SIPPING Working Group C. Boulton, Ed.
Internet-Draft Avaya
Expires: October 27, 2008 J. Rosenberg
Cisco Systems
G. Camarillo
Ericsson
April 25, 2008
Best Current Practices for NAT Traversal for SIP
draft-ietf-sipping-nat-scenarios-08
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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
documenting corresponding flows.
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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. Re-use of Connections . . . . . . . . . . . . . . . . 8
3.2. Media Traversal . . . . . . . . . . . . . . . . . . . . . 8
3.2.1. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . 9
3.2.2. STUN . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2.3. TURN . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.2.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.2.5. Solution Profiles . . . . . . . . . . . . . . . . . . 11
4. NAT Traversal Scenarios . . . . . . . . . . . . . . . . . . . 12
4.1. Basic NAT SIP Signaling Traversal . . . . . . . . . . . . 12
4.1.1. Registration (Registrar/Proxy Co-Located) . . . . . . 12
4.1.2. Registration(Registrar/Proxy not Co-Located) . . . . . 16
4.1.3. Initiating a Session . . . . . . . . . . . . . . . . . 19
4.1.4. Receiving an Invitation to a Session . . . . . . . . . 21
4.2. Basic NAT Media Traversal . . . . . . . . . . . . . . . . 26
4.2.1. Endpoint Independent NAT . . . . . . . . . . . . . . . 27
4.2.2. Address and Port Dependant NAT . . . . . . . . . . . . 47
5. IPv4-IPv6 Transition . . . . . . . . . . . . . . . . . . . . . 55
5.1. IPv4-IPv6 Transition for SIP Signaling . . . . . . . . . . 55
5.2. IPv4-IPv6 Transition for Media . . . . . . . . . . . . . . 56
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 58
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 58
7.1. Normative References . . . . . . . . . . . . . . . . . . . 58
7.2. Informative References . . . . . . . . . . . . . . . . . . 61
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61
Intellectual Property and Copyright Statements . . . . . . . . . . 62
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1. Introduction
NAT (Network Address Translators) traversal has long been identified
as a complex problem when considered in the context of the Session
Initiation Protocol (SIP)[RFC3261] and it's associated media such as
Real Time Protocol (RTP)[RFC3550]. The problem is exacerbated 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 behaviors can be found in 'NAT Behavioral Requirements for
Unicast UDP' [RFC4787].
The IETF has been active on many specifications for the traversal of
NAT, including STUN[I-D.ietf-behave-rfc3489bis],
ICE[I-D.ietf-mmusic-ice], symmetric response[RFC3581], symmetric
RTP[RFC4961], TURN[I-D.ietf-behave-turn], SIP
Outbound[I-D.ietf-sip-outbound], SDP attribute for RTCP[RFC3605], 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 provides 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 is 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 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.
Normal SIP response routing over UDP causes the response to be
delivered to the source IP address specified in the topmost Via
header, or the "received" parameter of the topmost Via header. The
port is extracted from the SIP 'Via' header to complete the IP
address/port combination for returning the SIP response. While the
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destination for the response is correct, the port contained in the
SIP 'Via' 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 10923 --------
| |-------------------->--->-----------------------| |
| | | | |
| Client | |port 5060 SIP Response | Proxy |
| | x<------------------------| |
| | | | |
-------- | --------
|
|
|
Figure 1: Failed Response
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, for example, 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: Failed Request
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 impact of connection reuse of connection
orientated transports (TCP, TLS, etc) is discussed in more detail in
the connection reuse specification[I-D.ietf-sip-connect-reuse]. The
approach 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 may vary from a
number seconds to hours. SIP signaling can not be relied upon to
keep alive connections for the following two 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.
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RTP [RFC3550] is one of the most common media transport types used in
SIP signaling. Negotiation of RTP occurs with a SIP session
establishment using the Session Description Protocol(SDP) [RFC2327]
and a SIP offer/answer exchange[RFC3264]. 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: Failed Media
The connection addresses of the clients behind the NATs will
nominally contain a private IPv4 or IPv6 address that is not routable
across the public Internet. Exacerbating matters even more would be
the tendency of Client A to send media to a destination address it
received in the signaling confirmation message -- an address that may
actually correspond to a host within the private network who is
suddenly faced with incoming RTP packets (likewise, Client B may send
media to a host within its private network who did not solicit these
packets.) And finally, to complicate the problem even further, a
number of different NAT topologies with different default behaviors
increases the difficulty of arriving at a single approach.
3. Solution Technology Outline Description
As mentioned previously, the traversal of SIP through existing NATs
can be divided into two discrete problem areas: getting the core
signaling across the NAT, and enabling media as specified by SDP in a
SIP offer/answer exchange to flow between endpoints.
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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 signaling 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 [RFC3261]. Once the SIP response is constructed, the
destination is still derived using the mechanisms described in RFC
3261 [RFC3261]. 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 request---------------------------------->| |
| Client |<---------------------------------send response| Client |
| A | | | B |
| | | | |
-------- | --------
|
|
|
Figure 4: Symmetric Response
The outgoing request from Client A opens a pin hole in the NAT.
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Client B would normally respond to the port available in the SIP Via
header, as illustrated in Figure 1. Client B honors the 'rport'
parameter in the SIP Via header and routes the response to port from
which it was sent. The exact functionality for this method of
response traversal is called 'Symmetric Response' and the details are
documented in RFC 3581 [RFC3581]. 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. Re-use of Connections
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'[I-D.ietf-sip-outbound]. The
document provides techniques that use a unique User Agent instance
identifier (instance-id) in association with a flow identifier
(reg-id). The combination of the two identifiers provides a key to a
particular connections (both UDP and TCP) that are stored in
association with registration bindings. On receiving an incoming
request to a SIP Address-Of-Record (AOR), a proxy/registrar routes to
the associated flow created by the registration and thus a route
through a NAT. It also provides a keepalive mechanism for clients to
keep NAT bindings alive. This is achieved by multiplexing a relative
ping/pong mechanism over the SIP signaling connection (STUN for UDP
and CRLF/operating system keepalive for reliable transports like
TCP). Usage of this specification is RECOMMENDED. This mechanism is
not transport specific and should be used for any transport protocol.
Even if the SIP Outbound draft is not used, clients generating SIP
requests 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. Deployments should also consider the mechanism described
in the Connection Reuse[I-D.ietf-sip-connect-reuse] specification for
routing bi-directional messages securely between trusted SIP Proxy
servers.
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
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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/RTCP
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/RTCP can be
used. This involves an SIP endpoint both sending and receiving RTP/
RTCP traffic from the same IP Address/Port combination. This
technique is known as 'Symmetric RTP/RTCP' and is documented in RFC
4961 [RFC4961]. 'Symmetric RTP/RTCP' SHOULD only solely 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.
3.2.1.1. RTCP
Normal practice when selecting a port for defining Real Time Control
Protocol(RTCP) [RFC3550] 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 [RFC3605] defines an SDP attribute that is
included to explicitly specify transport connection information for
RTCP so a separate, explicit NAT binding can be set up for the
purpose. The address details can be obtained using any appropriate
method including those detailed in this section (e.g. STUN, TURN,
ICE).
An alternative mechanism defined in [I-D.ietf-avt-rtp-and-rtcp-mux]
specifies 'muxing' both RTP and RTCP on the same IP/PORT combination.
Using this technique eliminates the problem and is RECOMENDED.
3.2.2. STUN
Simple Traversal of Underneath Network Address Translators(NAT) or
STUN is defined in RFC 3489bis [I-D.ietf-behave-rfc3489bis]. STUN is
a lightweight tool kit and protocol that 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. Using terminology defined in the draft 'NAT Behavioral
Requirements for Unicast UDP' [RFC4787], STUN does work with
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'Endpoint Independent Mapping' but does not work with either 'Address
Dependent Mapping' or 'Address and Port Dependent Mapping' type NATs.
Using STUN with either of the previous two NAT mappings to probe for
the external IP address/port representation 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.
As mentioned in Section 3.1.2, STUN is also used as a client-to-
server keep-alive mechanism to refresh NAT bindings.
3.2.3. TURN
As described in the Section 3.2.2, the STUN protocol does not work
for UDP traversal through certain identified NAT mappings.
'Obtaining Relay Addresses from Simple Traversal of UDP Through NAT
(known as TURN)' is a usage of the STUN protocol for deriving (from a
STUN server) an address that will be used to relay packet towards a
client. TURN provides an external address (globally routable) at a
STUN server that will act as a media relay which guarantees traffic
will reach the associated internal address. The full details of the
TURN specification are defined in [I-D.ietf-behave-turn]. 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[RFC3424] compliant protocol to
provide a unified solution. This is achieved by obtaining as many
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,
priority, transport. 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
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contained in [I-D.ietf-mmusic-ice].
An extension to ICE is also available for TCP based media
interactions and is documented in 'TCP Candidates with Interactive
Connectivity Establishment (ICE)'[I-D.ietf-mmusic-ice-tcp].
3.2.5. 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.
3.2.5.1. Primary Profile
A client falling into the 'Primary' profile supports ICE in
conjunction with STUN, TURN usage and RFC 3605 [RFC3605] and/or
[I-D.ietf-avt-rtp-and-rtcp-mux] 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.
It is RECOMMENDED that symmetric RTP and symmetric RTCP always be
used for bidirectional RTP media streams.
3.2.5.2. Consumer Profile
A client falling into the 'Consumer' profile supports STUN and RFC
3605 [RFC3605] and/or [I-D.ietf-avt-rtp-and-rtcp-mux] for RTCP. It
uses STUN to allocate bindings, and can also detect when it is behind
an Address Dependant or Address and Port Dependant 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.
It is RECOMMENDED that symmetric RTP and symmetric RTCP always be
used for bidirectional RTP media streams.
3.2.5.3. Minimal Profile
A client falling into the 'Minimal' profile will send/receive RTP/
RTCP from the same IP/port combination. This client requires
proprietary network based solutions to function in any NAT traversal
scenario.
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In summary, the following table provides a representation of the
profiles:
| Primary | Consumer | Minimal |
| Profile | Profile | Profile |
---------+----------+----------+---------+
ICE | Yes | No | No |
---------+----------+----------+---------+
STUN | Yes | Yes | No |
---------+----------+----------+---------+
TURN Use | Yes | No | No |
---------+----------+----------+---------+
RFC3605 | Yes | Yes | No |
---------+----------+----------+---------+
Symm. RTP| Yes | Yes | Yes |
(RFC4961)| | | |
---------+----------+----------+---------+
Figure 5: Profiles
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.
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 Outbound Connection Tuple |
|*************************************|
| | |
| |(4) 200 OK |
| |<-----------------|
| | |
|(4) 200 OK | |
|<-----------------| |
| | |
Figure 6
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 6 is
a standard REGISTER message:
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REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK
Max-Forwards: 70
Supported: path,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>;reg-id=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:
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 [RFC3581] MUST be obeyed. Make note
of both the 'connectionID' and 'sip.instance' contact header
parameters. They are used to establish an Outbound connection tuple
as defined in [I-D.ietf-sip-outbound]. 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.
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 Outbound Connection 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:
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REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
Max-Forwards: 70
Supported: path,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>;reg-id=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 SIP Outbound
specification [I-D.ietf-sip-outbound]. This creates an association
tuple as described in the previous example for future inbound
requests directed at the newly created registration binding with the
only difference that the association is with a TCP connection, not a
UDP pin hole binding.
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.
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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 Outbound Connection Tuple |
|********************************************************|
| | | |
| |(8)200 OK | |
| |<-----------------| |
| | | |
|(8)200 OK | | |
|<-----------------| | |
| | | |
Figure 7.
This scenario builds on the previous example contained in
Section 4.1.1.2. The primary difference being that the REGISTER
request is routed onwards from a Proxy Server to a separated
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Registrar. The important message to note is (5) in Figure 7. The
Edge proxy, on receiving a REGISTER request that contains a
'sip.instance' media feature tag, forms a unique flow identifier
token as discussed in [I-D.ietf-sip-outbound]. At this point, the
proxy server routes the SIP REGISTER message to the Registrar. The
proxy will create the connection tuple as described in SIP Outbound
at the same moment as the co-located example, but for subsequent
messages to arrive at the Proxy, the element needs to request to
remain in the SIP signaling path. To achieve this the proxy inserts
to REGISTER message (5) a SIP PATH extension header, as defined in
RFC 3327 [RFC3327]. The previously created flow association token is
inserted in a position within the Path header where it can easily be
retrieved at a later point when receiving messages to be routed to
the registration binding (in this case the user part of the SIP URI).
The REGISTER message of (5), when proxied in (6) looks 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: path,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:3HS28o8HAKJSH&&U@proxy.example.com;lr,ob>
Contact: <sip:client@client.example.com;transport=tcp>;
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E121>";reg-id=1
Content-Length: 0
This REGISTER request 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 AOR binding. This guarantees that
all requests for the new registration will be forwarded to the Edge
Proxy. In our example, the user part of the SIP 'Path' header URI
that was inserted by the Edge Proxy contains the unique token
identifying the flow to the client. On receiving subsequent
requests, the edge proxy will examine the user part of the pre-loaded
SIP 'route' header and extract the unique flow token for use in its
connection tuple comparison, as defined in the SIP Outbound
specification [I-D.ietf-sip-outbound]. 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.
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4.1.3. Initiating a Session
This section covers basic SIP signaling when initiating a call from
behind a NAT.
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 | |
| |<-----------------| |
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| | | |
|(8)200 OK | | |
|<-----------------| | |
| | | |
|(9)ACK | | |
|----------------->| | |
| | | |
| |(9)ACK | |
| |----------------->| |
| | | |
| | |(10) ACK |
| | |---------------->|
| | | |
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:
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:clientA@example.com;
gr=urn:uuid:ijed7ush-4jan-53120-aee5-e0aecwee6wef;grid=45a>
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,
responses 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 [RFC3581].
2. Secondly, the contact inserted contains the
GRUU[I-D.ietf-sip-gruu] previously obtained from the SIP 200 OK
response to the registration. Use of the GRUU ensures that any
SIP requests within the dialog sent in the opposite direction
will be able to traverse the NAT. This occurs using the
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mechanisms defined in the SIP Outbound
specification[I-D.ietf-sip-outbound]. A request arriving at the
entity which resolves to the GRUU (registrar/proxy) is then able
to determine a previously registered connection that will allow
the request to traverse the NAT and reach the intended endpoint.
4.1.3.2. Reliable Transport
When using a reliable transport such as TCP the call flow and
procedures for traversing a NAT are almost identical to those
described in Section 4.1.3.1. The primary difference when using
reliable transport protocols is that Symmetric response[RFC3581] are
not required for SIP responses to traverse a NAT. RFC 3261[RFC3261]
defines procedures for SIP response messages to be sent back on the
same connection on which the request arrived.
4.1.4. Receiving an Invitation to a Session
This section details scenarios where a client behind a NAT receives
an inbound request through a NAT. These scenarios build on the
previous registration scenario from Section 4.1.1 and Section 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.
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 request matches the registration binding previously
installed 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 flows.
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 same instance identifier associated with the flow.
The request is then dispatched on the appropriate flow. This is
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message (2) from Figure 9 and is as follows:
INVITE 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 flow associated with the registration
binding (look-up procedures in RFC 3263 [RFC3263] are overridden).
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
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 TCP and follows on from the
registration installed in the example from Section 4.1.2.
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Client NAT Proxy Registrar SIP Entity
| | | | |
|***********************************************************|
| Registration Binding Installed in |
| section 4.1.2 |
|***********************************************************|
| | | | |
| | | |(1)INVITE |
| | | |<-------------|
| | | | |
| | |(2)INVITE | |
| | |<-------------| |
| | | | |
| |(3)INVITE | | |
| |<-------------| | |
| | | | |
|(3)INVITE | | | |
|<-------------| | | |
| | | | |
| | | | |
Figure 10.
An INVITE request arrives at the Registrar with a destination
pointing to the AOR of that inserted in Section 4.1.2. The message
is illustrated by (1) in Figure 10 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 request matches the registration binding previously
installed at the Registrar and the INVITE request-URI is re-written
to the selected onward address. The Registrar also identifies that a
SIP PATH header was associated with the registration and pushes it
into the INVITE request in the form of a pre-loaded SIP Route header.
It then forwards the request on to the proxy identified in the SIP
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Route header as shown in (2) from Figure 10:
INVITE sip:client@client.example.com SIP/2.0
Via: SIP/2.0/TCP registrar.example.com;branch=z9hG4bK74fmljnc
Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
Route: <sip:3HS28o8HAKJSH&&U@proxy.example.com;lr,ob>
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 request then arrives at the outbound proxy for the client. The
proxy examines the request URI of the INVITE in conjunction with the
flow token that it previously inserted into the user part of the PATH
header SIP URI (which now appears in the user part of the Route
header in the incoming INVITE). The proxy locates the appropriate
flow and sends the message to the client, as shown in (3) from Figure
10:
INVITE sip:client@client.example.com SIP/2.0
Via: SIP/2.0/TCP proxy.example.com;branch=z9hG4nsi30dncmnl
Via: SIP/2.0/TCP registrar.example.com;branch=z9hG4bK74fmljnc
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
at the originator. The major difference being that this request will
not follow the address specified in the Request-URI when it reaches
the outbound proxy, as standard SIP rules would enforce but will be
sent on the flow associated with the registration binding as
indicated in the Route header(look-up procedures in RFC 3263
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[RFC3263] are overridden). This then allows the original connection/
mapping from the initial registration to the outbound proxy to be re-
used.
4.2. Basic NAT Media Traversal
This section provides example scenarios to demonstrate basic media
traversal using the techniques outlined earlier in this document.
In the flow diagrams STUN messages have been annotated for simplicity
as follows:
o The "Src" attribute represents the source transport address of the
message.
o The "Dest" attribute represents the destination transport address
of the message.
o The "Map" attribute represents the server reflexive (XOR-MAPPED-
ADDRESS STUN attribute) transport address.
o The "Rel" attribute represents the relayed (RELAY-ADDRESS STUN
attribute) transport address.
The meaning of each STUN attribute is extensively explained in the
core STUN[I-D.ietf-behave-rfc3489bis] and TURN
usage[I-D.ietf-behave-turn] drafts.
A number of ICE SDP attributes have also been included in some of the
examples. Detailed information on individual attributes can be
obtained from the core ICE specification[I-D.ietf-mmusic-ice].
The examples also contain a mechanism for representing transport
addresses. It would be confusing to include representations of
network addresses in the call flows and make them hard to follow.
For this reason network addresses will be represented using the
following annotation. The first component will contain the
representation of the client responsible for the address. For
example in the majority of the examples "L" (left client), "R" (right
client), NAT-PUB" (NAT public), PRIV (Private), and "STUN-PUB" (STUN
Public) are used. To allow for multiple addresses from the same
network element, each representation can also be followed by a
number. These can also be used in combination. For example "L-NAT-
PUB-1" would represent a public network address on the left hand side
NAT while "R-NAT-PUB-1" would represent a public network address in
the right hand side of the NAT. "L-PRIV-1" would represent a private
network address on the left hand side of the NAT while "R-PRIV-1"
represents a private address on the right hand side of the NAT.
It should also be noted that during the examples it might be
appropriate to signify an explicit part of a transport address. This
is achieved by adding either the '.address' or '.port' tag on the end
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of the representation. For example, 'L-PRIV-1.address' and 'L-PRIV-
1.port'.
4.2.1. Endpoint Independent NAT
This section demonstrates an example of a client both initiating and
receiving calls behind an 'Endpoint independent' NAT. An example is
included for both STUN and ICE with ICE being the RECOMMENDED
mechanism for media traversal.
4.2.1.1. STUN Solution
It is possible to traverse media through an 'Endpoint Independent NAT
using STUN. The remainder of this section provides simplified
examples of the 'Binding Discovery' STUN usage as defined in
[I-D.ietf-behave-rfc3489bis]. The STUN messages have been simplified
and do not include 'Shared Secret' requests used to obtain the
temporary username and password.
4.2.1.1.1. Initiating Session
The following example demonstrates media traversal through a NAT with
'Address Independent' properties using the STUN 'Binding Discovery'
usage. It is assumed in this example that the STUN client and SIP
Client are co-located on the same physical machine. Note that some
SIP signaling messages have been left out for simplicity.
Client NAT STUN [..]
Server
| | | |
|(1) BIND Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(2) BIND Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(3) BIND Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
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|(4) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|XOR=NAT-PUB-1 | | |
| | | |
|(5) BIND Req | | |
|Src=L-PRIV-2 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(6) BIND Req | |
| |Src=NAT-PUB-2 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(7) BIND Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-2 | |
| |Map=NAT-PUB-2 | |
| | | |
|(8) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-2 | | |
|Map=NAT-PUB-2 | | |
| | | |
|(9)SIP INVITE | | |
|----------------->| | |
| | | |
| |(10)SIP INVITE | |
| |------------------------------------>|
| | | |
| | |(11)SIP 200 OK |
| |<------------------------------------|
| | | |
|(12)SIP 200 OK | | |
|<-----------------| | |
| | | |
|========================================================|
|>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>>|
|========================================================|
| |
|========================================================|
|<<<<<<<<<<<<Incoming Media sent to NAT-PUB-1<<<<<<<<<<<<|
|========================================================|
| |
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|========================================================|
|>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>|
|========================================================|
| |
|========================================================|
|<<<<<<<<<<<<Incoming RTCP sent to NAT-PUB-2<<<<<<<<<<<<<|
|========================================================|
| | | |
|(13)SIP ACK | | |
|----------------->| | |
| | | |
| |(14) SIP ACK | |
| |------------------------------------>|
| | | |
Figure 7: Endpoint Independent NAT - Initiating
o On deciding to initiate a SIP voice session the client starts a
local STUN client on the interface and port that is to be used for
media (send/receive). The STUN client generates a standard
'Binding Discovery' request as indicated in (1) from Figure 7
which also highlights the source address and port for which the
client device wishes to obtain a mapping. The 'Binding Discovery'
request is sent through the NAT towards the public internet and
STUN server.
o Message (2) traverses the NAT and breaks out onto the public
internet towards the public STUN server. Note that the source
address of the 'Binding Discovery' request now represents the
public address and port from the public side of the NAT.
o The STUN server receives the request and processes it
appropriately. This results in a successful 'Binding Discovery'
response being generated and returned (3). The message contains
details of the XOR mapped public address (contained in the STUN
XOR-MAPPED-ADDRESS attribute) which is to be used by the
originating client to receive media (see 'Map=NAT-PUB-1' from
(3)).
o The 'Binding Discovery' response traverses back through the NAT
using the path created by the 'Binding Discovery' request and
presents the new XOR mapped address to the client (4). At this
point the process is repeated to obtain a second XOR-mapped
address (as shown in (5)-(8)) for an alternative local address
(Address has changed from "L-PRIV-1" to "L-PRIV-2") for an RTCP
port.
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
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populate the SDP of the SIP INVITE as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 $L-PRIV-1.address
c=IN IP4 $NAT-PUB-1.address
t=0 0
m=audio $NAT-PUB-1.port RTP/AVP 0
a=rtcp:$NAT-PUB-2.port
o Note that the XOR-mapped address obtained from the 'Binding
Discovery' transactions are inserted as the connection address for
the SDP (c=NAT-PUB-1.address). The Primary port for RTP is also
inserted in the SDP (m=audio NAT-PUB-1.port RTP/AVP 0). Finally,
the port gained from the additional 'Binding Discovery' is placed
in the RTCP attribute (as discussed in Section 3.2.1.1) for
traversal of RTCP (a=rtcp:NAT-PUB-2.port).
o The SIP signaling then traverses the NAT and sets up the SIP
session (9-12). 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 and RTCP 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 an 'Endpoint Independent' NAT using the STUN
'Binding Discovery' usage is very similar to the example outlined in
Section 4.2.1.1.1. Figure 8 illustrates the associated flow of
messages.
Client NAT STUN [..]
Server
| | | (1)SIP INVITE |
| |<-----------------|------------------|
| | | |
|(2) SIP INVITE | | |
|<-----------------| | |
| | | |
|(3) BIND Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(4) BIND Req | |
| |Src=NAT-PUB-1 | |
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| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(5) BIND Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
|(6) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|Map=NAT-PUB-1 | | |
| | | |
|(7) BIND Req | | |
|Src=L-PRIV-2 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(8) BIND Req | |
| |Src=NAT-PUB-2 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(9) BIND Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-2 | |
| |Map=NAT-PUB-2 | |
| | | |
|(10) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-2 | | |
|Map=NAT-PUB-2 | | |
| | | |
|(11)SIP 200 OK | | |
|----------------->| | |
| |(12)SIP 200 OK | |
| |------------------------------------>|
| | | |
|========================================================|
|>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>>|
|========================================================|
| | | |
|========================================================|
|<<<<<<<<<<<<<Incoming Media sent to L-PRIV-1<<<<<<<<<<<<|
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|========================================================|
| | | |
|========================================================|
|>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>|
|========================================================|
| | | |
|========================================================|
|<<<<<<<<<<<<<Incoming RTCP sent to L-PRIV-2<<<<<<<<<<<<<|
|========================================================|
| | | |
| | |(13)SIP ACK |
| |<------------------------------------|
| | | |
|(14)SIP ACK | | |
|<-----------------| | |
| | | |
Figure 8: Endpoint Independent NAT - Receiving
o On receiving an invitation to a SIP voice session (SIP INVITE
request) the User Agent starts a local STUN client on the
appropriate port on which it is to receive media. The STUN client
generates a standard 'Binding Discovery' request as indicated in
(3) from Figure 8 which also highlights the source address and
port for which the client device wishes to obtain a mapping. The
'Binding Discovery' request is sent through the NAT towards the
public internet and STUN Server.
o 'Binding Discovery' 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 The STUN server receives the request and processes it
appropriately. This results in a successful 'Binding Discovery'
response being generated and returned (5). The message contains
details of the mapped public address (contained in the STUN XOR-
MAPPED-ADDRESS attribute) which is to be used by the originating
client to receive media (see 'Map=NAT-PUB-1' from (5)).
o The 'Binding Discovery' response traverses back through the NAT
using the path created by the outgoing 'Binding Discovery' request
and presents the new XOR-mapped address to the client (6). At
this point the process is repeated to obtain a second XOR-mapped
address (as shown in (7)-(10)) for an alternative local address
(local port has now changed and is represented by L-PRIV-2 in (7))
for an RTCP port.
o The client now constructs a SIP 200 OK message (11) in response to
the original SIP INVITE requests. Note that traversal of SIP is
not covered in this example and is discussed in earlier sections
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of the document. SIP Provisional responses are also left out for
simplicity. The 200 OK response will use the addresses it has
obtained in the previous STUN transactions to populate the SDP of
the SIP 200 OK as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 $L-PRIV-1.address
c=IN IP4 $NAT-PUB-1.address
t=0 0
m=audio $NAT-PUB-1.port RTP/AVP 0
a=rtcp:$NAT-PUB-2.port
o Note that the XOR-mapped address obtained from the initial
'Binding Discovery' transaction is inserted as the connection
address for the SDP (c=NAT-PUB-1.address). The Primary port for
RTP is also inserted in the SDP (m=audio NAT-PUB-1.port RTP/AVP
0). Finally, the port gained from the additional 'Binding
Discovery' is placed in the RTCP attribute (as discussed in
Section 3.2.1.1) for traversal of RTCP (a=rtcp:NAT-PUB-2.port).
o The SIP signaling 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, regardless of the NAT type. The
following examples illustrate the traversal of an 'Endpoint
Independent' NAT when initiating the session. The example only
covers ICE in association with the 'Binding Discovery' and TURN
usage.
4.2.1.2.1. Initiating Session
The following example demonstrates an initiating traversal through an
'Endpoint independent' NAT using ICE.
L NAT STUN NAT R
Server
| | | | |
|(1) Alloc Req | | | |
|Src=L-PRIV-1 | | | |
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|Dest=STUN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(2) Alloc Req | | |
| |Src=L-NAT-PUB-1 | | |
| |Des=STUN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(3) Alloc Resp | | |
| |<---------------| | |
| |Src=STUN-PUB-1 | | |
| |Dest=L-NAT-PUB-1| | |
| |Map=L-NAT-PUB-1 | | |
| |Rel=STUN-PUB-2 | | |
| | | | |
|(4) Alloc Resp | | | |
|<---------------| | | |
|Src=STUN-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
|Rel=STUN-PUB-2 | | | |
| | | | |
|(5) STUN Req | | | |
|Src=L-PRIV-2 | | | |
|Dest=STUN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(6) Alloc Req | | |
| |Src=L-NAT-PUB-2 | | |
| |Dest=STUN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(7) Alloc Resp | | |
| |<---------------| | |
| |Src=STUN-PUB-1 | | |
| |Dest=NAT-PUB-2 | | |
| |Map=NAT-PUB-2 | | |
| |Rel=STUN-PUB-3 | | |
| | | | |
|(8) Alloc Resp | | | |
|<---------------| | | |
|Src=STUN-PUB-1 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
|Rel=STUN-PUB-3 | | | |
| | | | |
|(9) SIP INVITE | | | |
|------------------------------------------------->| |
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| | | | |
| | | |(10) SIP INVITE |
| | | |--------------->|
| | | | |
| | | |(11) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=STUN-PUB-1 |
| | | | |
| | |(12) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=STUN-PUB-1 | |
| | | | |
| | |(13) Alloc Res | |
| | |--------------->| |
| | |Src=STUN-PUB-1 | |
| | |Dest=R-NAT-PUB-1| |
| | |Map=R-NAT-PUB-1 | |
| | |Rel=STUN-PUB-4 | |
| | | | |
| | | |(14) Alloc Res |
| | | |--------------->|
| | | |Src=STUN-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | |Map=R-NAT-PUB-1 |
| | | |Rel=STUN-PUB-4 |
| | | | |
| | | |(15) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=STUN-PUB-1 |
| | | | |
| | |(16) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=STUN-PUB-1 | |
| | | | |
| | |(17) Alloc Res | |
| | |--------------->| |
| | |Src=STUN-PUB-1 | |
| | |Dest=R-NAT-PUB-2| |
| | |Map=R-NAT-PUB-2 | |
| | |Rel=STUN-PUB-5 | |
| | | | |
| | | |(18) Alloc Res |
| | | |--------------->|
| | | |Src=STUN-PUB-1 |
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| | | |Dest=R-PRIV-2 |
| | | |Map=R-NAT-PUB-2 |
| | | |Rel=STUN-PUB-5 |
| | | | |
| | | |(19) SIP 200 OK |
| |<-------------------------------------------------|
| | | | |
|(20) SIP 200 OK | | | |
|<---------------| | | |
| | | | |
|(21) SIP ACK | | | |
|------------------------------------------------->| |
| | | | |
| | | |(22) SIP ACK |
| | | |--------------->|
| | | | |
|(23) Bind Req | | | |
|------------------------>x | | |
|Src=L-PRIV-1 | | | |
|Dest=R-PRIV-1 | | | |
| | | | |
|(24) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-1 | | | |
|Dest=R-NAT-PUB-1| | | |
| | | | |
| |(25) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-1 | | |
| |Dest=R-NAT-PUB-1| | |
| | | | |
| | | |(26) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | | |
| | | |(27) Bind Res |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-NAT-PUB-1|
| | | |Map=L-NAT-PUB-1 |
| | | | |
| | |(28) Bind Res | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=L-NAT-PUB-1| |
| | |Map=L-NAT-PUB-1 | |
| | | | |
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|(29) Bind Res | | | |
|<---------------| | | |
|Src=R-NAT-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
| | | | |
|===================================================================|
|>>>>>>>>>>>>>>>>>>>>>Outgoing RTP sent from >>>>>>>>>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(30) Bind Req |
| | | x<-----------------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-PRIV-1 |
| | | | |
| | | |(31) Bind Req |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-NAT-PUB-1|
| | | | |
| | |(32) Bind Req | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=L-NAT-PUB-1| |
| | | | |
|(33) Bind Req | | | |
|<---------------| | | |
|Src=R-NAT-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
| | | | |
|(34) Bind Res | | | |
|--------------->| | | |
|Src=L-PRIV-1 | | | |
|Dest=R-NAT-PUB-1| | | |
|Map=R-NAT-PUB-1 | | | |
| | | | |
| |(35) Bind Res | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-1 | | |
| |Dest=R-NAT-PUB-1| | |
| |Map=R-NAT-PUB-1 | | |
| | | | |
| | | |(36) Bind Res |
| | | |--------------->|
| | | |Src=L-NAT-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | |Map=R-NAT-PUB-1 |
| | | | |
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|===================================================================|
|<<<<<<<<<<<<<<<<<<<<<Outgoing RTP sent from <<<<<<<<<<<<<<<<<<<<<<<|
|===================================================================|
|(37) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-1 | | | |
|Dest=R-NAT-PUB-1| | | |
|USE-CANDIDATE | | | |
| | | | |
| |(38) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-1 | | |
| |Dest=R-NAT-PUB-1| | |
| |USE-CANDIDATE | | |
| | | | |
| | | |(39) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | |USE-CANDIDATE |
| | | | |
| | | |(40) Bind Res |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-NAT-PUB-1|
| | | |Map=L-NAT-PUB-1 |
| | | | |
| | |(41) Bind Res | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=L-NAT-PUB-1| |
| | |Map=L-NAT-PUB-1 | |
| | | | |
|(42) Bind Res | | | |
|<---------------| | | |
|Src=R-NAT-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
| | | | |
|(43) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
| | | | |
| |(44) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-2 | | |
| |Dest=R-NAT-PUB-2| | |
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| | | | |
| | | |(45) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-2 |
| | | |Dest=R-PRIV-2 |
| | | | |
| | | |(46) Bind Res |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=L-NAT-PUB-2|
| | | |Map=L-NAT-PUB-2 |
| | | | |
| | |(47) Bind Res | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=L-NAT-PUB-2| |
| | |Map=L-NAT-PUB-2 | |
| | | | |
|(48) Bind Res | | | |
|<---------------| | | |
|Src=R-NAT-PUB-2 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
| | | | |
|===================================================================|
|>>>>>>>>>>>>>>>>>>>>>Outgoing RTCP sent from >>>>>>>>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(49) Bind Req |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=L-NAT-PUB-2|
| | | | |
| | |(50) Bind Req | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=L-NAT-PUB-2| |
| | | | |
|(51) Bind Req | | | |
|<---------------| | | |
|Src=R-NAT-PUB-2 | | | |
|Dest=L-PRIV-2 | | | |
| | | | |
|(52) Bind Res | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
|Map=R-NAT-PUB-2 | | | |
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| | | | |
| |(53) Bind Res | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-2 | | |
| |Dest=R-NAT-PUB-2| | |
| |Map=R-NAT-PUB-2 | | |
| | | | |
| | | |(54) Bind Res |
| | | |--------------->|
| | | |Src=L-NAT-PUB-2 |
| | | |Dest=R-PRIV-2 |
| | | |Map=R-NAT-PUB-2 |
| | | | |
|===================================================================|
|<<<<<<<<<<<<<<<<<<<<<Outgoing RTCP sent from <<<<<<<<<<<<<<<<<<<<<<|
|===================================================================|
|(55) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
|USE-CANDIDATE | | | |
| | | | |
| |(56) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-2 | | |
| |Dest=R-NAT-PUB-2| | |
| |USE-CANDIDATE | | |
| | | | |
| | | |(57) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-2 |
| | | |Dest=R-PRIV-2 |
| | | |USE-CANDIDATE |
| | | | |
| | | |(58) Bind Res |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=L-NAT-PUB-2|
| | | |Map=L-NAT-PUB-2 |
| | | | |
| | |(59) Bind Res | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=L-NAT-PUB-2| |
| | |Map=L-NAT-PUB-2 | |
| | | | |
|(60) Bind Res | | | |
|<---------------| | | |
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|Src=R-NAT-PUB-2 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
| | | | |
| | | | |
|(61) SIP INVITE | | | |
|------------------------------------------------->| |
| | | | |
| | | |(62) SIP INVITE |
| | | |--------------->|
| | | | |
| | | |(63) SIP 200 OK |
| |<-------------------------------------------------|
| | | | |
|(64) SIP 200 OK | | | |
|<---------------| | | |
| | | | |
|(65) SIP ACK | | | |
|------------------------------------------------->| |
| | | | |
| | | |(66) SIP ACK |
| | | |--------------->|
| | | | |
Figure 9: Endpoint Independent NAT with ICE
o On deciding to initiate a SIP voice session the SIP client 'L'
starts a local STUN client. The STUN client generates a standard
Allocate request as indicated in (1) from Figure 9 which also
highlights the source address and port combination for which the
client device wishes to obtain a mapping. The Allocate request is
sent through the NAT towards the public internet.
o The Allocate message (2) traverses the NAT and breaks out onto the
public internet towards the public STUN server. Note that the
source address of the Allocate request now represents the public
address and port from the public side of the NAT (L-NAT-PUB-1).
o The STUN server receives the Allocate request and processes
appropriately. This results in a successful Allocate response
being generated and returned (3). The message contains details of
the server reflexive address which is to be used by the
originating client to receive media (see 'Map=L-NAT-PUB-1') from
(3)). It also contains an appropriate relayed address that can be
used at the STUN server (see 'Rel=STUN-PUB-2').
o The Allocate response traverses back through the NAT using the
binding created by the initial Allocate request and presents the
new mapped address to the client (4). The process is repeated and
a second STUN derived set of address' are obtained, as illustrated
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in (5)-(8) in Figure 9. At this point the User Agent behind the
NAT has pairs of derived external server reflexive and relayed
representations. The client would be free to gather any number of
external representations using any UNSAF[RFC3424] compliant
protocol.
o The client now constructs a SIP INVITE message (9). 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[I-D.ietf-mmusic-ice], as shown
below in Figure 10 (*note - 'allOneLine' notation signifies line
continuation):
v=0
o=test 2890844526 2890842807 IN IP4 $L-PRIV-1
c=IN IP4 $L-PRIV-1.address
t=0 0
a=ice-pwd:$LPASS
a=ice-ufrag:$LUNAME
m=audio $L-PRIV-1.port RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:$L-PRIV-2.port
a=candidate:$L1 1 UDP 3102938476 $L-PRIV-1.address $L-PRIV-1.port typ local
a=candidate:$L1 2 UDP 3010948635 $L-PRIV-2.address $L-PRIV-2.port typ local
<allOneLine>
a=candidate:$L2 1 UDP 2203948363 $L-NAT-PUB-1.address $L-NAT-PUB-1.port typ srflx
raddr $L-PRIV-1.address rport $L-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L2 2 UDP 2172635342 $L-NAT-PUB-2.address $L-NAT-PUB-2.port typ srflx
raddr $L-PRIV-1.address rport $L-PRIV-2.port
</allOneLine>
<allOneLine>
a=candidate:$L3 1 UDP 1763546473 $STUN-PUB-2.address $STUN-PUB-2.port typ relay
raddr $L-PRIV-1.address rport $L-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L3 2 UDP 1109384746 $STUN-PUB-3.address $STUN-PUB-3.port typ relay
raddr $L-PRIV-1.address rport $L-PRIV-2.port
</allOneLine>
Figure 10: ICE SDP Offer
o The SDP has been constructed to include all the available
candidates that have been assembled. The first set of candidates
(as identified by Foundation $L1) contain two local addresses that
have the highest priority. They are also encoded into the
connection (c=) and media (m=) lines of the SDP. The second set
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of candidates, as identified by Foundation $L2, contains the two
server reflexive addresses obtained from the STUN server for both
RTP and RTCP traffic (identified by candidate-id $L2). This entry
has been given a priority lower than the pair $L1 by the client.
The third and final set of candidates represents the relayed
addresses (as identified by $L3) obtained from the STUN server.
This pair has the lowest priority and will be used as a last
resort if both $L1 or $L2 fail.
o The SIP signaling then traverses the NAT and sets up the SIP
session (9)-(10). 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 STUN connectivity checks.
o On receiving the SIP INVITE request (10) client 'R' also starts
local STUN servers on appropriate address/port combinations and
gathers potential candidate addresses to be encoded into the SDP
(as the originating client did). Steps (11-18) involve client 'R'
carrying out the same steps as client 'L'. This involves
obtaining local, server reflexive and relayed addresses. Client
'R' is now ready to generate an appropriate answer in the SIP 200
OK message (19). The example answer follows in Figure 10 (*note -
'allOneLine' notation signifies line continuation):
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v=0
o=test 3890844516 3890842803 IN IP4 $R-PRIV-1
c=IN IP4 $R-PRIV-1.address
t=0 0
a=ice-pwd:$RPASS
m=audio $R-PRIV-1.port RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:$R-PRIV-2.port
a=candidate:$L1 1 UDP 3303948573 $R-PRIV-1.address $R-PRIV-1.port typ local
a=candidate:$L1 2 UDP 3184756473 $R-PRIV-2.address $R-PRIV-2.port typ local
<allOneLine>
a=candidate:$L2 1 UDP 2984756463 $R-NAT-PUB-1.address $R-NAT-PUB-1.port typ srflx
raddr $R-PRIV-1.address rport $R-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L2 2 UDP 2605968473 $R-NAT-PUB-2.address $R-NAT-PUB-2.port typ srflx
raddr $R-PRIV-1.address rport $R-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L3 1 UDP 1453647488 $STUN-PUB-2.address $STUN-PUB-4.port typ relay
raddr $R-PRIV-1.address rport $R-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L3 2 UDP 1183948473 $STUN-PUB-3.address $STUN-PUB-5.port typ relay
raddr $R-PRIV-1.address rport $R-PRIV-1.port
</allOneLine>
Figure 11: ICE SDP Answer
o The two clients have now exchanged SDP using offer/answer and can
now continue with the ICE processing - User Agent 'L' assuming the
role controlling agent, as specified by ICE. The clients are now
required to form their Candidate check lists to determine which
will be used for the media streams. In this example User Agent
'L's 'Foundation 1' is paired with User Agent 'R's 'Foundation 1',
User Agent 'L's 'Foundation 2' is paired with User Agent 'R's
'Foundation 2', and finally User Agent 'L's 'Foundation 3' is
paired with User Agent 'R's 'Foundation 3'. User Agents 'L' and
'R' now have a complete candidate check list. Both clients now
use the algorithm provided in ICE to determine candidate pair
priorities and sort into a list of decreasing priorities. In this
example, both User Agent 'L' and 'R' will have lists that firstly
specifies the host address (Foundation $L1), then the server
reflexive address (Foundation $L2) and lastly the relayed address
(Foundation $L3). All candidate pairs have an associate state as
specified in ICE. At this stage, all of the candidate pairs for
User Agents 'L' and 'R' are initialized to the 'Frozen' state.
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The User Agents then scan the list and move the candidates to the
'Waiting' state. At this point both clients will periodically,
starting with the highest candidate pair priority, work their way
down the list issuing STUN checks from the local candidate to the
remote candidate (of the candidate pair). As a STUN Check is
attempted from each local candidate in the list, the candidate
pair state transitions to 'In-Progress'. As illustrated in (23),
client 'L' constructs a STUN connectivity check in an attempt to
validate the remote candidate address received in the SDP of the
200 OK (20) for the highest priority in the check list. As a
private address was specified in the active address in the SDP,
the STUN connectivity check fails to reach its destination causing
a STUN failure. Client 'L' transitions the state for this
candidate pair to 'Failed'. In the mean time, Client 'L' is
attempting a STUN connectivity check for the second candidate pair
in the returned SDP with the second highest priority (24). As can
be seen from messages (24) to (29), the STUN Bind request is
successful and returns a positive outcome for the connectivity
check. Client 'L' is now free to steam media to the candidate
pair. Client 'R' also carries out the same set of binding
requests. It firstly (in parallel) tries to contact the active
address contained in the SDP (30) which results in failure.
o In the mean time, a successful response to a STUN connectivity
check by User Agent 'R' (27) results in a tentative check in the
reverse direction - this is illustrated by messages (31) to (36).
Once this check has succeeded, User Agent 'R' can transition the
state of the appropriate candidate to 'Succeeded', and media can
be sent (RTP). The previously described check confirm on both
sides (User Agent 'L' and 'R') that connectivity can be achieved
using the appropriate candidate pair. User Agent 'L', as the
controlling client now sends another connectivity check for the
candidate pair, this time including the 'USE-CANDIDATE' attribute
as specified in ICE to signal the chosen candidate. This exchange
is illustrated in messages (37) to (42).
o As part of the process in this example, both 'L' and 'R' will now
complete the same connectivity checks for part 2 of the component
named for the favored 'Foundation' selected for use with RTCP.
The connectivity checks for part '2' of the candidate component
are shown in 'L'(43-48) and 'R'(49-54). Once this has succeeded,
User Agent 'L' as the controlling client sends another
connectivity check for the candidate pair. This time the 'USE-
CANDIDATE' attribute is again specified to signal the chosen
candidate for component '2'.
o The candidates have now been fully verified (and selected) and as
they are the highest priority, an updated offer (61-62) is now
sent from the offerer (client 'L') to the answerer (client 'R')
representing the new active candidates. The new offer would look
as follows:
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v=0
o=test 2890844526 2890842808 IN IP4 $L-PRIV-1
c=IN IP4 $L-NAT-PUB-1.address
t=0 0
a=ice-pwd:$LPASS
a=ice-ufrag:$LUNAME
m=audio $L-NAT-PUB-1.port RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:$L-NAT-PUB-2.port
<allOneLine>
a=candidate:$L2 1 UDP 2203948363 $L-NAT-PUB-1.address $L-NAT-PUB-1.port typ srflx
raddr $L-PRIV-1.address rport $L-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L2 2 UDP 2172635342 $L-NAT-PUB-2.address $L-NAT-PUB-2.port typ srflx
raddr $L-PRIV-1.address rport $L-PRIV-2.port
</allOneLine>
Figure 12: ICE SDP Updated Offer
o The resulting answer (63-64) for 'R' would look as follows:
v=0
o=test 3890844516 3890842804 IN IP4 $R-PRIV-1
c=IN IP4 $R-PRIV-1.address
t=0 0
a=ice-pwd:$RPASS
a=ice-ufrag:$RUNAME
m=audio $R-PRIV-1.port RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:$R-PRIV-2.port
<allOneLine>
a=candidate:$L2 1 UDP 2984756463 $R-NAT-PUB-1.address $R-NAT-PUB-1.port typ srflx
raddr $R-PRIV-1.address rport $R-PRIV-1.port
</allOneLine>
<allOneLine>
a=candidate:$L2 2 UDP 2605968473 $R-NAT-PUB-2.address $R-NAT-PUB-2.port typ srflx
raddr $R-PRIV-1.address rport $R-PRIV-2.port
</allOneLine>
Figure 13: ICE SDP Updated Answer
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4.2.2. Address and Port Dependant 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 case. The use of basic
STUN on its own will not guarantee traversal through every NAT type,
hence the recommendation that ICE is the preferred option.
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Client PORT/ADDRESS Dependant STUN [..]
NAT Server
| | | |
|(1) BIND Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(2) BIND Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(3) BIND Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
|(4) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|Map=NAT-PUB-1 | | |
| | | |
|(5)SIP INVITE | | |
|------------------------------------------------------->|
| | | |
| | |(6)SIP 200 OK |
| |<------------------------------------|
| | | |
|(7)SIP 200 OK | | |
|<-----------------| | |
| | | |
|========================================================|
|>>>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>|
|========================================================|
| | | |
| x=====================================|
| xIncoming Media sent to L-PRIV-1<<<<<<|
| x=====================================|
| | | |
|(8)SIP ACK | | |
|----------------->| | |
| |(9) SIP ACK | |
| |------------------------------------>|
| | | |
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Figure 14: Port/Address Dependant NAT with STUN - Failure
The example in Figure 14 is conveyed in the context of a client
behind the 'Port/Address Dependant' NAT initiating a call. It should
be noted that the same problem applies when a client receives a SIP
invitation and is behind a Port/Address Dependant NAT.
o In Figure 14 the client behind the NAT obtains a server reflexive
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).
o The external mapped address (server reflexive) 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 reflexive address supplied in the outgoing
INVITE request to traverse media back through the 'Port/Address
Dependent' NAT.
o A 'Port/Address Dependant' NAT has differing rules from the
'Endpoint Independent' type of NAT (as defined in RFC4787
[RFC4787]). For any internal IP address and port combination,
data sent to a different external destination does not provide the
same public mapping at the NAT. In Figure 14 the STUN query
produced a valid external mapping for receiving media. 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 do not originate from the
STUN server IP address and optionally port, will be dropped at the
NAT. Figure 14 shows the media being dropped at the NAT after (7)
and before (8). This then leads to one way audio.
4.2.2.2. TURN Usage Solution
As identified in Section 4.2.2.1, STUN provides a useful tool kit for
the traversal of the majority of NATs but fails with Port/Address
Dependant NAT. This led to the development of the TURN usage
solution [I-D.ietf-behave-turn] which uses the STUN toolkit to create
a profile. It allows a client to request a relayed address at the
STUN server rather than a reflexive representation. This then
introduces a media relay in the path for NAT traversal (as described
in Section 3.2.3). The following example explains how the TURN usage
solves the previous failure when using STUN to traverse a 'Port/
Address Dependant' type NAT.
L Port/Address Dependant STUN [..]
NAT Server
| | | |
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|(1) Alloc Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB-1 | | |
|----------------->| | |
| | | |
| |(2) Alloc Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB-1 | |
| |----------------->| |
| | | |
| |(3) Alloc Resp | |
| |<-----------------| |
| |Src=STUN-PUB-1 | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| |Rel=STUN-PUB-2 | |
| | | |
|(4) Alloc Resp | | |
|<-----------------| | |
|Src=STUN-PUB-1 | | |
|Dest=L-PRIV-1 | | |
|Map=NAT-PUB-1 | | |
|Rel=STUN-PUB-2 | | |
| | | |
|(5) Alloc Req | | |
|Src=L-PRIV-2 | | |
|Dest=STUN-PUB-1 | | |
|----------------->| | |
| | | |
| |(6) Alloc Req | |
| |Src=NAT-PUB-2 | |
| |Dest=STUN-PUB-1 | |
| |----------------->| |
| | | |
| |(7) Alloc Resp | |
| |<-----------------| |
| |Src=STUN-PUB-1 | |
| |Dest=NAT-PUB-2 | |
| |Map=NAT-PUB-2 | |
| |Rel=STUN-PUB-3 | |
| | | |
|(8) Alloc Resp | | |
|<-----------------| | |
|Src=STUN-PUB-1 | | |
|Dest=L-PRIV-2 | | |
|Map=NAT-PUB-2 | | |
|Rel=STUN-PUB-3 | | |
| | | |
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|(9)SIP INVITE | | |
|----------------->| | |
| | | |
| |(10)SIP INVITE | |
| |------------------------------------>|
| | | |
| | |(11)SIP 200 OK |
| |<------------------------------------|
| | | |
|(12)SIP 200 OK | | |
|<-----------------| | |
| | | |
|========================================================|
|>>>>>>>>>>>>>Outgoing Media sent from L-PRIV-1>>>>>>>>>>|
|========================================================|
| | | |
| | |==================|
| | |<<<Media Sent to<<|
| | |<<<<STUN-PUB-2<<<<|
| | |==================|
| | | |
|=====================================| |
|<Incoming Media Relayed to L-PRIV-1<<| |
|=====================================| |
| | | |
| | |==================|
| | |<<<RTCP Sent to<<>|
| | |<<<<STUN-PUB-3<<<<|
| | |==================|
| | | |
|=====================================| |
|<<Incoming RTCP Relayed to L-PRIV-2<<| |
|=====================================| |
| | | |
|(13)SIP ACK | | |
|----------------->| | |
| | | |
| |(14) SIP ACK | |
| |------------------------------------>|
| | | |
Figure 15: Port/Address Dependant NAT with TURN Usage - Success
o In this example, client 'L' issues a TURN allocate request(1) to
obtained a relay address at the STUN server. The request
traverses through the 'Port/Address Dependant' NAT and reaches the
STUN server (2). The STUN server generates an Allocate response
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(3) that contains both a server reflexive address (Map=NAT-PUB-1)
of the client and also a relayed address (Rel=STUN-PUB-2). The
relayed address maps to an address mapping on the STUN server
which is bound to the public pin hole that has been opened on the
NAT by the Allocate request. This results in any traffic sent to
the STUN server relayed address (Rel=STUN-PUB-2) being forwarded
to the external representation of the pin hole created by the
Allocate request(NAT-PUB-1).
o The TURN derived address (STUN-PUB-2) arrives back at the
originating client(4) in an Allocate response. This address can
then be used in the SDP for the outgoing SIP INVITE request as
shown in the following example (note that the example also
includes client 'L' obtaining a second relay address for use in
the RTCP attribute (5-8)):
o
v=0
o=test 2890844342 2890842164 IN IP4 $L-PRIV-1
c=IN IP4 $STUN-PUB-2.address
t=0 0
m=audio $STUN-PUB-2.port RTP/AVP 0
a=rtcp:$STUN-PUB-3.port
o On receiving the INVITE request, the UAS is able to stream media
and RTCP to the relay address (STUN-PUB-2 and STUN-PUB-3) at the
STUN server. As shown in Figure 15 (between messages (12) and
(13), the media from the UAS is directed to the relayed address at
the STUN server. The STUN server then forwards the traffic to the
open pin holes in the Port/Address Dependant NAT (NAT-PUB-1 and
NAT-PUB-2). The media traffic is then able to traverse the 'Port/
Address Dependant' NAT and arrives back at client 'L'.
o The TURN usage of STUN on its own will work for 'Port/Address
Dependent' 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 NAT
traversal and comes at a high processing cost.
4.2.2.3. ICE Solution
The previous two examples have highlighted the problem with using
core STUN usage for all forms of NAT traversal and a solution using
TURN usage for the Address/Port Dependant 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 core STUN, TURN usage and any other UNSAF[RFC3424]
compliant protocol to provide a list of prioritized addresses that
can be used for media traffic. Detailed examples of ICE can be found
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in Section 4.2.1.2.1. These examples are associated with an
'Endpoint Independent' type NAT but can be applied to any NAT type
variation, including 'Address/Port Dependant' type NAT. The ICE
procedures carried out are the same. For a list of candidate
addresses, a client will choose where to send media dependant on the
results of the STUN connectivity checks and associated priority
(highest priority wins). It should be noted that the inclusion of a
NAT displaying Address/Port Dependent properties does not
automatically result in relayed media. In fact, ICE processing will
avoid use of media with the exception of two clients which both
happen to be behind a NAT using Address/Port Dependent
characteristics. The connectivity checks and associated selection
algorithm enable traversal in this case. Figure 16 and following
description provide a guide as to how this is achieved using the ICE
connectivity checks. This is an abbreviated example that assumes
successful SIP offer/answer exchange and illustrates the connectivity
check flow.
L Port/Address Dependent Address Independent R
NAT NAT
|========================================================|
| SIP OFFER/ANSWER EXCHANGE |
|========================================================|
| | | |
| | |(1)Bind Req |
| | |<-----------------|
| | |Src=R=PRIV-1 |
| | |Dest=L-NAT-PUB-1 |
| | | |
| |(2)Bind Req | |
| x<-----------------| |
| |Src=R-NAT-PUB-1 | |
| |Dest=L-NAT-PUB-1 | |
| | | |
|(3)Bind Req | | |
|----------------->| | |
|Src=L-PRIV-1 | | |
|Dest=R-NAT-PUB-1 | | |
| | | |
| |(4)Bind Req | |
| |----------------->| |
| |Src=L-NAT-PUB-1 | |
| |Dest=R-NAT-PUB-1 | |
| | | |
| | |(5)Bind Req |
| | |----------------->|
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| | |Src=R-NAT-PUB-1 |
| | |Dest=R-PRIV-1 |
| | | |
| | |(6)Bind Resp |
| | |<-----------------|
| | |Src=R-PRIV-1 |
| | |Dest=L-NAT-PUB-1 |
| | | |
| |(7)Bind Resp | |
| |<-----------------| |
| |Src=R-NAT-PUB-1 | |
| |Dest=L-NAT-PUB-1 | |
| | | |
|(8)Bind Resp | | |
|<-----------------| | |
|Src=L-NAT-PUB-1 | | |
|Dest=L-PRIV-1 | | |
| | | |
| | |(9)Bind Req |
| | |<-----------------|
| | |Src=R-Priv-1 |
| | |Dest=L-NAT-PUB-1 |
| |(10)Bind Req | |
| |<-----------------| |
| |Src=R-NAT-PUB-1 | |
| |Dest=L-NAT-PUB-1 | |
| | | |
|(11)Bind Req | | |
|<-----------------| | |
|Src=L-NAT-PUB-1 | | |
| | | |
|(12)Bind Resp | | |
|----------------->| | |
|Src=L-PRIV-1 | | |
|Dest=L-NAT-PUB-1 | | |
| | | |
| |(13)Bind Resp | |
| |----------------->| |
| |Src=L-NAT-PUB-1 | |
| |Dest=R-NAT-PUB-1 | |
| | | |
| | |(14)Bind Resp |
| | |----------------->|
| | |Src=R-NAT-PUB-1 |
| | |Dest=R-PRIV-1 |
| | | |
|
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Figure 16: Single Port/Address Dependant NAT - Success
In this abbreviated example, Client R has already received a SIP
INVITE request and is starting its connectivity checks with Client L.
Client R generates a connectivity check (1) and sends to client L's
information as presented in the SDP offer. The request arrives at
client L's Port/Address dependent NAT and fails to traverse as there
is no security association. This would then move the connectivity
check to a failed state. In the mean time client L has received the
SDP answer in the SIP request and will also commence connectivity
checks. A check is dispatched (3) to Client R. The check is able to
traverse the NAT due to the association set up in the previously
failed check(1). The full Bind request/response is shown in steps
(3)-(8). As part of a candidate pair, Client R will now successfully
be able to complete the checks, as illustrated in steps (9)-(14).
The result is a successful pair of candidates that can be used
without the need to relay any media.
In conclusion, the only time media needs to be relayed is a result of
clients both behind Address/Port Dependant NAT type. As you can see
from the example in this section, neither side would be able to
complete connectivity checks with the exception of the Relayed
candidates.
5. IPv4-IPv6 Transition
This section describes how IPv6-only SIP user agents can communicate
with IPv4-only SIP user agents. While the techniques discussed in
this draft primarily contain examples of traversing NATs to allow
communications between hosts in private and public networks, they are
by no means limited to such scenarios. The same NAT traversal
techniques can also be used to establish communication in a
heterogeneous network environment -- e.g., communication between an
IPv4 host and an IPv6 host.
5.1. IPv4-IPv6 Transition for SIP Signaling
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
[I-D.camarillo-sipping-v6-transition]
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5.2. IPv4-IPv6 Transition for Media
Figure 17 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 the TURN usage 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 the TURN usage 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 18 explains how to use the TURN usage to obtain
an IPv4 address and how to use the ANAT semantics
[I-D.ietf-mmusic-anat] of the SDP grouping framework [RFC3388] 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|
+------+
Figure 17: IPv6-IPv4 transition scenario
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IPv6 SIP TURN IPv4 SIP
User Agent Server User Agent
| | |
| (1) STUN Allocate | |
| src=[2001:DB8::1]:30000 | |
|------------------------------------>| |
| (2) TURN Resp | |
| rel=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 18: IPv6-IPv4 translation with TURN
o The IPv6 SIP user agent obtains a TURN-derived IPv4 address by
issuing a STUN Allocate request (1). The STUN server generates a
response that contains a relayed IPv4 address using the TURN
usage. This IPv4 address maps to the IPv6 source address of the
STUN Allocate request, which the IPv6 address of the SIP user
agent. This results in any traffic being sent to the IPv4 address
provided by STUN server (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
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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 18.
6. Acknowledgments
The authors would like to thank the members of the IETF SIPPING WG
for their comments and suggestions. Detailed comments were provided
by Vijay Gurbani Francois Audet, kaiduan xie, Remi Denis-Courmont,
Hadriel Kaplan and Hans Persson.
7. References
7.1. Normative References
[RFC3261] 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.
[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation
Protocol (SIP): Locating SIP Servers", RFC 3263,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Boulton, Ed., et al. Expires October 27, 2008 [Page 58]
Internet-Draft NAT Scenarios April 2008
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC2327] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[RFC2766] Tsirtsis, G. and P. Srisuresh, "Network Address
Translation - Protocol Translation (NAT-PT)", RFC 2766,
February 2000.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the
Session Initiation Protocol (SIP) for Symmetric Response
Routing", RFC 3581, August 2003.
[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol
(SIP) Extension Header Field for Registering Non-Adjacent
Contacts", RFC 3327, December 2002.
[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H.
Schulzrinne, "Grouping of Media Lines in the Session
Description Protocol (SDP)", RFC 3388, December 2002.
[RFC3424] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 2002.
[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute
in Session Description Protocol (SDP)", RFC 3605,
October 2003.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
[I-D.ietf-sip-connect-reuse]
Mahy, R., Gurbani, V., and B. Tate, "Connection Reuse in
the Session Initiation Protocol (SIP)",
draft-ietf-sip-connect-reuse-09 (work in progress),
February 2008.
[I-D.ietf-behave-rfc3489bis]
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Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for (NAT) (STUN)",
draft-ietf-behave-rfc3489bis-15 (work in progress),
February 2008.
[I-D.ietf-behave-turn]
Rosenberg, J., Mahy, R., and P. Matthews, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)",
draft-ietf-behave-turn-07 (work in progress),
February 2008.
[I-D.ietf-sip-outbound]
Jennings, C. and R. Mahy, "Managing Client Initiated
Connections in the Session Initiation Protocol (SIP)",
draft-ietf-sip-outbound-13 (work in progress), March 2008.
[I-D.ietf-sip-gruu]
Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-15 (work in progress),
October 2007.
[I-D.ietf-mmusic-ice]
Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-19 (work in progress), October 2007.
[I-D.ietf-mmusic-anat]
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.
[I-D.ietf-mmusic-ice-tcp]
Rosenberg, J., "TCP Candidates with Interactive
Connectivity Establishment (ICE)",
draft-ietf-mmusic-ice-tcp-06 (work in progress),
February 2008.
[I-D.ietf-avt-rtp-and-rtcp-mux]
Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port",
draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress),
August 2007.
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7.2. Informative References
[I-D.camarillo-sipping-v6-transition]
Camarillo, G., "IPv6 Transcition in the Session Initiation
Protocol (SIP)", draft-camarillo-sipping-v6-transition-00
(work in progress), February 2005.
Authors' Addresses
Chris Boulton
Avaya
Eastern Business Park
St Mellons
Cardiff, South Wales CF3 5EA
Email: cboulton@avaya.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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