SIPPING Working Group C. Boulton, Ed.
Internet-Draft Avaya
Intended status: BCP J. Rosenberg
Expires: March 21, 2009 Cisco Systems
G. Camarillo
Ericsson
F. Audet
Nortel
September 17, 2008
Best Current Practices for NAT Traversal for Client-Server SIP
draft-ietf-sipping-nat-scenarios-09
Status of this Memo
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Abstract
Traversal of the Session Initiation Protocol (SIP) and the sessions
it establishes through Network Address Translators (NATs) 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. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 4
4. Solution Technology Outline Description . . . . . . . . . . . 7
4.1. SIP Signaling . . . . . . . . . . . . . . . . . . . . . . 8
4.1.1. Symmetric Response . . . . . . . . . . . . . . . . . . 8
4.1.2. Client Initiated Connections . . . . . . . . . . . . . 9
4.2. Media Traversal . . . . . . . . . . . . . . . . . . . . . 9
4.2.1. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . 10
4.2.2. RTCP . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.2.3. ICE/STUN/TURN . . . . . . . . . . . . . . . . . . . . 10
5. NAT Traversal Scenarios . . . . . . . . . . . . . . . . . . . 12
5.1. Basic NAT SIP Signaling Traversal . . . . . . . . . . . . 12
5.1.1. Registration (Registrar/Edge Proxy Co-Located) . . . . 12
5.1.2. Registration(Registrar/Edge Proxy not Co-Located) . . 16
5.1.3. Initiating a Session . . . . . . . . . . . . . . . . . 19
5.1.4. Receiving an Invitation to a Session . . . . . . . . . 22
5.2. Basic NAT Media Traversal . . . . . . . . . . . . . . . . 26
5.2.1. Endpoint Independent NAT . . . . . . . . . . . . . . . 27
5.2.2. Address and Port Dependant NAT . . . . . . . . . . . . 46
6. IPv4-IPv6 Transition . . . . . . . . . . . . . . . . . . . . . 54
6.1. IPv4-IPv6 Transition for SIP Signaling . . . . . . . . . . 54
6.2. IPv4-IPv6 Transition for Media . . . . . . . . . . . . . . 55
7. Security Considerations . . . . . . . . . . . . . . . . . . . 56
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 57
9. IAB Considerations . . . . . . . . . . . . . . . . . . . . . . 57
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 57
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 57
11.1. Normative References . . . . . . . . . . . . . . . . . . . 57
11.2. Informative References . . . . . . . . . . . . . . . . . . 59
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 60
Intellectual Property and Copyright Statements . . . . . . . . . . 61
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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. Details of different
NATs behavior can be found in 'NAT Behavioral Requirements for
Unicast UDP' [RFC4787].
The IETF has been active on many specifications for the traversal of
NATs, 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 NATs 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 4).
The best practices described in this document are for traditional
"client- server"-style SIP. It seems likely that other groups using
SIP, for example P2PSIP, will recommend these same practices between
a P2PSIP client and a P2PSIP peer, but will recommend different
practices for use between peers in a peer-to-peer network.
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. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
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3. Problem Statement
The traversal of SIP through NATs can be split into two categories
that both require attention - The core SIP signaling and associated
media traversal. This document assumes NATs that do not contain SIP-
aware Application Layer Gateways (ALG). Some NATs that are available
today contain such behavior that which makes much of the issues
discussed in the document not applicable. It should also be noted
that Session Border Controllers (SBC) doing 'hosted NAT traversal'
also makes many of the discussions in this document moot. More
information can be obtained from [I-D.ietf-sipping-sbc-funcs] and
[I-D.ietf-mmusic-media-path-middleboxes].
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
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
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Secondly, there are two cases where new requests re-use existing
connections. The first is 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. The second case is when 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.
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 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 4 of this document is
relevant for all SIP signaling in conjunction with connection reuse,
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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.
RTP [RFC3550] runs over UDP and 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) [RFC4566] 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
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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 unified approach. This
problem exists for all media transport protocols that might be NATted
(e.g., TCP, UDP, SCTP, DCCP).
In general the problems associated with NAT traversal can be
categorized as follows.
For signaling:
o Responses do not re-use the NAT mapping and filtering entries
created by the request.
o Inbound requests are filtered out by the NAT because there is no
long-term connection between the client and the proxy.
For media:
o Each endpoint has a variety of addresses. In different
situations, a different pair of (local endpoint, remote endpoint)
addresses should be used, and it is not clear when to use which
pair.
o Many NATs filter inbound packets if the local endpoint has not
recently sent an outbound packet to the sender [same problem as
second one under signaling].
o Classic RTCP usage is to run RTCP on the next highest port.
However, NATs do not necessarily preserve port adjacency.
o Classic RTP and RTCP usage is to use different 5-tuples for
traffic in each direction. Though not really a problem, doing
this through NATs is more work than using the same 5-tuple in both
directions.
4. 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 NATs, and enabling media as specified by SDP in a
SIP offer/answer exchange to flow between endpoints.
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4.1. SIP Signaling
SIP signaling has two areas that result in transactional failure when
traversing through NATs, as described in Section 3 of this document.
The remaining sub-sections describe appropriate solutions that result
in SIP signaling traversal through NATs, regardless of transport
protocol. It is RECOMMENDED that SIP compliant entities follow the
guidelines presented in this section to enable traversal of SIP
signaling through NATs.
4.1.1. Symmetric Response
As described in Section 3 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 NATs. 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 RFC 3581 [RFC3581] such as listening and
sending SIP requests/responses from the same port.
4.1.2. Client Initiated Connections
The second problem with SIP signaling, as defined in Section 3 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 NATs are documented in 'Managing Client
Initiated Connections in the Session Initiation
Protocol(SIP)'[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
connection (both UDP and TCP) that is 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 NATs. It
also provides a keepalive mechanism for clients to keep NATs bindings
alive. This is achieved by multiplexing a ping/pong mechanism over
the SIP signaling connection (STUN for UDP and CRLF/operating system
keepalive for reliable transports like TCP). Usage of
[I-D.ietf-sip-outbound] 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.
4.2. Media Traversal
The issues of media traversal through NATs is not straight forward
and requires the combination of a number of traversal methodologies.
The technologies outlined in the remainder of this section provide
the required solution set.
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4.2.1. Symmetric RTP/RTCP
The primary problem identified in Section 3 of this document is that
internal IP address/port combinations can not be reached from the
public side of NATs. In the case of media such as RTP, this will
result in no audio traversing NATs (as illustrated in Figure 3). To
overcome this problem, a technique called 'Symmetric RTP/
RTCP'[RFC4961] can be used. This involves a SIP endpoint both
sending and receiving RTP/RTCP traffic from the same IP address/port
combination. 'Symmetric RTP/RTCP' SHOULD only be used for traversal
of RTP through NATs when one of the participants in a media session
definitively knows that it is on the public network and is using a
'latching' technique as described in
[I-D.ietf-mmusic-media-path-middleboxes]. Symmetric RTP/RTCP is
important for everything that might want to traverse a NAT or speak
with an endpoint that is behind a NAT - even if the remote endpoint
is an SBC performing 'hosted NAT traversal'.
4.2.2. RTCP
Normal practice when selecting a port for defining RTP 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 certain types
of 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).
The use of RFC 3605 [RFC3605] MUST be supported. 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 but is still immature.
4.2.3. ICE/STUN/TURN
ICE, STUN and TURN are a suite of 3 inter-related protocols that
combine to provide a complete media traversal solution for NATs. The
following sections provide details of each component part.
4.2.3.1. STUN
Session Traversal Utilities for NAT or STUN is defined in RFC 3489bis
[I-D.ietf-behave-rfc3489bis]. STUN is a lightweight tool kit and
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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 NATs. 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 '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 NATs 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 4.1.2, STUN is also used as a client-to-
server keep-alive mechanism to refresh NAT bindings.
4.2.3.2. TURN
As described in the Section 4.2.3.1, the STUN protocol does not work
for UDP traversal through certain identified NAT mappings.
'Traversal Using Relays around NAT' is a usage of the STUN protocol
for deriving (from a TURN server) an address that will be used to
relay packets towards a client. TURN provides an external address
(globally routable) at a STUN server that will act as a media relay
which attempts to allow traffic to 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
would 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.
4.2.3.3. ICE
Interactive Connectivity Establishment (ICE) is the RECOMMENDED
method for traversal of existing NATs 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 (*note - an ICE endpoint can also use
non-IETF mechanisms (e.g., NAT-PMP, UPnP IGD) to learn public IP
addresses and ports, and populate a=candidate lines with that
information). 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
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connection information including a media address, priority and
transport protocol. The appropriate IP address/port combinations are
used in the order specified by the priority. A client compliant to
the ICE specification will then locally run STUN servers on all
addresses being advertised using ICE. Each instance will undertake
connectivity checks to ensure that a client can successfully receive
media on the advertised address. Only connections that pass the
relevant connectivity checks are used for media exchange. The full
details of the ICE methodology are contained in
[I-D.ietf-mmusic-ice].
5. NAT Traversal Scenarios
This section of the document includes detailed NAT traversal
scenarios for both SIP signaling and the associated media.
Signalling NAT traversal is achieved using [I-D.ietf-sip-outbound].
5.1. Basic NAT SIP Signaling Traversal
The following sub-sections concentrate on SIP signaling traversal of
NATs. The scenarios include traversal for both reliable and un-
reliable transport protocols.
5.1.1. Registration (Registrar/Edge Proxy Co-Located)
The set of scenarios in this section document basic signaling
traversal of a SIP REGISTER method through NATs.
5.1.1.1. UDP
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Registrar/
Bob NAT Edge Proxy
| | |
|(1) REGISTER | |
|----------------->| |
| | |
| |(1) REGISTER |
| |----------------->|
| | |
|*************************************|
| Create Outbound Connection Tuple |
|*************************************|
| | |
| |(2) 200 OK |
| |<-----------------|
| | |
|(2) 200 OK | |
|<-----------------| |
| | |
Figure 5: UDP Registration
In this example the client sends a SIP REGISTER request through a
NAT. The client will include an 'rport' parameter as described in
Section 4.1.1 of this document for allowing traversal of UDP
responses. The original request as illustrated in (1) in Figure 5 is
a standard REGISTER message:
Message 1:
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Contact: <sip:bob@192.168.1.2 >;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Content-Length: 0
This SIP transaction now generates a SIP 200 OK response, as depicted
in (2) from Figure 5:
Message 2:
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SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2;rport=8050;branch=z9hG4bKnashds7;
received=172.16.3.4
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>;tag=6AF99445E44A
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Require: outbound
Contact: <sip:bob@192.168.1.2 >;reg-id=1;expires=3600
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Content-Length: 0
The response will be sent to the address appearing in the 'received'
parameter of the SIP 'Via' header (address 172.16.3.4). 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 'reg-id' 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 URIs to represent a globally routable
address as perceived on the public side of a NAT.
5.1.1.2. Connection Oriented Transport
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Registrar/
Bob NAT Edge Proxy
| | |
|(1) REGISTER | |
|----------------->| |
| | |
| |(1) REGISTER |
| |----------------->|
| | |
|*************************************|
| Create Outbound Connection Tuple |
|*************************************|
| | |
| |(2) 200 OK |
| |<-----------------|
| | |
|(2) 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, beyond the procedures
defined in [I-D.ietf-sip-outbound]. SIP responses will re-use the
connection created for the initial REGISTER request, (1) from
Figure 6:
Message 1:
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Contact: <sip:bob@192.168.1.2;transport=tcp>;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Content-Length: 0
Message 2:
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SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>;tag=6AF99445E44A
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Require: outbound
Contact: <sip:bob@192.168.1.2;transport=tcp>;reg-id=1;expires=3600
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Content-Length: 0
This example was included to show the inclusion of the +sip.instance
Contact header parameter 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.
5.1.2. Registration(Registrar/Edge 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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Bob NAT Edge Proxy Registrar
| | | |
|(1) REGISTER | | |
|----------------->| | |
| | | |
| |(1) REGISTER | |
| |----------------->| |
| | | |
| | |(2) REGISTER |
| | |----------------->|
| | | |
|********************************************************|
| Create Outbound Connection Tuple |
|********************************************************|
| | | |
| | |(3) 200 OK |
| | |<-----------------|
| |(4)200 OK | |
| |<-----------------| |
| | | |
|(4)200 OK | | |
|<-----------------| | |
| | | |
Figure 7: Registration(Registrar/Proxy not Co-Located)
This scenario builds on the previous example contained in
Section 5.1.1.2. The primary difference being that the REGISTER
request is routed onwards from a Proxy Server to a separated
Registrar. The important message to note is (1) 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 proxy needs to indicate its need
to remain in the SIP signaling path. To achieve this the proxy
inserts to REGISTER message (2) 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 (1) includes a
SIP Route header for the edge proxy.
Message 1:
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REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Route: <sip:ep1.example.com;lr>
Contact: <sip:bob@192.168.1.2;transport=tcp>;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Content-Length: 0
When proxied in (2) looks as follows:
Message 2:
REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/TCP ep1.example.com;branch=z9hG4bKnuiqisi
Via: SIP/2.0/TCP 192.168.1.2;branch=z9hG4bKnashds7
Max-Forwards: 69
From: Bob <sip:bob@example.com>;tag=7F94778B653B
To: Bob <sip:bob@example.com>
Call-ID: 16CB75F21C70
CSeq: 1 REGISTER
Supported: path, outbound
Contact: <sip:bob@192.168.1.2;transport=tcp>;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF>"
Path: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob>
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 all but 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 5.1.4.2 of this document.
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5.1.3. Initiating a Session
This section covers basic SIP signaling when initiating a call from
behind a NAT.
5.1.3.1. UDP
Initiating a call using UDP (the Edge Proxy and Authoritative Proxy
funcationality are co-located).
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Edge Proxy/
Bob NAT Auth. Proxy Alice
| | | |
|(1) INVITE | | |
|----------------->| | |
| | | |
| |(1) INVITE | |
| |----------------->| |
| | | |
| | |(2) INVITE |
| | |---------------->|
| | | |
| | |(3)180 RINGING |
| | |<----------------|
| | | |
| |(4)180 RINGING | |
| |<-----------------| |
| | | |
|(4)180 RINGING | | |
|<-----------------| | |
| | | |
| | |(5)200 OK |
| | |<----------------|
| | | |
| |(6)200 OK | |
| |<-----------------| |
| | | |
|(6)200 OK | | |
|<-----------------| | |
| | | |
|(7)ACK | | |
|----------------->| | |
| | | |
| |(7)ACK | |
| |----------------->| |
| | | |
| | |(8) ACK |
| | |---------------->|
| | | |
Figure 8: Initiating a Session - UDP
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:
Message 1:
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INVITE sip:alice@a.example SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Bob <sip:bob@example.com>;tag=ldw22z
To: Alice <sip:alice@a.example>
Call-ID: 95KGsk2V/Eis9LcpBYy3
CSeq: 1 INVITE
Supported: outbound
Route: <sip:ep1.example.com;lr>
Contact: <sip:bob@192.168.1.2;ob>
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 5.1.1.1,
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 4.1.1 and defined
in RFC 3581 [RFC3581].
2. Secondly, the contact inserted contains to ensure that all new
requests will be sent to the same flow. Alternatively, a GRUU
might have been used. See 4.3/[I-D.ietf-sip-outbound].
In (2), the proxy inserts itself in the Via, adds the rport port
number in the previous Via header, adds the received parameter in the
previous Via, removes the Route header, and inserts a Record-Route
with a token.
Message 2:
INVITE sip:alice@172.16.1.4 SIP/2.0
Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4bKnuiqisi
Via: SIP/2.0/UDP 192.168.1.2;rport=8050;branch=z9hG4bKnashds7;
received=172.16.3.4
Max-Forwards: 69
From: Bob <sip:bob@example.com>;tag=ldw22z
To: Alice <sip:alice@a.example>
Call-ID: 95KGsk2V/Eis9LcpBYy3
CSeq: 1 INVITE
Supported: outbound
Record-Route: <sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr>
Contact: <sip:bob@192.168.1.2;ob>
Content-Type: application/sdp
Content-Length: ...
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[SDP not shown]
5.1.3.2. Connection-oriented 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 5.1.3.1. 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. See section 9.5/
[I-D.ietf-sip-outbound] for an example call flow of an outgoing call.
5.1.4. 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 5.1.1 and Section 5.1.2
in this document.
5.1.4.1. Registrar/Proxy Co-located
The SIP signaling on the interior of the network (behind the user's
proxy) 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 5.1.1.1.
Edge Proxy
Bob NAT Auth. Proxy Alice
| | | |
|*******************************************************|
| Registration Binding Installed in |
| section 5.1.1.1 |
|*******************************************************|
| | | |
| | |(1)INVITE |
| | |<----------------|
| | | |
| |(2)INVITE | |
| |<-----------------| |
| | | |
|(2)INVITE | | |
|<-----------------| | |
| | | |
| | | |
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Figure 9: Receiving an Invitation to a Session
An INVITE request arrives at the Authoritative Proxy with a
destination pointing to the AOR of that inserted in Section 5.1.1.1.
The message is illustrated by (1) in Figure 9 and looks as follows:
INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: External Alice <sip:alice@example.com>;tag=02935
To: Bob <sip:bob@example.com>
Call-ID: klmvCxVWGp6MxJp2T2mb
CSeq: 1 INVITE
Contact: <sip:alice@172.16.1.4>
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
message (2) from Figure 9 and is as follows:
INVITE sip:bob@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4kmlds893jhsd
Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d
Max-Forwards: 69
From: Alice <sip:alice@example.com>;tag=02935
To: client bob <sip:bob@example.com>
Call-ID: klmvCxVWGp6MxJp2T2mb
CSeq: 1 INVITE
Contact: <sip:alice@172.16.1.4>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
It is a standard SIP INVITE request with no additional functionality.
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The major difference being that this request will not be forwarded to
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.
5.1.4.2. Edge Proxy/Authoritative 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 UDP and follows on from the
registration installed in the example from Section 5.1.2.
Bob NAT Edge Proxy Auth. Proxy Alice
| | | | |
|***********************************************************|
| Registration Binding Installed in |
| section 5.1.2 |
|***********************************************************|
| | | | |
| | | |(1)INVITE |
| | | |<-------------|
| | | | |
| | |(2)INVITE | |
| | |<-------------| |
| | | | |
| |(3)INVITE | | |
| |<-------------| | |
| | | | |
|(3)INVITE | | | |
|<-------------| | | |
| | | | |
| | | | |
Figure 10: Registrar/Proxy Not Co-located
An INVITE request arrives at the Authoritative Proxy with a
destination pointing to the AOR of that inserted in Section 5.1.2.
The message is illustrated by (1) in Figure 10 and looks as follows:
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INVITE sip:bob@example.com SIP/2.0
Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: Alice <sip:alice@example.com>;tag=02935
To: Bob <sip:bob@example.com>
Call-ID: klmvCxVWGp6MxJp2T2mb
CSeq: 1 INVITE
Contact: <sip:external@172.16.1.4>
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
Route header as shown in (2) from Figure 10:
INVITE sip:bob@client.example.com SIP/2.0
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bK74fmljnc
Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d
Route: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob>
Max-Forwards: 69
From: Alice <sip:alice@example.net>;tag=02935
To: Bob <sip:Bob@example.com>
Call-ID: klmvCxVWGp6MxJp2T2mb
CSeq: 1 INVITE
Contact: <sip:alice@172.16.1.4>
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:
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INVITE sip:bob@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP ep1.example.com;branch=z9hG4nsi30dncmnl
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4bK74fmljnc
Via: SIP/2.0/UDP 172.16.1.4;branch=z9hG4bK74huHJ37d
Record-Route: <sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr>
Max-Forwards: 68
From: Alice <sip:Alice@example.net>;tag=02935
To: bob <sip:bob@example.com>
Call-ID: klmvCxVWGp6MxJp2T2mb
CSeq: 1 INVITE
Contact: <sip:alice@172.16.1.4>
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
[RFC3263] are overridden). This then allows the original connection/
mapping from the initial registration to the outbound proxy to be re-
used.
5.2. 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 [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
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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 of the left hand side
NAT while "R-NAT-PUB-1" would represent a public network address of
the right hand side of the NAT. "L-PRIV-1" would represent a private
network address of the left hand side of the NAT while "R-PRIV-1"
represents a private address of 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
of the representation. For example, 'L-PRIV-1.address' and 'L-PRIV-
1.port'.
The use of '$' signifies variable parts in example SIP messages.
5.2.1. 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.
5.2.1.1. 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 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.
5.2.1.1.1. 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
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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 | |
| | | |
|(4) BIND Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|Map=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 | | |
|<-----------------| | |
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|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<<<<<<<<<<<<|
|========================================================|
| |
|========================================================|
|>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>|
|========================================================|
| |
|========================================================|
|<<<<<<<<<<<<Incoming RTCP sent to NAT-PUB-2<<<<<<<<<<<<<|
|========================================================|
| | | |
|(13)SIP ACK | | |
|----------------->| | |
| | | |
| |(14) SIP ACK | |
| |------------------------------------>|
| | | |
Figure 11: 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 11
which also highlights the source address and port for which the
client device wishes to obtain a mapping. The 'Binding Discovery'
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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 a second 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 Section 5.1. The INVITE request will use the addresses it has
obtained in the previous STUN transactions to populate the SDP of
the SIP INVITE as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 $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 $NAT-PUB-1.port RTP/AVP 0). Finally,
the port gained from the additional 'Binding Discovery' is placed
in the RTCP attribute (as discussed in Section 4.2.2) for
traversal of RTCP (a=rtcp:$NAT-PUB-2.port).
o The SIP signaling then traverses the NAT and sets up the SIP
session (9-12). Note that the left 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 to the left hand client
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.
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5.2.1.1.2. 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 5.2.1.1.1. Figure 12 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 | |
| |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 | |
| |----------------->| |
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| | | |
| |(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<<<<<<<<<<<<|
|========================================================|
| | | |
|========================================================|
|>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2>>>>>>>>>>>>|
|========================================================|
| | | |
|========================================================|
|<<<<<<<<<<<<<Incoming RTCP sent to L-PRIV-2<<<<<<<<<<<<<|
|========================================================|
| | | |
| | |(13)SIP ACK |
| |<------------------------------------|
| | | |
|(14)SIP ACK | | |
|<-----------------| | |
| | | |
Figure 12: 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
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(3) from Figure 12 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 a second 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 Section 5.1. 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 second 'Binding Discovery'
is placed in the RTCP attribute (as discussed in Section 4.2.2)
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 left hand 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 from the right hand client
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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.
5.2.1.2. ICE Solution
The preferred solution for media traversal of NAT is using ICE, as
described in Section 4.2.3.3, 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.
5.2.1.2.1. Initiating Session
The following example demonstrates an initiating traversal through an
'Endpoint independent' NAT using ICE.
L NAT TURN NAT R
Server
| | | | |
|(1) Alloc Req | | | |
|Src=L-PRIV-1 | | | |
|Dest=TURN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(2) Alloc Req | | |
| |Src=L-NAT-PUB-1 | | |
| |Dest=TURN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(3) Alloc Resp | | |
| |<---------------| | |
| |Src=TURN-PUB-1 | | |
| |Dest=L-NAT-PUB-1| | |
| |Map=L-NAT-PUB-1 | | |
| |Rel=TURN-PUB-2 | | |
| | | | |
|(4) Alloc Resp | | | |
|<---------------| | | |
|Src=TURN-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
|Rel=TURN-PUB-2 | | | |
| | | | |
|(5) Alloc Req | | | |
|Src=L-PRIV-2 | | | |
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|Dest=TURN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(6) Alloc Req | | |
| |Src=L-NAT-PUB-2 | | |
| |Dest=TURN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(7) Alloc Resp | | |
| |<---------------| | |
| |Src=TURN-PUB-1 | | |
| |Dest=NAT-PUB-2 | | |
| |Map=NAT-PUB-2 | | |
| |Rel=TURN-PUB-3 | | |
| | | | |
|(8) Alloc Resp | | | |
|<---------------| | | |
|Src=TURN-PUB-1 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
|Rel=TURN-PUB-3 | | | |
| | | | |
|(9) SIP INVITE | | | |
|------------------------------------------------->| |
| | | | |
| | | |(10) SIP INVITE |
| | | |--------------->|
| | | | |
| | | |(11) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=TURN-PUB-1 |
| | | | |
| | |(12) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=TURN-PUB-1 | |
| | | | |
| | |(13) Alloc Res | |
| | |--------------->| |
| | |Src=TURN-PUB-1 | |
| | |Dest=R-NAT-PUB-1| |
| | |Map=R-NAT-PUB-1 | |
| | |Rel=TURN-PUB-4 | |
| | | | |
| | | |(14) Alloc Res |
| | | |--------------->|
| | | |Src=TURN-PUB-1 |
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| | | |Dest=R-PRIV-1 |
| | | |Map=R-NAT-PUB-1 |
| | | |Rel=TURN-PUB-4 |
| | | | |
| | | |(15) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=TURN-PUB-1 |
| | | | |
| | |(16) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=TURN-PUB-1 | |
| | | | |
| | |(17) Alloc Res | |
| | |--------------->| |
| | |Src=TURN-PUB-1 | |
| | |Dest=R-NAT-PUB-2| |
| | |Map=R-NAT-PUB-2 | |
| | |Rel=TURN-PUB-5 | |
| | | | |
| | | |(18) Alloc Res |
| | | |--------------->|
| | | |Src=TURN-PUB-1 |
| | | |Dest=R-PRIV-2 |
| | | |Map=R-NAT-PUB-2 |
| | | |Rel=TURN-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 | | | |
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|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 | |
| | | | |
|(29) Bind Res | | | |
|<---------------| | | |
|Src=R-NAT-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
| | | | |
|===================================================================|
|>>>>>>>>>>>>>>>>>>Outgoing RTP sent from L-PRIV-1 >>>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(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| |
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| | | | |
|(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 |
| | | | |
|===================================================================|
|<<<<<<<<<<<<<<<<<<Outgoing RTP sent from R-PRIV-1 <<<<<<<<<<<<<<<<<|
|===================================================================|
|(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 |
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| | | |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| | |
| | | | |
| | | |(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 | | | |
| | | | |
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|===================================================================|
|>>>>>>>>>>>>>>>>>>Outgoing RTCP sent from L-PRIV-2 >>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(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 | | | |
| | | | |
| |(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 R-PRIV-2<<<<<<<<<<<<<<<<<|
|===================================================================|
|(55) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
|USE-CANDIDATE | | | |
| | | | |
| |(56) Bind Req | | |
| |-------------------------------->| |
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| |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 | | | |
|<---------------| | | |
|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 |
| | | |--------------->|
| | | | |
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Figure 13: 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 TURN
Allocate request as indicated in (1) from Figure 13 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 to the public internet
towards the public TURN 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 TURN server receives the Allocate request and processes it
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 TURN-relayed address that
can be used at the STUN server (see 'Rel=TURN-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
in (5)-(8) in Figure 13. 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 14:
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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 2130706431 $L-PRIV-1.address $L-PRIV-1.port
typ host
a=candidate:$L1 2 UDP 2130706430 $L-PRIV-2.address $L-PRIV-2.port
typ host
a=candidate:$L2 1 UDP 1694498815 $L-NAT-PUB-1.address $L-NAT-PUB-1.port
typ srflx raddr $L-PRIV-1.address rport $L-PRIV-1.port
a=candidate:$L2 2 UDP 1694498814 $L-NAT-PUB-2.address $L-NAT-PUB-2.port
typ srflx raddr $L-PRIV-1.address rport $L-PRIV-2.port
a=candidate:$L3 1 UDP 16777215 $STUN-PUB-2.address $STUN-PUB-2.port
typ relay raddr $L-PRIV-1.address rport $L-PRIV-1.port
a=candidate:$L3 2 UDP 16777214 $STUN-PUB-3.address $STUN-PUB-3.port
typ relay raddr $L-PRIV-1.address rport $L-PRIV-2.port
Figure 14: 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
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
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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 14:
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 2130706431 $R-PRIV-1.address $R-PRIV-1.port
typ host
a=candidate:$L1 2 UDP 2130706430 $R-PRIV-2.address $R-PRIV-2.port
typ host
a=candidate:$L2 1 UDP 1694498815 $R-NAT-PUB-1.address $R-NAT-PUB-1.port
typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port
a=candidate:$L2 2 UDP 1694498814 $R-NAT-PUB-2.address $R-NAT-PUB-2.port
typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port
a=candidate:$L3 1 UDP 16777215 $STUN-PUB-2.address $STUN-PUB-4.port
typ relay raddr $R-PRIV-1.address rport $R-PRIV-1.port
a=candidate:$L3 2 UDP 16777214 $STUN-PUB-3.address $STUN-PUB-5.port
typ relay raddr $R-PRIV-1.address rport $R-PRIV-1.port
Figure 15: 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.
The User Agents then scan the list and move the candidates to the
'Waiting' state. At this point both clients will periodically,
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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 send media to the peer using the
candidate pair. Immediately after sending its 200 Okay, 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 (31-36) 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
a=candidate:$L2 1 UDP 2203948363 $L-NAT-PUB-1.address $L-NAT-PUB-1.port
typ srflx raddr $L-PRIV-1.address rport $L-PRIV-1.port
a=candidate:$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
Figure 16: 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
a=candidate:$L2 1 UDP 2984756463 $R-NAT-PUB-1.address $R-NAT-PUB-1.port
typ srflx raddr $R-PRIV-1.address rport $R-PRIV-1.port
a=candidate:$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
Figure 17: ICE SDP Updated Answer
5.2.2. Address and Port Dependant NAT
5.2.2.1. STUN Failure
This section highlights that while using STUN techniques 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 18: Port/Address-Dependant NAT with STUN - Failure
The example in Figure 18 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 18 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 5.2.1.1.1).
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 18 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 18 shows the media being dropped at the NAT after (7)
and before (8). This then leads to one way audio.
5.2.2.2. TURN Solution
As identified in Section Section 5.2.2.1, STUN provides a useful tool
for the traversal of the majority of NATs but fails with Port/Address
Dependent NAT. The TURN extensions [I-D.ietf-behave-turn] address
this scenario. TURN extends STUN to allow a client to request a
relayed address at the TURN server rather than a reflexive
representation. This then introduces a media relay in the path for
NAT traversal (as described in Section 4.2.3.2). The following
example explains how TURN solves the previous failure when using STUN
to traverse a 'Port/ Address Dependent' type NAT.
L Port/Address-Dependant STUN [..]
NAT Server
| | | |
|(1) Alloc Req | | |
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|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 | | |
| | | |
|(9)SIP INVITE | | |
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|----------------->| | |
| | | |
| |(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 19: Port/Address-Dependant NAT with TURN - 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
(3) that contains both a server reflexive address (Map=NAT-PUB-1)
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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 TURN 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)):
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 19 (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 TURN 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.
5.2.2.3. ICE Solution
The previous two examples have highlighted the problem with using
core STUN for all forms of NAT traversal and a solution using TURN
for the Address/Port-Dependent NAT case. The RECOMMENDED mechanism
for traversing all varieties of NAT is using ICE, as detailed in
Section 4.2.3.3. ICE makes use of core STUN, TURN 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 in Section 5.2.1.2.1. 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
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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 relay 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 20 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 Endpoint-Independent R
L-NAT R-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 |
| | |----------------->|
| | |Src=L-NAT-PUB-1 |
| | |Dest=R-PRIV-1 |
| | | |
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| | |(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=R-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=R-NAT-PUB-1 | | |
|Dest=L-PRIV-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=L-NAT-PUB-1 |
| | |Dest=R-PRIV-1 |
| | | |
|
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Figure 20: 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 NAT binding. 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.
6. 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.
6.1. 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.ietf-sipping-v6-transition]
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6.2. IPv4-IPv6 Transition for Media
Figure 21 shows a network of IPv6 SIP user agents that has a relay
with a pool of public IPv4 addresses. The IPv6 SIP user agents of
this IPv6 network need to communicate with users on the IPv4
Internet. To do so, the IPv6 SIP user agents use TURN to obtain a
set of public IPv4 address and port pairs from the relay (for RTP and
RTCP). The mechanism that an IPv6 SIP user agent follows to obtain
public IPv4 address and port pairs from a relay using TURN is the
same as the one followed by a user agent with a private IPv4 address
to obtain public IPv4 address and port pairs. The example below
explains how a UA in an IPv6-only network can use ICE
[I-D.ietf-mmusic-ice] to communicate with a SIP Phone in an IPv4-only
network. Note that no server reflective addresses are used in this
example.
+----------+
| / \ |
/SIP \
/Phone \
/ \
------------
| |
| |
192.0.2.2:25000 | | 192.0.2.2:25123
RTP RTCP
+-------------+
| TURN Server |
+-------------+
IPv4 Network | |
+---------+
| |
----------------------| NAT |---------------------
| |
+---------+
IPv6 Network | |
| |
| |
[2001:DB8::1]:30000 RTP RTCP [2001:DB8::1]:30001
+----------+
| IPv6 SIP |
| UA |
+----------+
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Figure 21: IPv6-IPv4 transition scenario
The IPv6 UA obtains a TURN-derived IPv4 address and port pair for its
RTP port and another one for its RTCP port by issuing 2 TURN Allocate
requests. The TURN server generates responses containing relayed
IPv4 addressee and port pairs for both RTP and RTCP ports. These
IPv4 addresses and port pairs map to the IPv6 source addressee and
port pairs. The result of any UDP packets sent to the IPv4 address
and port pairs provided by the TURN server (i.e., 192.0.2.2:25000 for
RTP and 192.0.2.2:25123 for RTCP) with be redirected to the IPv6 IP
address and port pairs of the SIP UA (i.e., [2001:DB8::1]:30000 for
RTP and [2001:DB8::1]:30001 for RTCP).
When the UA builds the original Offer, it includes 2 candidates: one
for the host IPv6 address and another for the relay IPv4 address.
When computing the priority for the candidate, we will use a type
preference of 126 for the host address candidate, and of 0 for the
relay address candidate, a local preference of 65535 for both
candidates, and a component ID of 1 for RTP and 2 for RTCP for both
candidates. This will generate a priority of 2130706431 for the host
address, and of 16777215 for the relay address. The default
candidate is the relay address candidate. The Offer will look as
follows.
v=0
o=test 2890844342 2890842164 IN IP6 2001:DB8::1
c=IN IP4 192.0.2.2
t=0 0
a=ice-pwd:asd88fgpdd777uzjYhagZg
a=ice-ufrag:8hhY
m=audio 25000 RTP/AVP 0
a=rtcp:25123
a=candidate:1 1 UDP 2130706431 [2001:DB8::1] 30000 typ host
a=candidate:1 2 UDP 2130706430 [2001:DB8::1] 30001 typ host
a=candidate:2 1 UDP 16777215 192.0.2.2 25000 typ relay
raddr [2001:DB8::1] rport 30000
a=candidate:2 2 UDP 16777214 192.0.2.2 25123 typ relay
raddr [2001:DB8::1] rport 30001
The Offer is sent in an INVITE request which gets routed to the IPv4-
only UA, which will choose the IPv4 candidate as per normal ICE
procedures.
7. Security Considerations
There are no Security Considerations beyond the ones inherited by
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reference.
8. IANA Considerations
There are no IANA Considerations.
9. IAB Considerations
There are no IAB considerations.
10. Acknowledgments
The authors would like to thank the members of the IETF SIPPING WG
for their comments and suggestions. Expert review and contribution
was provided by Francois Audet.
Detailed comments were provided by Vijay Gurbani, kaiduan xie, Remi
Denis-Courmont, Hadriel Kaplan, Phillip Matthews, Dan Wing, Spencer
Dawkins and Hans Persson.
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC2766] Tsirtsis, G. and P. Srisuresh, "Network Address
Boulton, Ed., et al. Expires March 21, 2009 [Page 57]
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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.
[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-11 (work in progress),
July 2008.
[I-D.ietf-behave-rfc3489bis]
Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for (NAT) (STUN)",
draft-ietf-behave-rfc3489bis-18 (work in progress),
July 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-09 (work in progress), July 2008.
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[I-D.ietf-sip-outbound]
Jennings, C. and R. Mahy, "Managing Client Initiated
Connections in the Session Initiation Protocol (SIP)",
draft-ietf-sip-outbound-15 (work in progress), June 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-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.
11.2. Informative References
[I-D.ietf-sipping-sbc-funcs]
Hautakorpi, J., Camarillo, G., Penfield, B., Hawrylyshen,
A., and M. Bhatia, "Requirements from SIP (Session
Initiation Protocol) Session Border Control Deployments",
draft-ietf-sipping-sbc-funcs-06 (work in progress),
June 2008.
[I-D.ietf-mmusic-media-path-middleboxes]
Stucker, B. and H. Tschofenig, "Analysis of Middlebox
Interactions for Signaling Protocol Communication along
the Media Path",
draft-ietf-mmusic-media-path-middleboxes-01 (work in
progress), July 2008.
[I-D.ietf-sipping-v6-transition]
Camarillo, G., "IPv6 Transition in the Session Initiation
Protocol (SIP)", draft-ietf-sipping-v6-transition-07 (work
in progress), August 2007.
[RFC3424] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 2002.
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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
Francois Audet
Nortel
4655 Great America Parkway
Santa Clara CA 95054
US
Email: audet@nortel.com
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