per standard SIP operation but will be sent to the value that has
   been stamped in the 'rport' parameter of the SIP 'Via' header (port
   8050).  For the response to successfully traverse the NAT, all of the
   conventions defined in RFC 3581 [6] MUST be obeyed.  Make note of the
   both the 'connectionID' and 'sip.instance' contact header parameters.
   They are used to establish a connection re-use tuple as defined in
   [11].  The connection tuple creation is clearly shown in Figure 5.
   This ensures that any inbound request that causes a registration
   lookup will result in the re-use of the connection path established
   by the registration.  This exonerates the need to manipulate contact
   header URI's to represent a globally routable address as perceived on
   the public side of a NAT.  The subsequent messages defined in (3) and
   (4) from Figure 5 use the same mechanics for NAT traversal.

   [Editors note: Will provide more details on heartbeat mechanism in
   next revision]

   [Editors note: Can complete full flows if required on heartbeat
   inclusion]

4.1.1.2.  Reliable Transport

















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           Client              NAT             Registrar
             |                  |                  |
             |(1) REGISTER      |                  |
             |----------------->|                  |
             |                  |(1) REGISTER      |
             |                  |----------------->|
             |                  |(2) 401 Unauth    |
             |                  |<-----------------|
             |(2) 401 Unauth    |                  |
             |<-----------------|                  |
             |(3) REGISTER      |                  |
             |----------------->|                  |
             |                  |(3) REGISTER      |
             |                  |----------------->|
             |*************************************|
             |    Create Connection Re-use Tuple   |
             |*************************************|
             |                  |(4) 200 OK        |
             |                  |<-----------------|
             |(4) 200 OK        |                  |
             |<-----------------|                  |
             |                  |                  |

     Figure 6.

   Traversal of SIP REGISTER requests/responses using a reliable,
   connection orientated protocol such as TCP does not require any
   additional core SIP signaling extensions.  SIP responses will re-use
   the connection created for the initial REGISTER request, (1) from
   Figure 6:


    REGISTER sip:proxy.example.com SIP/2.0
    Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
    Max-Forwards: 70
    Supported: gruu
    From: Client <sip:client@example.com>;tag=djks809834
    To: Client <sip:client@example.com>
    Call-ID: 763hdc783hcnam73@example.com
    CSeq: 1 REGISTER
    Contact: <sip:client@client.example.com;transport=tcp>;connectioId=1
         ;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E121>"
    Content-Length: 0

   This example was included to show the inclusion of the connection re-
   use Contact header parameters as defined in the Connection Re-use
   draft [11].  This creates an association tuple as described in the
   previous example for future inbound requests directed at the newly



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   created registration binding with the only difference that the
   association is with a TCP connection, not a UDP pin hole binding.

   [Editors note: Will provide more details on heartbeat mechanism in
   next revision]

   [Editors note: Can complete full flows on inclusion of heartbeat
   mechanism]

4.1.2.  Registration(Registrar/Proxy not Co-Located)

   This section demonstrates traversal mechanisms when the Registrar
   component is not co-located with the edge proxy element.  The
   procedures described in this section are identical, regardless of
   transport protocol and so only one example will be documented in the
   form of TCP.


     Client              NAT               Proxy            Registrar
       |                  |                  |                  |
       |(1) REGISTER      |                  |                  |
       |----------------->|                  |                  |
       |                  |(1) REGISTER      |                  |
       |                  |----------------->|                  |
       |                  |                  |(2) REGISTER      |
       |                  |                  |----------------->|
       |                  |                  |(3) 401 Unauth    |
       |                  |                  |<-----------------|
       |                  |(4) 401 Unauth    |                  |
       |                  |<-----------------|                  |
       |(4)401 Unauth     |                  |                  |
       |<-----------------|                  |                  |
       |(5)REGISTER       |                  |                  |
       |----------------->|                  |                  |
       |                  |(5)REGISTER       |                  |
       |                  |----------------->|                  |
       |                  |                  |(6)REGISTER       |
       |                  |                  |----------------->|
       |                  |                  |(7)200 OK         |
       |                  |                  |<-----------------|
       |********************************************************|
       |             Create Connection Re-use Tuple             |
       |********************************************************|
       |                  |(8)200 OK         |                  |
       |                  |<-----------------|                  |
       |(8)200 OK         |                  |                  |
       |<-----------------|                  |                  |
       |                  |                  |                  |



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     Figure 7.

   This scenario builds on that contained in section 4.1.1.2.  This time
   the REGISTER request is routed onwards to a separated Registrar.  The
   important message to note is (6) in Figure 7.  At this point, the
   proxy server routes the SIP REGISTER message to the Registrar.  The
   proxy will create the connection re-use tuple at the same moment as
   the co-located example but for subsequent messages to arrive at the
   Proxy, the element needs to request to stay in the signaling path.
   The REGISTER message (5) contains a SIP PATH extension header, as
   defined in RFC 3327 [7].  REGISTER message (5) would look as follows:


    REGISTER sip:registrar.example.com SIP/2.0
    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4njkca8398hadjaa
    Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
    Max-Forwards: 70
    Supported: gruu
    From: Client <sip:client@example.com>;tag=djks809834
    To: Client <sip:client@example.com>
    Call-ID: 763hdc783hcnam73@example.com
    CSeq: 1 REGISTER
    Path: <sip:sip%3Aclient%40example.com@proxy.example.com;lr>
    Contact: <sip:client@client.example.com;transport=tcp>;connectioId=1
         ;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E121>"
    Content-Length: 0


   This results in the Path header being stored along with the AOR and
   it's associated binding at the Registrar.  The URI contained in the
   Path header will be inserted as a pre-loaded SIP 'Route' header into
   any request that arrives at the Registrar and is directed towards the
   associated binding.  This guarantees that all requests for the new
   Registration will be forwarded to the edge proxy.  The user part of
   the SIP 'Path' header URI that was inserted by the edge proxy
   contains an escaped form of the original AOR that was contained in
   the REGISTER request.  On receiving subsequent requests, the edge
   proxy will examine the user part of the pre-loaded SIP 'route' header
   and extract the original AOR for use in its connection tuple
   comparison, as defined in the connection re-use draft [11].  An
   example which builds on this scenario (showing an inbound request to
   the AOR) is detailed in section 4.1.4.2 of this document.

4.1.3.  Initiating a Session

   This section covers basic SIP signaling when initiating a call from
   behind a NAT.




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4.1.3.1.  UDP

   Initiating a call using UDP.



      Client              NAT               Proxy              [..]
        |                  |                  |
        |(1) INVITE        |                  |                 |
        |----------------->|                  |                 |
        |                  |(1) INVITE        |                 |
        |                  |----------------->|                 |
        |                  |(2) 407 Unauth    |                 |
        |                  |<-----------------|                 |
        |(2) 407 Unauth    |                  |                 |
        |<-----------------|                  |                 |
        |(3) INVITE        |                  |                 |
        |                  |(3) INVITE        |                 |
        |                  |----------------->|                 |
        |                  |                  |(4) INVITE       |
        |                  |                  |---------------->|
        |                  |                  |(5)180 RINGING   |
        |                  |                  |<----------------|
        |                  |(6)180 RINGING    |                 |
        |                  |<-----------------|                 |
        |(6)180 RINGING    |                  |                 |
        |<-----------------|                  |                 |
        |                  |                  |(7)200 OK        |
        |                  |                  |<----------------|
        |                  |(8)200 OK         |                 |
        |                  |<-----------------|                 |
        |(8)200 OK         |                  |                 |
        |<-----------------|                  |                 |
        |(9)ACK            |                  |                 |
        |----------------->|                  |                 |
        |                  |(9)ACK            |                 |
        |                  |----------------->|                 |
        |                  |                  |(10) ACK         |
        |                  |                  |---------------->|
        |                  |                  |(11)             |

     Figure 8.

   The initiating client generates an INVITE request that is to be sent
   through the NAT to a Proxy server.  The INVITE message is represented
   in Figure 8 by (1) and is as follows:





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   INVITE sip:clientB@example.com SIP/2.0
   Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK74husdHG
   Max-Forwards: 70
   Route: <sip:proxy.example.com;lr>
   From: clientA <sip:clientA@example.com>;tag=7skjdf38l
   To: clientB <sip:clientB@example.com>
   Call-ID: 8327468763423@example.com
   CSeq: 1 INVITE
   Contact: <sip:im_a_gruu@proxy.example.com>
   Content-Type: application/sdp
   Content-Length: ..

   [SDP not shown]

   There are a number of points to note with this message:
   1.  Firstly, as with the registration example in section 4.1.1.1,
       reponses to this request will not automatically pass back through
       a NAT and so the SIP 'Via' header 'rport' is included as
       described in the 'Symmetric response' section(3.1.1) and defined
       in RFC 3581 [6].
   2.  Secondly, the contact inserted contains the GRUU previously
       obtained from the registration.
   3.  [Editors Note: TODO - Expand description of GRUU and connection
       re-use]

4.1.3.2.  Reliable Transport

   [Editors note: TODO]

4.1.4.  Receiving an Invitation to a Session

   This section details scenarios where a client behind a NAT receives
   an inbound request through the NAT.  These scenarios build on the
   previous registration scenario from sections 4.1.1 and 4.1.2 in this
   document.

4.1.4.1.  Registrar/Proxy Co-located

   The core SIP signaling associated with this call flow is not impacted
   directly by the transport protocol and so only one example scenario
   is necessary.  The example uses UDP and follows on from the
   registration installed in the example from section 4.1.1.1.









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       Client              NAT         Registrar/Proxy       SIP Entity
         |                  |                  |                 |
         |*******************************************************|
         |           Registration Binding Installed in           |
         |                    section 4.1.1.1                    |
         |*******************************************************|
         |                  |                  |                 |
         |                  |                  |(1)INVITE        |
         |                  |                  |<----------------|
         |                  |(2)INVITE         |                 |
         |                  |<-----------------|                 |
         |(2)INVITE         |                  |                 |
         |<-----------------|                  |                 |
         |                  |                  |                 |
         |                  |                  |                 |


     Figure 9.

   The core SIP signaling associated with this call flow is not impacted
   directly by the transport protocol and so only one example scenario
   is necessary.  The example uses UDP and follows on from the
   registration installed in section 4.1.1.1.  An INVITE request arrives
   at the Registrar with a destination pointing to the AOR of that
   inserted in section 4.1.1.1.  The message is illustrated by (1) in
   Figure 9 and looks as follows:


   INVITE sip:client@example.com SIP/2.0
   Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
   Max-Forwards: 70
   From: External <sip:External@external.example.com>;tag=7893hd
   To: client <sip:client@example.com>
   Call-ID: 8793478934897@external.example.com
   CSeq: 1 INVITE
   Contact: <sip:external@192.0.1.4>
   Content-Type: application/sdp
   Content-Length: ..

   [SDP not shown]

   The INVITE matches the registration binding at the Registrar and the
   INVITE request-URI is re-written to the selected onward address.  The
   proxy then examines the request URI of the INVITE and compares with
   its list of current open connections/mappings.  It uses the incoming
   AOR to commence the check for associated open connections/mappings.
   Once matched, the proxy checks to see if the unique instance
   identifier (+sip.instance) associated with the binding equals the



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   same instance identifier associated with the binding.  If more than
   one result is matched, the lowest 'connectionID' Contact parameter
   will be used.  This is message (2) from Figure 9 and is as follows:


   INVITE sip:sip:client@client.example.com SIP/2.0
   Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4kmlds893jhsd
   Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
   Max-Forwards: 70
   From: External <sip:External@external.example.com>;tag=7893hd
   To: client <sip:client@example.com>
   Call-ID: 8793478934897@external.example.com
   CSeq: 1 INVITE
   Contact: <sip:external@192.0.1.4>
   Content-Type: application/sdp
   Content-Length: ..

   [SDP not shown]

   It is a standard SIP INVITE request with no additional functionality.
   The major difference being that this request will not follow the
   address specified in the Request-URI, as standard SIP rules would
   enforce but will be sent on the connection/mapping associated with
   the registration binding.  This then allows the original connection/
   mapping from the initial registration process to be re-used.

4.1.4.2.  Registrar/Proxy Not Co-located



   Client          NAT    Proxy      Registrar       SIP Entity
     |              |              |              |              |
     |***********************************************************|
     |            Registration Binding Installed in              |
     |                      section 4.1.2                        |
     |***********************************************************|
     |              |              |              |(1)INVITE     |
     |              |              |              |<-------------|
     |              |              |(2)INVITE     |              |
     |              |              |<-------------|              |
     |              |(3)INVITE     |              |              |
     |              |<-------------|              |              |
     |(3)INVITE     |              |              |              |
     |<-------------|              |              |              |
     |              |              |              |              |
     |              |              |              |              |





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     Figure 9.

4.2.  Basic NAT Media Traversal

   This section provides example scenarios to demonstrate basic media
   traversal using the techniques outlined earlier in this document.

4.2.1.  Port Restricted Cone NAT

   This section demonstrates an example of a client both initiating and
   receiving calls behind a 'Restricted Cone' NAT.  The examples have
   been included to represent both 'Restricted' and 'Port Restricted'
   NAT media traversal.  An example is included for both STUN and ICE
   with ICE being the RECOMENDED method.

4.2.1.1.  STUN Solution

   It is possible to traverse media through a 'Restricted Cone NAT'
   using STUN.

4.2.1.1.1.  Initiating Session

   The following example demonstrates media traversal through a
   'Restricted Cone' NAT using STUN.  It is assumed in this example that
   the STUN client and SIP Client are co-located on the same machine.
   Note that some SIP signalling messages have been left out for
   simplicity.



     Client              NAT               STUN                [..]
                                          Server
       |                  |                  |                  |
       |(1) STUN Req      |                  |                  |
       |src=10.0.1.1:5301 |                  |                  |
       |----------------->|                  |                  |
       |                  |(2) STUN Req      |                  |
       |                  |src=1.2.3.4:5601  |                  |
       |                  |----------------->|                  |
       |                  |(3) STUN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=1.2.3.4:5601  |                  |
       |                  |dest=1.2.3.4:5601 |                  |
       |(4) STUN Resp     |                  |                  |
       |<-----------------|                  |                  |
       |map=1.2.3.4:5601  |                  |                  |
       |dest=10.0.1.1:5301|                  |                  |
       |(5) STUN Req      |                  |                  |



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       |src=10.0.1.1:5302 |                  |                  |
       |----------------->|                  |                  |
       |                  |(6) STUN Req      |                  |
       |                  |src=1.2.3.4:5608  |                  |
       |                  |----------------->|                  |
       |                  |(7) STUN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=1.2.3.4:5608  |                  |
       |                  |dest=1.2.3.4:5608 |                  |
       |(8) STUN Resp     |                  |                  |
       |<-----------------|                  |                  |
       |map=1.2.3.4:5608  |                  |                  |
       |dest=10.0.1.1:5302|                  |                  |
       |(9)SIP INVITE     |                  |                  |
       |----------------->|                  |                  |
       |                  |(10)SIP INVITE    |                  |
       |                  |------------------------------------>|
       |                  |                  |(11)SIP 200 OK    |
       |                  |<------------------------------------|
       |(12)SIP 200 OK    |                  |                  |
       |<-----------------|                  |                  |
       |========================================================|
       |>>>>>>>>>>>Outgoing Media sent to 1.2.3.4:5601>>>>>>>>>>|
       |========================================================|
       |========================================================|
       |<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<|
       |========================================================|
       |(13)SIP ACK       |                  |                  |
       |----------------->|                  |                  |
       |                  |(14) SIP ACK      |                  |
       |                  |------------------------------------>|
       |                  |                  |                  |


   Figure 18: Restricted NAT with STUN - Initiating

   o  On deciding to initiate a SIP voice session the VOIP client starts
      a local STUN client.  The STUN client generates a standard STUN
      request as indicated in (1) from Figure 18 which also highlights
      the source address and port for which the client device wishes to
      obtain a mapping.  The STUN request is sent through the NAT
      towards the public internet.
   o  STUN message (2) traverses the NAT and breaks out onto the public
      internet towards the public STUN server.  Note that the source
      address of the STUN requests now represents the public address and
      port from the public side of the NAT.





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   o
   o  The STUN server receives the request and processes it
      appropriately.  This results in a successful STUN response being
      generated and returned (3).  The message contains details of the
      mapped public address (contained in the STUN MAPPED-ADDRESS
      attribute) which is to be used by the originating client to
      receive media (see 'map=' from (3)).
   o  The STUN response traverses back through the NAT using the binding
      created by the STUN request and presents the new mapped address to
      the client (4).  At this point the process is repeated to obtain a
      second mapped address (as shown in (5)-(8)) for an alternative
      local address (local port has now changed from 5301 to 5302 in
      (5)).
   o  The client now constructs a SIP INVITE message(9).  Note that
      traversal of SIP is not covered in this example and is discussed
      in earlier sections of the document.  The INVITE request will use
      the addresses it has obtained in the previous STUN transactions to
      populate the SDP of the SIP INVITE as shown below:

      v=0
      o=test 2890844526 2890842807 IN IP4 10.0.1.1
      c=IN IP4 1.2.3.4
      t=0 0
      m=audio 5601 RTP/AVP 0
      a=rtcp:5608


   o  Note that the mapped address obtained from the STUN transactions
      are inserted as the connection address for the SDP (c=1.2.3.4).
      The Primary port for RTP is also inserted in the SDP (m=audio 5601
      RTP/AVP 0).  Finally, the port gained from the additional STUN
      binding is placed in the RTCP attribute (as discussed in
      Section 3.2.5) for traversal of RTCP (a=rtcp:5608).
   o  The SIP signalling then traverses the NAT and sets up the SIP
      session (10-12).  Note that the client transmits media as soon as
      the 200 OK to the INVITE arrives at the client (12).  Up until
      this point the incoming media will not pass through the NAT as no
      outbound association has been created with the far end client.
      Two way media communication has now been established.

4.2.1.1.2.  Receiving Session Invitation

   Receiving a session for a 'Restricted Cone' NAT using STUN is very
   similar to the example outlined in Section 4.2.1.1.1.  Figure 20
   illustrates the associated flow of messages.






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     Client              NAT               STUN                [..]
                                          Server
       |                  |                  | (1)SIP INVITE    |
       |                  |<-----------------|------------------|
       |(2) SIP INVITE    |                  |                  |
       |<-----------------|                  |                  |
       |                  |                  |                  |
       |(3) STUN Req      |                  |                  |
       |src=10.0.1.1:5301 |                  |                  |
       |----------------->|                  |                  |
       |                  |(4) STUN Req      |                  |
       |                  |src=1.2.3.4:5601  |                  |
       |                  |----------------->|                  |
       |                  |(5) STUN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=1.2.3.4:5601  |                  |
       |                  |dest=1.2.3.4:5601 |                  |
       |(6) STUN Resp     |                  |                  |
       |<-----------------|                  |                  |
       |map=1.2.3.4:5601  |                  |                  |
       |dest=10.0.1.1:5301|                  |                  |
       |(7) STUN Req      |                  |                  |
       |src=10.0.1.1:5302 |                  |                  |
       |----------------->|                  |                  |
       |                  |(8) STUN Req      |                  |
       |                  |src=1.2.3.4:5608  |                  |
       |                  |----------------->|                  |
       |                  |(9) STUN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=1.2.3.4:5608  |                  |
       |                  |dest=1.2.3.4:5608 |                  |
       |(10) STUN Resp    |                  |                  |
       |<-----------------|                  |                  |
       |map=1.2.3.4:5608  |                  |                  |
       |dest=10.0.1.1:5302|                  |                  |
       |(11)SIP 200 OK    |                  |                  |
       |----------------->|                  |                  |
       |                  |(12)SIP 200 OK    |                  |
       |                  |------------------------------------>|
       |========================================================|
       |>>>>>>>>>>>Outgoing Media sent to 1.2.3.4:5601>>>>>>>>>>|
       |========================================================|
       |========================================================|
       |<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<|
       |========================================================|
       |                  |                  |(13)SIP ACK       |
       |                  |<------------------------------------|
       |(14)SIP ACK       |                  |                  |



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       |<-----------------|                  |                  |
       |                  |                  |                  |


   Figure 20: Restricted NAT with STUN - Receiving

   o  On receiving an invitation to a SIP voice session the VOIP client
      starts a local STUN client.  The STUN client generates a standard
      STUN request as indicated in (3) from Figure 20 which also
      highlights the source address and port for which the client device
      wishes to obtain a mapping.  The STUN request is sent through the
      NAT towards the public internet.
   o  STUN message (4) traverses the NAT and breaks out onto the public
      internet towards the public STUN server.  Note that the source
      address of the STUN requests now represents the public address and
      port from the public side of the NAT.
   o
   o  The STUN server receives the request and processes it
      appropriately.  This results in a successful STUN response being
      generated and returned (5).  The message contains details of the
      mapped public address (contained in the STUN MAPPED-ADDRESS
      attribute) which is to be used by the originating client to
      receive media (see 'map=' from (5)).
   o  The STUN response traverses back through the NAT using the binding
      created by the STUN request and presents the new mapped address to
      the client (6).  At this point the process is repeated to obtain a
      second mapped address (as shown in (7)-(10)) for an alternative
      local address (local port has now changed from 5301 to 5302 in
      (7)).
   o  The client now constructs a SIP 200 OK message (11).  Note that
      traversal of SIP is not covered in this example and is discussed
      in earlier sections of the document.  The 200 OK response will use
      the addresses it has obtained in the previous STUN transactions to
      populate the SDP of the SIP INVITE as shown below:

      v=0
      o=test 2890844526 2890842807 IN IP4 10.0.1.1
      c=IN IP4 1.2.3.4
      t=0 0
      m=audio 5601 RTP/AVP 0
      a=rtcp:5608


   o  Note that the mapped address obtained from the initial STUN
      transaction is inserted as the connection address for the SDP
      (c=1.2.3.4).  The Primary port for RTP is also inserted in the SDP
      (m=audio 5601 RTP/AVP 0).  Finally, the port gained from the
      additional binding is placed in the RTCP attribute (as discussed



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      in Section 3.2.5) for traversal of RTCP (a=rtcp:5608).
   o  The SIP signalling then traverses the NAT and sets up the SIP
      session (11-14).  Note that the client transmits media as soon as
      the 200 OK to the INVITE is sent to the UAC(11).  Up until this
      point the incoming media will not pass through the NAT as no
      outbound association has been created with the far end client.
      Two way media communication has now been established.

4.2.1.2.  ICE Solution

   The preferred solution for media traversal of NAT is using ICE, as
   described in Section 3.2.4.  The following examples illustrate the
   traversal of a 'Port Restricted Cone' NAT for both an initiating and
   receiving client.  The example only covers ICE in association with
   STUN and TURN.

4.2.1.2.1.  Initiating Session

   The following example demonstrates an initiating traversal through a
   'Restricted Cone' NAT using ICE.



     Client              NAT               STUN              TURN      *
                                          Server            Server
       |                  |                  |                 |       |
       |(1) STUN Req      |                  |                 |       |
       |src=10.0.1.1:5301 |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(2) STUN Req      |                 |       |
       |                  |src=1.2.3.4:5601  |                 |       |
       |                  |----------------->|                 |       |
       |                  |(3) STUN Resp     |                 |       |
       |                  |<-----------------|                 |       |
       |                  |map=1.2.3.4:5601  |                 |       |
       |                  |dest=1.2.3.4:5601 |                 |       |
       |(4) STUN Resp     |                  |                 |       |
       |<-----------------|                  |                 |       |
       |map=1.2.3.4:5601  |                  |                 |       |
       |dest=10.0.1.1:5301|                  |                 |       |
       |(5) STUN Req      |                  |                 |       |
       |src=10.0.1.1:5311 |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(6) STUN Req      |                 |       |
       |                  |src=1.2.3.4:5611  |                 |       |
       |                  |----------------->|                 |       |
       |                  |(7) STUN Resp     |                 |       |
       |                  |<-----------------|                 |       |



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       |                  |map=1.2.3.4:5611  |                 |       |
       |                  |dest=1.2.3.4:5611 |                 |       |
       |(8) STUN Resp     |                  |                 |       |
       |<-----------------|                  |                 |       |
       |map=1.2.3.4:5611  |                  |                 |       |
       |dest=10.0.1.1:5311|                  |                 |       |
       |(9) TURN Allocate |                  |                 |       |
       |src=10.0.1.1:5302 |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(10) TURN Allocate|                 |       |
       |                  |src=1.2.3.4:5608  |                 |       |
       |                  |----------------------------------->|       |
       |                  |(11) TURN Resp    |                 |       |
       |                  |<-----------------------------------|       |
       |                  |map=2.3.4.5:5608  |                 |       |
       |                  |dest=1.2.3.4:5608 |                 |       |
       |(12) TURN Resp    |                  |                 |       |
       |<-----------------|                  |                 |       |
       |map=2.3.4.5:5608  |                  |                 |       |
       |dest=10.0.1.1:5302|                  |                 |       |
       |(13) TURN Allocate|                  |                 |       |
       |src=10.0.1.1:5312 |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(14) TURN Allocate|                 |       |
       |                  |src=1.2.3.4:5618  |                 |       |
       |                  |----------------------------------->|       |
       |                  |(15) TURN Resp    |                 |       |
       |                  |<-----------------------------------|       |
       |                  |map=2.3.4.5:5618  |                 |       |
       |                  |dest=1.2.3.4:5618 |                 |       |
       |(16) TURN Resp    |                  |                 |       |
       |<-----------------|                  |                 |       |
       |map=2.3.4.5:5618  |                  |                 |       |
       |dest=10.0.1.1:5312|                  |                 |       |
       |(17)SIP INVITE    |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(18)SIP INVITE    |                 |       |
       |                  |------------------------------------------->|
       |                  |                  |(19)SIP 200 OK   |       |
       |                  |<-------------------------------------------|
       |(20)SIP 200 OK    |                  |                 |       |
       |<-----------------|                  |                 |       |
       |(21)STUN Req      |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(22) STUN Req     |                 |       |
       |                  |------------------------------------------->|
       |                  |                  |(23)STUN Resp    |       |
       |                  |<-------------------------------------------|



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       |(24)STUN Resp     |                  |                 |       |
       |<-----------------|                  |                 |       |
       |===============================================================|
       |>>>>>>>>>>>Outgoing Media sent from 10.1.1.1:5301>>>>>>>>>>>>>>|
       |===============================================================|
       |                  |                  |(25) STUN Req    |       |
       |                  |<-------------------------------------------|
       |(26)STUN Req      |                  |                 |       |
       |<-----------------|                  |                 |       |
       |(27)STUN Resp     |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |                  |(28)STUN Resp    |       |
       |                  |------------------------------------------->|
       |===============================================================|
       |<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<<<<<<<<|
       |===============================================================|
       |(29)SIP ACK       |                  |                 |       |
       |----------------->|                  |                 |       |
       |                  |(30) SIP ACK      |                 |       |
       |                  |------------------------------------------->|
       |                  |                  |                 |       |


   Figure 22: Restricted NAT with ICE - Initiating

   o  On deciding to initiate a SIP voice session the VOIP client starts
      a local STUN and TURN client.  The STUN client generates a
      standard STUN request as indicated in (1) from Figure 22 which
      also highlights the source address and port for which the client
      device wishes to obtain a mapping.  The STUN request is sent
      through the NAT towards the public internet.
   o  STUN message (2) traverses the NAT and breaks out onto the public
      internet towards the public STUN server.  Note that the source
      address of the STUN requests now represents the public address and
      port from the public side of the NAT.
   o
   o  The STUN server receives the request and processes appropriately.
      This results in a successful STUN response being generated and
      returned (3).  The message contains details of the mapped public
      address (contained in the STUN MAPPED-ADDRESS attribute) which is
      to be used by the originating client to receive media (see 'map=')
      from (3)).
   o  The STUN response traverses back through the NAT using the binding
      created by the STUN request and presents the new mapped address to
      the client (4).  The process is repeated and a second STUN derived
      address is obtained, as illustrated in (5)-(8) in Figure 22.
      While the STUN client is obtaining addresses, the TURN client will
      also be attempting to obtain external representations.  The TURN



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      Allocate message is constructed in association with the local IP
      address and port combination (9).  The TURN Allocate message is
      then sent from the client to the external TURN server via the NAT
      (10).  The TURN server processes the Allocate request and returns
      an appropriate response(11).  The response contains the 'Mapped-
      Address'(defined in STUN specification) attribute which contains
      the external representation that the TURN server will provide for
      the internal mapping.  The TURN response then traverses back
      through the NAT and returns the newly allocated external
      representation to the originating client(12).The process is
      repeated and a second TURN derived address is obtained, as
      illustrated in (13)-(16) in Figure 22.  At this point the client
      behind the NAT has a pair of STUN external representations and
      TURN equivalents.  The client would be free to gather any number
      of external representations using any UNSAF[9] compliant protocol.
   o  The client now constructs a SIP INVITE message (17).  The INVITE
      request will use the addresses it has obtained in the previous
      STUN/TURN interactions to populate the SDP of the SIP INVITE.
      This should be carried out in accordance with the semantics
      defined in the ICE specification[15], as shown below in Figure 23
      (*note - /* signifies line continuation):

      v=0
      o=test 2890844526 2890842807 IN IP4 10.0.1.1
      c=IN IP4 1.2.3.4
      t=0 0
      m=audio 5601 RTP/AVP 0
      a=candidate:H83jksd 1.0 rtp_uname_frag_1 rtp_pass_1 1.2.3.4 5601
        /* rtcp_uname_frag_1 rtcp_pass_1 1.2.3.4 5611
      a=candidate:Hye73hd 0.8 rtp_uname_frag_2 rtp_pass_2 1.2.3.4 5608
        /* rtcp_uname_frag_2 rtcp_pass_2 1.2.3.4 5618
      a=candidate:H82hjjh 0.5 rtp_uname_frag_3 rtp_pass_3 1.2.3.4 5600


      Figure 23: ICE SDP

   o  The SDP has been constructed to include all the available
      addresses that have been assembled.  The first 'candidate' address
      contains the two STUN derived addresses for both RTP and RTCP
      traffic.  This entry has been given the highest priority (1.0) by
      the client and also inserted as the default address.
   o  The second 'candidate' address contains the two TURN derived
      addresses for both RTP and RTCP traffic.  This entry has been
      given the second highest priority (0.8).
   o  The third and final 'candidate' address contains a local interface
      address that has not been derived externally.  This entry has been
      given the lowest priority (0.5).




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   o  The SIP signalling then traverses the NAT and sets up the SIP
      session (18)-(20).  On advertising a candidate address, the client
      should have a local STUN server running on each advertised
      candidate address.  This is for the purpose of responding to
      incoming connectivity checks.  In this example, after sending the
      INVITE and receiving a 200 OK, the client initiates an outgoing
      STUN connectivity check to the selected remote interfaces
      (21)-(24) (*Note - this process will be repeated for every
      advertised address which is not shown in the diagram for
      simplicity).  On receiving a STUN response, the client is able to
      stream media to the remote destination (*Note - if further STUN
      connectivity responses are received after the client has started
      streaming media with a higher priority, it will be used instead).
      The remote destination will also carry out similar STUN
      connectivity checks (25)-(28) which then allows media to be
      streamed to the client behind the NAT using the advertised
      connections.  Two way audio is now possible between the two
      clients.

4.2.1.2.2.  Receiving Session Invitation

   This example is similar to that described in Section 4.2.1.2.1.  The
   client behind a NAT is receiving the incoming ICE Initiate in a SIP
   INVITE request.



     Client              NAT               STUN               TURN     *
                                          Server             Server
       |                  |                  |                  |      |
       |                  |                  |(1)SIP INVITE     |      |
       |                  |<-------------------------------------------|
       |(2)SIP INVITE     |                  |                  |      |
       |<-----------------|                  |                  |      |
       |                  |                  |                  |      |
       |(3) STUN Req      |                  |                  |      |
       |src=10.0.1.1:5301 |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |(4) STUN Req      |                  |      |
       |                  |src=1.2.3.4:5601  |                  |      |
       |                  |----------------->|                  |      |
       |                  |(5) STUN Resp     |                  |      |
       |                  |<-----------------|                  |      |
       |                  |map=1.2.3.4:5601  |                  |      |
       |                  |dest=1.2.3.4:5601 |                  |      |
       |(6) STUN Resp     |                  |                  |      |
       |<-----------------|                  |                  |      |
       |map=1.2.3.4:5601  |                  |                  |      |



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       |dest=10.0.1.1:5301|                  |                  |      |
       |(7) STUN Req      |                  |                  |      |
       |src=10.0.1.1:5311 |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |(8) STUN Req      |                  |      |
       |                  |src=1.2.3.4:5611  |                  |      |
       |                  |----------------->|                  |      |
       |                  |(9) STUN Resp     |                  |      |
       |                  |<-----------------|                  |      |
       |                  |map=1.2.3.4:5611  |                  |      |
       |                  |dest=1.2.3.4:5611 |                  |      |
       |(10) STUN Resp    |                  |                  |      |
       |<-----------------|                  |                  |      |
       |map=1.2.3.4:5611  |                  |                  |      |
       |dest=10.0.1.1:5311|                  |                  |      |
       |(11) TURN Allocate|                  |                  |      |
       |src=10.0.1.1:5302 |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |(12) TURN Allocate|                  |      |
       |                  |src=1.2.3.4:5608  |                  |      |
       |                  |------------------------------------>|      |
       |                  |(13) TURN Resp    |                  |      |
       |                  |<------------------------------------|      |
       |                  |map=2.3.4.5:5608  |                  |      |
       |                  |dest=1.2.3.4:5608 |                  |      |
       |(14) TURN Resp    |                  |                  |      |
       |<-----------------|                  |                  |      |
       |map=2.3.4.5:5608  |                  |                  |      |
       |dest=10.0.1.1:5302|                  |                  |      |
       |(15) TURN Allocate|                  |                  |      |
       |src=10.0.1.1:5312 |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |(16) TURN Allocate|                  |      |
       |                  |src=1.2.3.4:5618  |                  |      |
       |                  |------------------------------------>|      |
       |                  |(17) TURN Resp    |                  |      |
       |                  |<------------------------------------|      |
       |                  |map=2.3.4.5:5618  |                  |      |
       |                  |dest=1.2.3.4:5618 |                  |      |
       |(18) TURN Resp    |                  |                  |      |
       |<-----------------|                  |                  |      |
       |map=2.3.4.5:5618  |                  |                  |      |
       |dest=10.0.1.1:5312|                  |                  |      |
       |(19)SIP 200 OK    |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |(20)SIP 200 OK    |                  |      |
       |                  |------------------------------------------->|
       |(21)STUN Req      |                  |                  |      |



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       |----------------->|                  |                  |      |
       |                  |(22) STUN Req     |                  |      |
       |                  |------------------------------------------->|
       |                  |                  |(23)STUN Resp     |      |
       |                  |<-------------------------------------------|
       |(24)STUN Resp     |                  |                  |      |
       |<-----------------|                  |                  |      |
       |===============================================================|
       |>>>>>>>>>>>Outgoing Media sent from 10.1.1.1:5301>>>>>>>>>>>>>>|
       |===============================================================|
       |                  |                  |(25) STUN Req     |      |
       |                  |<-------------------------------------------|
       |(26)STUN Req      |                  |                  |      |
       |<-----------------|                  |                  |      |
       |(27)STUN Resp     |                  |                  |      |
       |----------------->|                  |                  |      |
       |                  |                  |(28)STUN Resp     |      |
       |                  |------------------------------------------->|
       |===============================================================|
       |<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<<<<<<<<|
       |===============================================================|
       |                  |                  |(29)SIP ACK       |      |
       |                  |<-------------------------------------------|
       |(30)SIP ACK       |                  |                  |      |
       |<-----------------|                  |                  |      |
       |                  |                  |                  |      |


   Figure 24: Restricted NAT with ICE - Receiving

   o  As mentioned previously, this example is similar to that described
      in Section 4.2.1.2.1.  For this reason, some of the description
      may reference the previous example.  The scenario starts with the
      client behind the NAT receiving a SIP INVITE(1) reques (ICE
      initiate message).
   o  On receiving the SIP INVITE the client is able to collect all
      possible addresses available for media interaction (e.g.  Local
      addresses, STUN derived, TURN derived).  See detail from
      Section 4.2.1.2.1 for explanation on accumulating all possible
      media addresses (Steps (3)-(18) in Figure 24).
   o  The client will perform connectivity checks on all addresses
      received in the SIP INVITE message(21)-(24).  Note that steps
      (21)-(24) will be repeated for every address offered in the SIP
      INVITE request.  This is not shown in the diagram for simplicity.
      On receiving a response to a STUN connectivity check, the client
      will start streaming media (*Note - if further STUN connectivity
      responses are received after the client has started streaming meda
      with a higher priority, it will be used instead).



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   o  The STUN connectivity checks will then occur in the opposite
      direction, as illustrated in Section 4.2.1.2.1.  A STUN server
      running on each advertised address will respond to incoming STUN
      connectivity requests(25)-(28).
   o  Bi-directional audio can now occur between the two clients.

4.2.2.  Symmetric NAT

4.2.2.1.  STUN Failure

   This section highlights that while STUN is the preferred mechanism
   for traversal of NAT, it does not solve every cases.  The use of STUN
   on its own will not guarantee traversal through every NAT type, hence
   the recommendation that ICE be the prefered option.





































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     Client            SYMMETRIC           STUN                [..]
                         NAT              Server
       |                  |                  |                  |
       |(1) STUN Req      |                  |                  |
       |src=10.0.1.1:5301 |                  |                  |
       |----------------->|                  |                  |
       |                  |(2) STUN Req      |                  |
       |                  |src=1.2.3.4:5601  |                  |
       |                  |----------------->|                  |
       |                  |(3) STUN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=1.2.3.4:5601  |                  |
       |                  |dest=1.2.3.4:5601 |                  |
       |(4) STUN Resp     |                  |                  |
       |<-----------------|                  |                  |
       |map=1.2.3.4:5601  |                  |                  |
       |dest=10.0.1.1:5301|                  |                  |
       |(5)SIP INVITE     |                  |                  |
       |----------------->|                  |                  |
       |                  |(6)SIP INVITE     |                  |
       |                  |------------------------------------>|
       |                  |                   |(7)SIP 200 OK    |
       |                  |<------------------------------------|
       |(8)SIP 200 OK     |                  |                  |
       |<-----------------|                  |                  |
       |========================================================|
       |>>>>>>>>>>>Outgoing Media sent from 10.0.1.1:5301>>>>>>>|
       |========================================================|
       |                  x=====================================|
       |                  xIncoming Media sent to 1.2.3.4:5601<<|
       |                  x=====================================|
       |(9)SIP ACK        |                  |                  |
       |----------------->|                  |                  |
       |                  |(10) SIP ACK      |                  |
       |                  |------------------------------------>|
       |                  |                  |                  |


   Figure 25: Symmetric NAT with STUN - Failure

   The example in Figure 25 is conveyed in the context of the client
   behind the Symmetric NAT initiating a call.  It should be noted that
   the same problem applies when a client receives a SIP invitation and
   is behind a Symmetric NAT.
   o  In Figure 25 the client behind the NAT obtains an external
      representation using standard STUN mechanisms (1)-(4) that have
      been used in previous examples in this document (e.g
      Section 4.2.1.1.1).



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   o  The external mapped address obtained is also used in the outgoing
      SDP contained in the SIP INVITE request(5).
   o  In this example the client is still able to send media to the
      external client.  The problem occurs when the client outside the
      NAT tries to use the address supplied in the outgoing INVITE
      request to traverse media back through the Symmetric NAT.
   o  A symmetric NAT has differing rules from the Cone variety of NAT.
      For any internal IP address and port mapping, data sent to
      different external addresses does not provide the same public
      mapping at the NAT.  In Figure 25 the STUN query produced a valid
      external mapping.  This mapping, however, can only be used in the
      context of the original STUN request that was sent to the STUN
      server.  Any packets that attempt to use the mapped address, that
      does not come from the STUN server IP address and port, will be
      dropped at the NAT.  Figure 25 shows the media being dropped at
      the NAT after (8).

4.2.2.2.  TURN Solution

   As identified in Section 4.2.2.1, STUN provides a useful tool for the
   traversal of the majority of NATs but fails with symmetric type NAT.
   This led to the development of the TURN solution[13] which introduces
   a media relay in the path for NAT traversal (as described in
   Section 3.2.3).  The following example explains how TURN solves the
   previous failure when using STUN to traverse a symmetric NAT.


























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     Client            SYMMETRIC           TURN                [..]
                         NAT              Server
       |                  |                  |                  |
       |(1) TURN Allocate |                  |                  |
       |src=10.0.1.1:5301 |                  |                  |
       |----------------->|                  |                  |
       |                  |(2) TURN Allocate |                  |
       |                  |src=1.2.3.4:5601  |                  |
       |                  |----------------->|                  |
       |                  |(3) TURN Resp     |                  |
       |                  |<-----------------|                  |
       |                  |map=2.3.4.5:5601  |                  |
       |                  |dest=1.2.3.4:5601 |                  |
       |(4) TURN Resp     |                  |                  |
       |<-----------------|                  |                  |
       |map=2.3.4.5:5601  |                  |                  |
       |dest=10.0.1.1:5301|                  |                  |
       |(5)SIP INVITE     |                  |                  |
       |----------------->|                  |                  |
       |                  |(6)SIP INVITE     |                  |
       |                  |------------------------------------>|
       |                  |                   |(7)SIP 200 OK    |
       |                  |<------------------------------------|
       |(8)SIP 200 OK     |                  |                  |
       |<-----------------|                  |                  |
       |========================================================|
       |>>>>>>>>>>>Outgoing Media sent from 10.0.1.1:5301>>>>>>>|
       |========================================================|
       |                  |                  |==================|
       |                  |                  |<<<Media Sent to<<|
       |                  |                  |<<< 2.3.4.5:5601<<|
       |                  |                  |==================|
       |=====================================|                  |
       |<Incoming Media Sent to 1.2.3.4:5601<|                  |
       |=====================================|                  |
       |(9)SIP ACK        |                  |                  |
       |----------------->|                  |                  |
       |                  |(10) SIP ACK      |                  |
       |                  |------------------------------------>|
       |                  |                  |                  |


   Figure 26: Symmetric NAT with TURN - Success

   o  The client obtains a TURN derived address by issuing a TURN
      allocate request(1).  The request traverses through the symmetric
      NAT and reaches the TURN server (2).  The Turn server generates a
      response that contains an external representation.  The



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      representation maps to an address mapping on the TURN server which
      is bound to the public pin hole in the NAT, opened by the TURN
      request.  This results in any traffic being sent to the TURN
      server representation (2.3.4.5:5601) will be redirected to the
      external representation of the pin hole created by the TURN
      request(1.2.3.4:5601).
   o  The TURN derived address (2.3.4.5:5601) arrives back at the
      originating client(4).  This address can then be used in the SDP
      for the outgoing SIP INVITE request as shown below (note that the
      RTCP attribute would have been obtained by another TURN derived
      address which is not shown in the call flow for simplicity):
   o

      v=0
      o=test 2890844342 2890842164 IN IP4 10.0.1.1
      c=IN IP4 2.3.4.5
      t=0 0
      m=audio 5601 RTP/AVP 0
      a=rtcp:5608


   o  On receiving the INVITE request, the UAS is able to stream media
      to the TURN derived address (2.3.4.5:5601).  As shown in
      Figure 26, the media from the UAS is directed to the TURN derived
      address at the TURN server.  The TURN server then redirects the
      traffic to the open pin hole in the symmetric NAT (1.2.3.4:5601).
      The media traffic is then able to traverse the symmetric NAT and
      arrives back at the client.
   o  The TURN solution on its own will work for Symmetric and other
      types of NAT mentioned in this specification but should only be
      used as a last resort.  The relaying of media through an external
      entity is not an efficient mechanism for all NAT traversal.

4.2.2.3.  ICE Solution

   The previous two examples have highlighted the problem with using
   STUN for all forms of NAT traversal and a solution using TURN for the
   symmetric NAT case.  As mentioned previously in this document, the
   RECOMMENDED mechanism for traversing all varieties of NAT is using
   ICE, as detailed in Section 3.2.4.  Ice makes use of STUN, TURN and
   any other UNSAF [9] compliant protocol to provide a list of
   prioritised addresses that can be used for media traffic.  Detailed
   examples of ICE can be found in Section 4.2.1.2.1 and in
   Section 4.2.1.2.2.  These examples are associated with a 'Port
   Restricted' type NAT but can be applied to any NAT type variation,
   including 'Symmetric' type NAT.  The procedures are the same and of
   the list of candidate addresses, a client will choose where to send
   media dependant on the results of the STUN connectivity checks on



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   each candidate address and the associated priority (highest priority
   wins).  For more information see the core ICE specification[15]

4.3.  Advanced NAT media Traversal Using ICE

4.3.1.  Full Cone --> Full Cone traversal

4.3.1.1.  Without NAT

4.3.1.1.1.  Initiating Session

4.3.1.1.2.  Receiving Session Invitation

4.3.1.2.  With NAT

4.3.1.2.1.  Initiating Session

4.3.1.2.2.  Receiving Session Invitation

4.3.2.  Port Restricted Cone --> Port Restricted Cone traversal

4.3.2.1.  Without NAT

4.3.2.1.1.  Initiating Session

4.3.2.1.2.  Receiving Session Invitation

4.3.2.2.  With NAT

4.3.2.2.1.  Initiating Session

4.3.2.2.2.  Receiving Session Invitation

4.3.3.  Internal TURN Server (Enterprise Deployment)

4.3.3.1.  Peer in same Enterprise

4.3.3.2.  Peer in same Enterprise - Separated by NAT

4.3.3.3.  Peer outside Enterprise

4.4.  Intercepting Intermediary (B2BUA)

4.5.  IPv4-IPv6 Transition

   This section describes how IPv6-only SIP user agents can communicate
   with IPv4-only SIP user agents.




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4.5.1.  IPv4-IPv6 Transition for SIP Signalling

   IPv4-IPv6 translations at the SIP level usually take place at dual-
   stack proxies that have both IPv4 and IPv6 DNS entries.  Since this
   translations do not involve NATs that are placed in the middle of two
   SIP entities, they fall outside the scope of this document.  A
   detailed description of this type of translation can be found in [18]

4.5.2.  IPv4-IPv6 Transition for Media

   Figure 28 shows a network of IPv6 SIP user agents that has a relay
   with a pool of public IPv4 addresses.  The IPv6 SIP user agents of
   this IPv6 network need to communicate with users on the IPv4
   Internet.  To do so, the IPv6 SIP user agents use TURN to obtain a
   public IPv4 address from the relay.  The mechanism that an IPv6 SIP
   user agent follows to obtain a public IPv4 address from a relay using
   TURN is the same as the one followed by a user agent with a private
   IPv4 address to obtain a public IPv4 address.  The example in
   Figure 29 explains how to use TURN to obtain an IPv4 address and how
   to use the ANAT semantics [17] of the SDP grouping framework [8] to
   provide both IPv4 and IPv6 addresses for a particular media stream.




                             +----------+
                             |   /  \   |
                                /SIP \
                               /Phone \
                              /        \
                             ------------

           IPv4 Network
                             192.0.2.0/8
                             +---------+
                             |         |
       ----------------------|   NAT   |--------------------------
                             |         |
                             +---------+
           IPv6 Network

                                     ++
                                     ||
                               +-----++
                               | IPv6 |
                               | SIP  |
                               | user |
                               | agent|



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                               +------+


   Figure 28: IPv6-IPv4 transition scenario


    IPv6 SIP                               TURN              IPv4 SIP
   User Agent                             Server            User Agent
       |                                     |                  |
       |         (1) TURN Allocate           |                  |
       |         src=[2001:DB8::1]:30000     |                  |
       |------------------------------------>|                  |
       |         (2) TURN Resp               |                  |
       |         map=192.0.2.2:25000         |                  |
       |         dest=[2001:DB8::1]:30000    |                  |
       |<------------------------------------|                  |
       |         (3) SIP INVITE              |                  |
       |------------------------------------------------------->|
       |         (4) SIP 200 OK              |                  |
       |<-------------------------------------------------------|
       |                                     |                  |
       |=====================================|                  |
       |>>>>>>>>>> Outgoing Media >>>>>>>>>>>|                  |
       |=====================================|                  |
       |                                     |==================|
       |                                     |>>>>>> Media >>>>>|
       |                                     |==================|
       |                                     |                  |
       |                                     |==================|
       |                                     |<<<<<< Media <<<<<|
       |                                     |==================|
       |=====================================|                  |
       |<<<<<<<<<< Outgoing Media <<<<<<<<<<<|                  |
       |=====================================|                  |
       |                                     |                  |
       |         (5) SIP ACK                 |                  |
       |------------------------------------------------------->|
       |                                     |                  |


   Figure 29: IPv6-IPv4 translation with TURN

   o  The IPv6 SIP user agent obtains a TURN-derived IPv4 address by
      issuing a TURN allocate request (1).  The TURN server generates a
      response that contains the public IPv4 address.  This IPv4 address
      maps to the IPv6 source address of the TURN allocate request,
      which the IPv6 address of the SIP user agent.  This results in any
      traffic being sent to the IPv4 address provided by TURN server



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      (192.0.2.2:25000) will be redirected to the IPv6 address of the
      SIP user agent ([2001:DB8::1]:30000).
   o  The TURN-derived address (192.0.2.2:25000) arrives back at the
      originating user agent (2).  This address can then be used in the
      SDP for the outgoing SIP INVITE request.  The user agent builds
      two media lines, one with its IPv6 address and the other with the
      IPv4 address that was just obtained.  The user agent groups both
      media lines using the ANAT semantics as shown below (note that the
      RTCP attribute in the IPv4 media line would have been obtained by
      another TURN-derived address which is not shown in the call flow
      for simplicity).


      v=0
      o=test 2890844342 2890842164 IN IP6 2001:DB8::1
      t=0 0
      a=group:ANAT 1 2
      m=audio 20000 RTP/AVP 0
      c=IN IP6 2001:DB8::1
      a=mid:1
      m=audio 25000 RTP/AVP 0
      c=IN IP4 192.0.2.2
      a=rtcp:25001
      a=mid:2

   o  On receiving the INVITE request, the user agent server rejects the
      IPv6 media line by setting its port to zero in the answer and
      starts sending media to the IPv4 address in the offer.  The IPv6
      user agent sends media through the relay as well, as shown in
      Figure 29.

4.6.  ICE with RTP/TCP

   TODO


5.  Acknowledgments

   The authors would like to thank the members of the IETF SIPPING WG
   for their comments and suggestions.  Detailed comments were provided
   by Francois Audet, kaiduan xie and Hans Persson.


6.  References







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6.1.  Normative References

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

   [2]   Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
         "RTP: A Transport Protocol for Real-Time Applications",
         RFC 1889, January 1996.

   [3]   Handley, M. and V. Jacobson, "SDP: Session Description
         Protocol", RFC 2327, April 1998.

   [4]   Tsirtsis, G. and P. Srisuresh, "Network Address Translation -
         Protocol Translation (NAT-PT)", RFC 2766, February 2000.

   [5]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", RFC 3264, June 2002.

   [6]   Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
         Initiation Protocol (SIP) for Symmetric Response Routing",
         RFC 3581, August 2003.

   [7]   Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
         Extension Header Field for Registering Non-Adjacent Contacts",
         RFC 3327, December 2002.

   [8]   Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,
         "Grouping of Media Lines in the Session Description Protocol
         (SDP)", RFC 3388, December 2002.

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

   [10]  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.

   [11]  Jennings, C. and A. Hawrylyshen, "SIP Conventions for UAs with
         Outbound Only Connections", draft-jennings-sipping-outbound-01
         (work in progress), February 2005.

   [12]  Rosenberg, J., "Obtaining and Using Globally Routable User
         Agent (UA) URIs (GRUU) in the  Session Initiation Protocol
         (SIP)", draft-ietf-sip-gruu-06 (work in progress),
         October 2005.




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   [13]  Rosenberg, J., "Traversal Using Relay NAT (TURN)",
         draft-rosenberg-midcom-turn-08 (work in progress),
         September 2005.

   [14]  Wing, D., "Symmetric RTP and RTCP Considered Helpful",
         draft-wing-mmusic-symmetric-rtprtcp-01 (work in progress),
         October 2004.

   [15]  Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Methodology for Network  Address Translator (NAT) Traversal for
         Offer/Answer Protocols", draft-ietf-mmusic-ice-06 (work in
         progress), October 2005.

   [16]  Rosenberg, J., "Simple Traversal of UDP Through Network Address
         Translators (NAT) (STUN)", draft-ietf-behave-rfc3489bis-02
         (work in progress), July 2005.

   [17]  Camarillo, G., "The Alternative Network Address Types Semantics
         (ANAT) for theSession  Description Protocol (SDP) Grouping
         Framework", draft-ietf-mmusic-anat-02 (work in progress),
         October 2004.

6.2.  Informative References

   [18]  Camarillo, G., "IPv6 Transcition in the Session Initiation
         Protocol (SIP)", draft-camarillo-sipping-v6-transition-00 (work
         in progress), February 2005.
























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Authors' Addresses

   Chris Boulton
   Ubiquity Software Corporation
   Eastern Business Park
   St Mellons
   Cardiff, South Wales  CF3 5EA

   Email: cboulton@ubiquitysoftware.com


   Jonathan Rosenberg
   Cisco Systems
   600 Lanidex Plaza
   Parsippany, NJ  07054

   Email: jdrosen@cisco.com


   Gonzalo Camarillo
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: Gonzalo.Camarillo@ericsson.com

























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