SIPPING Working Group C. Boulton
Internet-Draft Ubiquity Software Corporation
Expires: April 4, 2005 J. Rosenberg
Cisco Systems
October 2004
Best Current Practices for NAT Traversal for SIP
draft-ietf-sipping-nat-scenarios-02
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Copyright Notice
Copyright (C) The Internet Society (2004).
Abstract
Traversal of the Session Initiation Protocol (SIP) and the sessions
it establishes through Network Address Translators (NAT) is a complex
problem. Currently there are many deployment scenarios and traversal
mechanisms for media traffic. This document aims to provide concrete
recommendations and a unified method for NAT traversal as well as
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documenting corresponding call flows.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 3
3. Solution Technology Outline Description . . . . . . . . . . . 6
3.1 SIP Signaling . . . . . . . . . . . . . . . . . . . . . . 7
3.1.1 Symmetric Response . . . . . . . . . . . . . . . . . . 7
3.1.2 Connection Re-use . . . . . . . . . . . . . . . . . . 8
3.2 Media Traversal . . . . . . . . . . . . . . . . . . . . . 8
3.2.1 Symmetric RTP . . . . . . . . . . . . . . . . . . . . 8
3.2.2 STUN . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2.3 TURN . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2.4 ICE . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2.5 RTCP Attribute . . . . . . . . . . . . . . . . . . . . 10
3.2.6 Solution Profiles . . . . . . . . . . . . . . . . . . 10
4. NAT Traversal Scenarios . . . . . . . . . . . . . . . . . . . 11
4.1 Basic NAT SIP Signaling Traversal . . . . . . . . . . . . 11
4.1.1 Registration (Registrar/Proxy Co-Located . . . . . . . 11
4.1.2 Registration(Registrar/Proxy not Co-Located) . . . . . 15
4.1.3 Initiating a Session . . . . . . . . . . . . . . . . . 16
4.1.4 Receiving an Invitation to a Session . . . . . . . . . 18
4.2 Basic NAT Media Traversal . . . . . . . . . . . . . . . . 21
4.2.1 Port Restricted Cone NAT . . . . . . . . . . . . . . . 21
4.2.2 Symmetric NAT . . . . . . . . . . . . . . . . . . . . 33
4.3 Advanced NAT media Traversal Using ICE . . . . . . . . . . 36
4.3.1 Full Cone --> Full Cone traversal . . . . . . . . . . 37
4.3.2 Port Restricted Cone --> Port Restricted Cone
traversal . . . . . . . . . . . . . . . . . . . . . . 37
4.3.3 Internal TURN Server (Enterprise Deployment) . . . . . 37
4.4 Intercepting Intermediary (B2BUA) . . . . . . . . . . . . 37
4.5 IPV4/IPV6 . . . . . . . . . . . . . . . . . . . . . . . . 37
4.6 ICE with RTP/TCP . . . . . . . . . . . . . . . . . . . . . 37
5. References . . . . . . . . . . . . . . . . . . . . . . . . . . 37
5.1 Normative References . . . . . . . . . . . . . . . . . . . 37
5.2 Informative References . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 39
Intellectual Property and Copyright Statements . . . . . . . . 40
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1. Introduction
NAT (Network Address Translators) traversal has long been identified
as a large problem when considered in the context of the Session
Initiation Protocol (SIP)[1] and it's associated media such as Real
Time Protocol (RTP)[2]. The problem is further confused by the
variety of NATs that are available in the market place today and the
large number of potential deployment scenarios. Detail of different
NAT types can be found in RFC 3489bis [14].
The IETF has produced many specifications for the traversal of NAT,
including STUN, ICE, rport, symmetric RTP, TURN, connection reuse,
SDP attribute for RTCP, and others. These each represent a part of
the solution, but none of them gives the overall context for how the
NAT traversal problem is decomposed and solved through this
collection of specifications. This document serves to meet that
need.
This document attempts to provide a definitive set of 'Best Common
Practices' to demonstrate the traversal of SIP and it's associated
media through NAT devices. The document does not propose any new
functionality but does draw on existing solutions for both core SIP
signaling and media traversal (as defined in section 3).
The draft will be split into distinct sections as follows:
1. A clear definition of the problem statement
2. Description of proposed solutions for both SIP protocol signaling
and media signaling
3. A set of basic and advanced call flow scenarios
2. Problem Statement
The traversal of SIP through NAT can be split into two categories
that both require attention - The core SIP signaling and associated
media traversal.
The core SIP signaling has a number of issues when traversing through
NATs.
Firstly, the default operation for SIP response generation using
unreliable protocols such as the Unicast Datagram Protocol (UDP)
results in responses being generated at the User Agent Server (UAS)
being sent to the source address, as specified in either the SIP
'Via' header or the 'received' parameter (as defined in RFC 3261
[1]). The port is extracted from the SIP 'Via' header to complete
the IP address/port combination for returning the SIP response.
While the destination is correct, the port contained in the SIP 'Via'
header represents the listening port of the originating client and
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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 Network NAT Public Network
|
|
|
-------- SIP Request |open port 5650 --------
| |----------------------->--->-----------------------| |
| | | | |
| Client | |port 5060 SIP Response | Proxy |
| | x<------------------------| |
| | | | |
-------- | --------
|
|
|
Figure 1
Secondly, when using a reliable, connection orientated transport
protocol such as TCP, SIP has an inherent mechanism that results in
SIP responses reusing the connection that was created/used for the
corresponding transactional request. The SIP protocol does not
provide a mechanism that allows new requests generated in the
opposite direction (Previously occupying the role of UAS for the last
transaction) to use the existing TCP connection created between the
client and the server during registration. This results in the
registered contact address not being bound to the "connection" in the
case of TCP. Requests are then blocked at the NAT, as illustrated in
Figure 2. This problem also exists for unreliable transport
protocols such as UDP where external NAT mappings need to be re-used
to reach a SIP entity on the private side of the network.
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Private Network NAT Public Network
|
|
|
-------- (UAC 8023) REGISTER/Response (UAS 5060) --------
| |----------------------->---<-----------------------| |
| | | | |
| Client | |5060 INVITE (UAC 8015)| Proxy |
| | x<------------------------| |
| | | | |
-------- | --------
|
|
|
Figure 2
In Figure 2 the original REGISTER request is sent from the client on
port 8023 and received on port 5060, establishing a reliable
connection and opening a pin-hole in the NAT. The generation of a
new request from the proxy results in a request destined for the
registered entity (Contact IP address) which is not reachable from
the public network. This results in the new SIP request attempting
to create a connection to a private network address. This problem
would be solved if the original connection was re-used. While this
problem has been discussed in the context of connection orientated
protocols such as TCP, the problem exists for SIP signaling using any
transport protocol. The solution proposed for this problem in
section 3 of this document is relevant for all SIP signaling,
regardless of the transport protocol.
NAT policy can dictate that connections should be closed after a
period of inactivity. This period of inactivity can range
drastically from a number seconds to hours. Pure SIP signaling can
not be relied upon to keep alive connections for a number of reasons.
Firstly, SIP entities can sometimes have no signaling traffic for
long periods of time which has the potential to exceed he inactivity
timer, 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 resource.
Media associated with SIP calls also has problems traversing NAT.
RTP [2]] is on if the most common media transport type used in SIP
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signaling. Negotiation of RTP occurs with a SIP session
establishment using the Session Description Protocol(SDP) [3] and a
SIP offer/answer exchange[4]. 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 it's 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>=====================>RTP>======X | B |
| | X=====<RTP<=====================<RTP<=========<| |
-------- | | --------
| |
| |
| |
Figure 3
The connection address representing both clients are not available
on the public internet and traffic can be sent from both clients
through their NATs. The problem occurs when the traffic attempts to
traverse media through the foreign (not local) NAT. The connection
address extracted from the SDP payload is that of an internal
address, and so not resolvable from the public side of the NAT. To
complicate the problem further, a number of different NAT topologies
with different default behaviors increase the difficulty of proposing
a single solution.
3. Solution Technology Outline Description
When analyzing issues associated with traversal of SIP through
existing NAT, it has been identified that the problem can be split
into two clear solution areas as defined in section 2 of this
document. The traversal of the core protocol signaling and the
traversal of the associated media as specified in the Session
Description Payload (SDP) of a SIP offer/answer exchange[4]. The
following sub-sections outline solutions that enable core SIP
signaling and its associated media to traverse NATs.
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3.1 SIP Signaling
SIP signaling has two areas that result in transactional failure when
traversing through NAT, as described in section 2 of this document.
The remaining sub-sections describe appropriate solutions that result
in SIP signalling traversal through NAT, regardless of transport
protocol. IT is RECOMMEDED that SIP compliant entities follow the
guidelines presented in this section to enable traversal of SIP
signaling through NATs.
3.1.1 Symmetric Response
As described in section 2 of this document, when using an unreliable
transport protocol such as UDP, SIP responses are sent to the IP
address and port combination contained in the SIP 'Via' header field
(or default port for the appropriate transport protocol if not
present). This can result in responses being blocked at a NAT. In
such circumstances, SIP signaling requires a mechanism that will
allow entities to override the basic response generation mechanism in
RFC 3261 [1]. Once the SIP response is constructed, the destination
is still derived using the mechanisms described in RFC 3261 [1]. The
port (to which the response will be sent), however, will not equal
that specified in the SIP 'Via' header field but will be the port
from which the original request was sent. This results in the
pin-hole opened for the requests traversal of the NAT being reused,
in a similar manner to that of reliable connection orientated
transport protocols such as TCP. Figure 4 illustrates the response
traversal through the open pin hole using this method.
Private Network NAT Public Network
|
|
|
-------- | --------
| | | | |
| |send/receive | send/receive| |
| Client |port 5060-----<<->>-------------<<->>-----port 5060| Client |
| A | | | B |
| | | | |
-------- | --------
|
|
|
Figure 4
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The exact functionality for this method of response traversal is
called 'Symmetric Response' and the details are documented in RFC
3581 [5]. Additional requirements are imposed on SIP entities in
this specification such as listening and sending SIP
requests/responses from the same port.
3.1.2 Connection Re-use
The second problem with sip signaling, as defined in Section 3.1.2,
is to allow incoming requests to be properly routed. This is
addressed in [9], which allows the reuse of a TCP connection or UDP
5-tuple for incoming requests. That draft also provides keepalive
mechanisms based on using STUN to the SIP server. Usage of this
specification is RECOMMENDED. This mechanism is not transport
specific and should be used for any transport protocol.
Even if this draft is not used, clients SHOULD use the same IP
address and port (i.e., socket) for both transmission and receipt of
SIP messages. Doing so allows for the vast majority of industry
provided solutions to properly function.
3.2 Media Traversal
This document has already provided guidelines that recommend using
extensions to the core SIP protocol to enable traversal of NATs.
While ultimately not desirable, the additions are relatively straight
forward and provide a simple, universal solution for varying types of
NAT deployment. The issues of media traversal through NATs is not as
straight forward and requires the combination of a number of
traversal methodologies. The technologies outlined in the remainder
of this section provide the required solution set.
3.2.1 Symmetric RTP
The primary problem identified in section 2 of this document is that
internal IP address/port combinations can not be reached from the
public side of a NAT. In the case of media such as RTP, this will
result in no audio traversing a NAT(as illustrated in Figure 3). To
overcome this problem, a technique called 'Symmetric' RTP can be
used. This involves an SIP endpoint both sending and receiving RTP
traffic from the same IP Address/Port combination. This technique
also requires intelligence by a client on the public internet as it
identifies that incoming media for a particular session does not
match the information that was conveyed in the SDP. In this case
the client will ignore the SDP address/port combination and return
RTP to the IP address/port combination identified as the source of
the incoming media. This technique is known as 'Symmetric RTP' and
is documented in [12]. 'Symmetric RTP' SHOULD only be used for
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traversal of RTP through NAT when one of the participants in a media
session definitively knows that it is on the public network.
3.2.2 STUN
Simple Traversal of User Datagram Protocol(UDP) through Network
Address Translators(NAT) or STUN is defined in RFC 3489 [8]. It
provides a lightweight protocol that allows entities to probe and
discover the type of NAT that exist between itself and external
entities. It also provides details of the external IP address/port
combination used by the NAT device to represent the internal entity
on the public facing side of a NAT. On learning of such an external
representation, a client can use accordingly as the connection
address in SDP to provide NAT traversal. STUN only works with Full
Cone, Restricted Cone and Port Restricted Cone type NATs. STUN does
not work with Symmetric NATs as the technique used to probe for the
external IP address port representation using a STUN server will
provide a different result to that required for traversal by an
alternative SIP entity. The IP address/port combination deduced for
the STUN server would be blocked for incoming packets from an
alterative SIP entity.
3.2.3 TURN
As mentioned in the previous section, the STUN protocol does not work
for UDP traversal through a Symmetric style NAT. Traversal Using
Relay NAT (TURN) provides the solution for UDP traversal of symmetric
NAT. TURN is extremely similar to STUN in both syntax and operation.
It provides an external address at a TURN server that will act as a
relay and guarantee traffic will reach the associated internal
address. The full details of the TURN specification are defined in
[11]. A TURN service will almost always provide media traffic to a
SIP entity but it is RECOMMENDED that this method only be used as a
last resort and not as a general mechanism for NAT traversal. This
is because using TURN has high performance costs when relaying media
traffic.
3.2.4 ICE
Interactive Connectivity Establishment (ICE) is the RECOMMENDED
method for traversal of existing NAT if Symmetric RTP is not
appropriate. ICE is a methodology for using existing technologies
such as STUN, TURN and any other UNSAF[7] compliant protocol to
provide a unified solution. This is achieved by obtaining as many
representative IP address/port combinations as possible using
technologies such as STUN/TURN etc. Once the addresses are
accumulated, they are all included in the SDP exchange in a new media
attribute called 'candidate'. Each 'candidate' SDP attribute entry
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has detailed connection information including a media addresses
(including optional RTCP information), priority, username, password
and a unique session ID. The appropriate IP address/port
combinations are used in the correct order depending on the specified
priority. A client compliant to the ICE specification will then
locally run instances of STUN servers on all addresses being
advertised using ICE. Each instance will undertake connectivity
checks to ensure that a client can successfully receive media on the
advertised address. Only connections that pass the relevant
connectivity checks are used for media exchange. The full details of
the ICE methodology are contained in [13].
3.2.5 RTCP Attribute
Normal practice when selecting a port for defining Real Time Control
Protocol(RTCP) [2] is for consecutive order numbering (i.e select an
incremented port for RTCP from that used for RTP). This assumption
causes RTCP traffic to break when traversing many NATs due to blocked
ports. To combat this problem a specific address and port need to be
specified in the SDP rather than relying on such assumptions. RFC
3605 [5] defines an SDP attribute that is included to explicitly
specify transport connection information for RTCP. The address
details can be obtained using any appropriate method including those
detailed previously in this section (e.g. STUN, TURN).
3.2.6 Solution Profiles
This draft has documented a number of technology solutions for the
traversal of media through differing NAT deployments. A number of
'profiles' will now be defined that categorize varying levels of
support for the technologies described.
3.2.6.1 Primary Profile
A client falling into the 'Primary' profile supports ICE in
conjunction with STUN, TURN and RFC 3605 [5] for RTCP. ICE is used
in all cases and falls back to standard operation when dealing with
non-ICE clients. A client which falls into the 'Primary' profile
will be maximally interoperable and function in a rich variety of
environments including enterprise, consumer and behind all variety of
NAT.
3.2.6.2 Consumer Profile
A client falling into the 'Consumer' profile supports STUN and RFC
3605 [5] for RTCP. It uses STUN to allocate bindings, and can also
detect when it is in the unfortunate situation of being behind a
'Symmetric' NAT, although it simply cannot function in this case.
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These clients will only work in deployment situations where the
access is sufficiently controlled to know definitively that there
won't be Symmetric NAT. This is hard to guarantee as users can
always pick up their client and connect via a different access
network.
3.2.6.3 Minimal Profile
A client falling into the 'Minimal' profile will send/receive RTP
form the same IP/port combination. This client requires proprietary
network based solutions to function in any NAT traversal scenario.
All clients SHOULD support the 'Primary Profile', MUST support the
'Minimal Profile' and MAY support the 'Consumer Profile'.
4. NAT Traversal Scenarios
This section of the document includes detailed NAT traversal
scenarios for both SIP signaling and the associated media.
4.1 Basic NAT SIP Signaling Traversal
The following sub-sections concentrate on SIP signaling traversal of
NAT. The scenarios include traversal for both reliable and
un-reliable transport protocols.
[Editors Note: The scenarios are still in early construction and a
couple have been included as a hint of direction - All comments
welcome for next release]
4.1.1 Registration (Registrar/Proxy Co-Located
The set of scenarios in this section document basic signaling
traversal of a SIP REGISTER method through a NAT.
4.1.1.1 UDP
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Client NAT Proxy
| | |
|(1) REGISTER | |
|----------------->| |
| |(1) REGISTER |
| |----------------->|
| |(2) 401 Unauth |
| |<-----------------|
|(2) 401 Unauth | |
|<-----------------| |
|(3) REGISTER | |
|----------------->| |
| |(3) REGISTER |
| |----------------->|
|*************************************|
| Create Connection Re-use Tuple |
|*************************************|
| |(4) 200 OK |
| |<-----------------|
|(4) 200 OK | |
|<-----------------| |
| | |
Figure 5.
Figure 5
In this example the client sends a SIP REGISTER request through a NAT
which is challenged using the Digest authentication scheme. The
client will include an 'rport' parameter as described in section
3.1.1 of this document for allowing traversal of UDP responses. The
original request as illustrated in (1) in Figure 5 is a standard
REGISTER message:
REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK
Max-Forwards: 70
Supported: gruu
From: Client <sip:client@example.com>;tag=djks8732
To: Client <sip:client@example.com>
Call-ID: 763hdc73y7dkb37@example.com
CSeq: 1 REGISTER
Contact: <sip:client@client.example.com>; connectioId=1
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000A95A0E120>"
Content-Length: 0
This proxy now generates a SIP 401 response to challenge for
authentication, as depicted in (2) from Figure 5.:
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SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP client.example.com:5060;rport=8050;branch=z9hG4bK;received=192.0.1.2
From: Client <sip:client@example.com>;tag=djks8732
To: Client <sip:client@example.com>;tag=876877
Call-ID: 763hdc73y7dkb37@example.com
CSeq: 1 REGISTER
WWW-Authenticate: [not shown]
Content-Length: 0
The response will be sent to the address appearing in the 'received'
parameter of the SIP 'Via' header (address 192.0.1.2). The response
will not be sent to the port deduced from the SIP 'Via' header, as
per standard SIP operation but will be sent to the value that has
been stamped in the 'rport' parameter of the SIP 'Via' header (port
8050). For the response to successfully traverse the NAT, all of the
conventions defined in RFC 3581 [5] MUST be obeyed. Make note of the
both the 'connectionID' and 'sip.instance' contact header parameters.
They are used to establish a connection re-use tuple as defined in
[9]. The connection tuple creation is clearly shown in Figure 5.
This ensures that any inbound request that causes a registration
lookup will result in the re-use of the connection path established
by the registration. This exonerates the need to manipulate contact
header URI's to represent a globally routable address as perceived on
the public side of a NAT. The subsequent messages defined in (3) and
(4) from Figure 5 use the same mechanics for NAT traversal.
[Editors note: Will provide more details on heartbeat mechanism in
next revision]
[Editors note: Can complete full flows if required on heartbeat
inclusion]
4.1.1.2 Reliable Transport
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Client NAT Registrar
| | |
|(1) REGISTER | |
|----------------->| |
| |(1) REGISTER |
| |----------------->|
| |(2) 401 Unauth |
| |<-----------------|
|(2) 401 Unauth | |
|<-----------------| |
|(3) REGISTER | |
|----------------->| |
| |(3) REGISTER |
| |----------------->|
|*************************************|
| Create Connection Re-use Tuple |
|*************************************|
| |(4) 200 OK |
| |<-----------------|
|(4) 200 OK | |
|<-----------------| |
| | |
Figure 6.
Traversal of SIP REGISTER messages request/responses using a
reliable, connection orientated protocol such as TCP does not require
any additional core SIP signaling extensions. SIP responses will
re-use the connection created for the initial REGISTER request, (1)
from Figure 6:
REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
Max-Forwards: 70
Supported: gruu
From: Client <sip:client@example.com>;tag=djks809834
To: Client <sip:client@example.com>
Call-ID: 763hdc783hcnam73@example.com
CSeq: 1 REGISTER
Contact: <sip:client@client.example.com;transport=tcp>; connectioId=1
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000A95A0E121>"
Content-Length: 0
This example was included to show the inclusion of the of the
connection re-use Contact header parameters as defined in the
Connection Re-use draft [9]. This creates an association tuple as
described in the previous example for future inbound requests
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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.
[Editors note: Will provide more details on heartbeat mechanism in
next revision]
[Editors note: Can complete full flows on inclusion of heartbeat
mechanism]
4.1.2 Registration(Registrar/Proxy not Co-Located)
This section demonstrates traversal mechanisms when the Registrar
component is not co-located with the edge proxy element. The
procedures described in this section are identical, regardless of
transport protocol and so only one example will be documented in the
form of TCP.
Client NAT Proxy Registrar
| | | |
|(1) REGISTER | | |
|----------------->| | |
| |(1) REGISTER | |
| |----------------->| |
| | |(2) REGISTER |
| | |----------------->|
| | |(3) 401 Unauth |
| | |<-----------------|
| |(4) 401 Unauth | |
| |<-----------------| |
|(4)401 Unauth | | |
|<-----------------| | |
|(5)REGISTER | | |
|----------------->| | |
| |(5)REGISTER | |
| |----------------->| |
| | |(6)REGISTER |
| | |----------------->|
| | |(7)200 OK |
| | |<-----------------|
|********************************************************|
| Create Connection Re-use Tuple |
|********************************************************|
| |(8)200 OK | |
| |<-----------------| |
|(8)200 OK | | |
|<-----------------| | |
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| | | |
Figure 7.
This scenario builds on that contained in section 4.1.1.2. This time
the REGISTER request is routed onwards to a separated Registrar. The
important message to note is (5) in Figure 7. At this point, the
proxy server routes the SIP REGISTER message to the Registrar. The
proxy will create the connection re-use tuple at the same moment as
the co-located example but for subsequent messages to arrive at the
Proxy, the element needs to request to stay in the signaling path.
REGISTER message (5) contains a SIP PATH extension header, as defined
in RFC 3327 [6]. REGISTER message (5) would look as follows:
REGISTER sip:registrar.example.com SIP/2.0
Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4njkca8398hadjaa
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
Max-Forwards: 70
Supported: gruu
From: Client <sip:client@example.com>;tag=djks809834
To: Client <sip:client@example.com>
Call-ID: 763hdc783hcnam73@example.com
CSeq: 1 REGISTER
Path: <sip:sip%3Aclient%40example.com@proxy.example.com;lr>
Contact: <sip:client@client.example.com;transport=tcp>; connectioId=1
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000A95A0E121>"
Content-Length: 0
This results in the path header being stored along with the AOR and
it's associated binding at the Registrar. The URI contained in the
PATH will be inserted as a pre-loaded SIP 'Route' header into any
request that arrives at the Registrar and is directed towards the
associated binding. This guarantees that all requests for the new
Registration will be forwarded to the edge proxy. The user part of
the SIP 'Path' header URI that was inserted by the edge proxy
contains an escaped form of the original AOR that was contained in
the REGISTER request. On receiving subsequent requests, the edge
proxy will examine the user part of the pre-loaded SIP 'route' header
and extract the original AOR for use in it's connection tuple
comparison, as defined in the connection re-use draft [9]. An
example which will build on this scenario (showing an inbound request
to the AOR) is detailed in section 4.1.4.2 of this document.
4.1.3 Initiating a Session
This section covers basic SIP signaling when initiating a call from
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behind a NAT.
4.1.3.1 UDP
Initiating a call using UDP.
Client NAT Proxy [..]
| | |
|(1) INVITE | | |
|----------------->| | |
| |(1) INVITE | |
| |----------------->| |
| |(2) 407 Unauth | |
| |<-----------------| |
|(2) 407 Unauth | | |
|<-----------------| | |
|(3) INVITE | | |
| |(3) INVITE | |
| |----------------->| |
| | |(4) INVITE |
| | |---------------->|
| | |(5)180 RINGING |
| | |<----------------|
| |(6)180 RINGING | |
| |<-----------------| |
|(6)180 RINGING | | |
|<-----------------| | |
| | |(7)200 OK |
| | |<----------------|
| |(8)200 OK | |
| |<-----------------| |
|(8)200 OK | | |
|<-----------------| | |
|(9)ACK | | |
|----------------->| | |
| |(9)ACK | |
| |----------------->| |
| | |(10) ACK |
| | |---------------->|
| | |(11) |
Figure 8.
The initiating client generates an INVITE request that is to be sent
through the NAT to a Proxy server. The INVITE message is represented
in Figure 8 by (1) and is as follows:
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INVITE sip:clientB@example.com SIP/2.0
Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK74husdHG
Max-Forwards: 70
Route: <sip:proxy.example.com;lr>
From: clientA <sip:clientA@example.com>;tag=7skjdf38l
To: clientB <sip:clientB@example.com>
Call-ID: 8327468763423@example.com
CSeq: 1 INVITE
Contact: <sip:im_a_gruu@proxy.example.com>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
There are a number of points to note with this message:
1. Firstly, as with the registration example in section 4.1.1.1,
reponses to this request will not automatically pass back through
a NAT and so the SIP 'Via' header 'rport' is included as
described in the 'Symmetric response' section(3.1.1) and defined
in RFC 3581 [5].
2. Secondly, the contact inserted contains the GRUU previously
obtained from the registration.
3. [Editors Note: TODO - Expand description of GRUU and connection
re-use]
4.1.3.2 Reliable Transport
[Editors note: TODO]
4.1.4 Receiving an Invitation to a Session
This section details scenarios where a client behind a NAT receives
an inbound request through the NAT. These scenarios build on the
previous registration scenario from sections 4.1.1 and 4.1.2 in this
document.
4.1.4.1 Registrar/Proxy Co-located
The core SIP signaling associated with this call flow is not impacted
directly by the transport protocol and so only one example scenario
is necessary. The example uses UDP and follows on from the
registration installed in the example from section 4.1.1.1.
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Client NAT Registrar/Proxy SIP Entity
| | | |
|*******************************************************|
| Registration Binding Installed in |
| section 4.1.1.1 |
|*******************************************************|
| | | |
| | |(1)INVITE |
| | |<----------------|
| |(2)INVITE | |
| |<-----------------| |
|(2)INVITE | | |
|<-----------------| | |
| | | |
| | | |
Figure 9.
The core SIP signaling associated with this call flow is not impacted
directly by the transport protocol and so only one example scenario
is necessary. The example uses UDP and follows on from the
registration installed in section 4.1.1.1. An INVITE request arrives
at the Registrar with a destination pointing to the AOR of that
inserted in section 4.1.1.1. The message is illustrated by (1) in
Figure 9 and looks as follows:
INVITE sip:client@example.com SIP/2.0
Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: External <sip:External@external.example.com>;tag=7893hd
To: client <sip:client@example.com>
Call-ID: 8793478934897@external.example.com
CSeq: 1 INVITE
Contact: <sip:external@192.0.1.4>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
The INVITE matches the registration binding at the Registrar and the
INVITE request-URI is re-written to the selected onward address. The
proxy then examines the request URI of the INVITE and compares with
it's list of current open connections/mappings. It uses the incoming
AOR to commence the check for associated open connections/mappings.
Once matched, the proxy checks to see if the unique instance
identifier (+sip.instance)associated with the binding equals the same
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instance identifier associated with the binding. If more than one
results are matched, the lowest 'connectionID' Contact parameter will
be used. This is message (2) from Figure 9 and is as follows:
INVITE sip:sip:client@client.example.com SIP/2.0
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4kmlds893jhsd
Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: External <sip:External@external.example.com>;tag=7893hd
To: client <sip:client@example.com>
Call-ID: 8793478934897@external.example.com
CSeq: 1 INVITE
Contact: <sip:external@192.0.1.4>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
It is a standard SIP INVITE request with no additional functionality.
The major difference being that this request will not follow the
address specified in the Request-URI, as standard SIP rules would
enforce but will be sent on the connection/mapping associated with
the registration binding. This then allows the original
connection/mapping from the initial registration process to be
re-used.
4.1.4.2 Registrar/Proxy Not Co-located
Client NAT Proxy Registrar SIP Entity
| | | | |
|***********************************************************|
| Registration Binding Installed in |
| section 4.1.2 |
|***********************************************************|
| | | |(1)INVITE |
| | | |<-------------|
| | |(2)INVITE | |
| | |<-------------| |
| |(3)INVITE | | |
| |<-------------| | |
|(3)INVITE | | | |
|<-------------| | | |
| | | | |
| | | | |
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Figure 9.
4.2 Basic NAT Media Traversal
This section provides example scenarios to demonstrate basic media
traversal using the techniques outlined earlier in this document.
4.2.1 Port Restricted Cone NAT
This section demonstrates an example of a client both initiating and
receiving calls behind a 'Restricted Cone' NAT. The examples have
been included to represent both 'Restricted' and 'Port Restricted'
NAT media traversal. An example is included for both STUN and ICE
with ICE being the RECOMENDED method.
4.2.1.1 STUN Solution
It is possible to traverse media through a 'Restricted Cone NAT'
using STUN.
4.2.1.1.1 Initiating Session
The following example demonstrates media traversal through a
'Restricted Cone' NAT using STUN. It is assumed in this example that
the STUN client and SIP Client are co-located on the same machine.
Note that some SIP signalling messages have been left out for
simplicity.
Client NAT STUN [..]
Server
| | | |
|(1) STUN Req | | |
|src=10.0.1.1:5301 | | |
|----------------->| | |
| |(2) STUN Req | |
| |src=1.2.3.4:5601 | |
| |----------------->| |
| |(3) STUN Resp | |
| |<-----------------| |
| |map=1.2.3.4:5601 | |
| |dest=1.2.3.4:5601 | |
|(4) STUN Resp | | |
|<-----------------| | |
|map=1.2.3.4:5601 | | |
|dest=10.0.1.1:5301| | |
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|(5) STUN Req | | |
|src=10.0.1.1:5302 | | |
|----------------->| | |
| |(6) STUN Req | |
| |src=1.2.3.4:5608 | |
| |----------------->| |
| |(7) STUN Resp | |
| |<-----------------| |
| |map=1.2.3.4:5608 | |
| |dest=1.2.3.4:5608 | |
|(8) STUN Resp | | |
|<-----------------| | |
|map=1.2.3.4:5608 | | |
|dest=10.0.1.1:5302| | |
|(9)SIP INVITE | | |
|----------------->| | |
| |(10)SIP INVITE | |
| |------------------------------------>|
| | |(11)SIP 200 OK |
| |<------------------------------------|
|(12)SIP 200 OK | | |
|<-----------------| | |
|========================================================|
|>>>>>>>>>>>Outgoing Media sent to 1.2.3.4:5601>>>>>>>>>>|
|========================================================|
|========================================================|
|<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<|
|========================================================|
|(13)SIP ACK | | |
|----------------->| | |
| |(14) SIP ACK | |
| |------------------------------------>|
| | | |
Figure 18: Restricted NAT with STUN - Initiating
o On deciding to initiate a SIP voice session the VOIP client starts
a local STUN client. The STUN client generates a standard STUN
request as indicated in (1) from Figure 18 which also highlights
the source address and port for which the client device wishes to
obtain a mapping. The STUN request is sent through the NAT
towards the public internet.
o STUN message (2) traverses the NAT and breaks out onto the public
internet towards the public STUN server. Note that the source
address of the STUN requests now represents the public address and
port from the public side of the NAT.
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o
o The STUN server receives the request and processes it
appropriately. This results in a successful STUN response being
generated and returned (3). The message contains details of the
mapped public address (contained in the STUN MAPPED-ADDRESS
attribute) which is to be used by the originating client to
receive media (see 'map=' from (3)).
o The STUN response traverses back through the NAT using the binding
created by the STUN request and presents the new mapped address to
the client (4). At this point the process is repeated to obtain a
second mapped address (as shown in (5)-(8)) for an alternative
local address (local port has now changed from 5301 to 5302 in
(5)).
o The client now constructs a SIP INVITE message(9). Note that
traversal of SIP is not covered in this example and is discussed
in earlier sections of the document. The INVITE request will use
the addresses it has obtained in the previous STUN transactions to
populate the SDP of the SIP INVITE as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 10.0.1.1
c=IN IP4 1.2.3.4
t=0 0
m=audio 5601 RTP/AVP 0
a=rtcp:5608
o Note that the mapped address obtained from the STUN transactions
are inserted as the connection address for the SDP (c=1.2.3.4).
The Primary port for RTP is also inserted in the SDP (m=audio 5601
RTP/AVP 0). Finally, the port gained from the additional STUN
binding is placed in the RTCP attribute (as discussed in
Section 3.2.5)for traversal of RTCP (a=rtcp:5608).
o The SIP signalling then traverses the NAT and sets up the SIP
session (10-12). Note that the client transmits media as soon as
the 200 OK to the INVITE arrives at the client(12). Up until this
point the incoming media will not pass through the NAT as no
outbound association has been created with the far end client.
Two way media communication has now been established.
4.2.1.1.2 Receiving Session Invitation
Receiving a session for a 'Restricted Cone' NAT using STUN is very
similar to the example outlined in Section 4.2.1.1.1. Figure 20
illustrates the associated flow of messages.
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Client NAT STUN [..]
Server
| | | (1)SIP INVITE |
| |<-----------------|------------------|
|(2) SIP INVITE | | |
|<-----------------| | |
| | | |
|(3) STUN Req | | |
|src=10.0.1.1:5301 | | |
|----------------->| | |
| |(4) STUN Req | |
| |src=1.2.3.4:5601 | |
| |----------------->| |
| |(5) STUN Resp | |
| |<-----------------| |
| |map=1.2.3.4:5601 | |
| |dest=1.2.3.4:5601 | |
|(6) STUN Resp | | |
|<-----------------| | |
|map=1.2.3.4:5601 | | |
|dest=10.0.1.1:5301| | |
|(7) STUN Req | | |
|src=10.0.1.1:5302 | | |
|----------------->| | |
| |(8) STUN Req | |
| |src=1.2.3.4:5608 | |
| |----------------->| |
| |(9) STUN Resp | |
| |<-----------------| |
| |map=1.2.3.4:5608 | |
| |dest=1.2.3.4:5608 | |
|(10) STUN Resp | | |
|<-----------------| | |
|map=1.2.3.4:5608 | | |
|dest=10.0.1.1:5302| | |
|(11)SIP 200 OK | | |
|----------------->| | |
| |(12)SIP 200 OK | |
| |------------------------------------>|
|========================================================|
|>>>>>>>>>>>Outgoing Media sent to 1.2.3.4:5601>>>>>>>>>>|
|========================================================|
|========================================================|
|<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<|
|========================================================|
| | |(13)SIP ACK |
| |<------------------------------------|
|(14)SIP ACK | | |
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|<-----------------| | |
| | | |
Figure 20: Restricted NAT with STUN - Receiving
o On receiving an invitation to a SIP voice session the VOIP client
starts a local STUN client. The STUN client generates a standard
STUN request as indicated in (3) from Figure 20 which also
highlights the source address and port for which the client device
wishes to obtain a mapping. The STUN request is sent through the
NAT towards the public internet.
o STUN message (4) traverses the NAT and breaks out onto the public
internet towards the public STUN server. Note that the source
address of the STUN requests now represents the public address and
port from the public side of the NAT.
o
o The STUN server receives the request and processes appropriately.
This results in a successful STUN response being generated and
returned (5). The message contains details of the mapped public
address (contained in the STUN MAPPED-ADDRESS attribute) which is
to be used by the originating client to receive media (see 'map='
from (5)).
o The STUN response traverses back through the NAT using the binding
created by the STUN request and presents the new mapped address to
the client (6). At this point the process is repeated to obtain a
second mapped address (as shown in (7)-(10)) for an alternative
local address (local port has now changed from 5301 to 5302 in
(7)).
o The client now constructs a SIP 200 OK message(11). Note that
traversal of SIP is not covered in this example and is discussed
in earlier sections of the document. The 200 OK response will use
the addresses it has obtained in the previous STUN transactions to
populate the SDP of the SIP INVITE as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 10.0.1.1
c=IN IP4 1.2.3.4
t=0 0
m=audio 5601 RTP/AVP 0
a=rtcp:5608
o Note that the mapped address obtained from the initial STUN
transaction is inserted as the connection address for the SDP
(c=1.2.3.4). The Primary port for RTP is also inserted in the SDP
(m=audio 5601 RTP/AVP 0). Finally, the port gained from the
additional binding is placed in the RTCP attribute (as discussed
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in Section 3.2.5)for traversal of RTCP (a=rtcp:5608).
o The SIP signalling then traverses the NAT and sets up the SIP
session (11-14). Note that the client transmits media as soon as
the 200 OK to the INVITE is sent to the UAC(11). Up until this
point the incoming media will not pass through the NAT as no
outbound association has been created with the far end client.
Two way media communication has now been established.
4.2.1.2 ICE Solution
The preferred solution for media traversal of NAT is using ICE, as
described in Section 3.2.4. The following examples illustrate the
traversal of a 'Port Restricted Cone' NAT for both an initiating and
receiving client. The example only covers ICE in association with
STUN and TURN.
4.2.1.2.1 Initiating Session
The following example demonstrates an initiating traversal through a
'Restricted Cone' NAT using ICE.
Client NAT STUN TURN [..]
Server Server
| | | | |
|(1) STUN Req | | | |
|src=10.0.1.1:5301 | | | |
|----------------->| | | |
| |(2) STUN Req | | |
| |src=1.2.3.4:5601 | | |
| |----------------->| | |
| |(3) STUN Resp | | |
| |<-----------------| | |
| |map=1.2.3.4:5601 | | |
| |dest=1.2.3.4:5601 | | |
|(4) STUN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5601 | | | |
|dest=10.0.1.1:5301| | | |
|(5) STUN Req | | | |
|src=10.0.1.1:5311 | | | |
|----------------->| | | |
| |(6) STUN Req | | |
| |src=1.2.3.4:5611 | | |
| |----------------->| | |
| |(7) STUN Resp | | |
| |<-----------------| | |
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| |map=1.2.3.4:5611 | | |
| |dest=1.2.3.4:5611 | | |
|(8) STUN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5611 | | | |
|dest=10.0.1.1:5311| | | |
|(9) TURN Allocate | | | |
|src=10.0.1.1:5302 | | | |
|----------------->| | | |
| |(10) TURN Allocate| | |
| |src=1.2.3.4:5608 | | |
| |------------------------------------>| |
| |(11) TURN Resp | | |
| |<------------------------------------| |
| |map=1.2.3.4:5608 | | |
| |dest=1.2.3.4:5608 | | |
|(12) TURN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5608 | | | |
|dest=10.0.1.1:5302| | | |
|(13) TURN Allocate| | | |
|src=10.0.1.1:5312 | | | |
|----------------->| | | |
| |(14) TURN Allocate| | |
| |src=1.2.3.4:5618 | | |
| |------------------------------------>| |
| |(15) TURN Resp | | |
| |<------------------------------------| |
| |map=1.2.3.4:5618 | | |
| |dest=1.2.3.4:5618 | | |
|(16) TURN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5618 | | | |
|dest=10.0.1.1:5312| | | |
|(17)SIP INVITE | | | |
|----------------->| | | |
| |(18)SIP INVITE | | |
| |-------------------------------------------->|
| | |(19)SIP 200 OK | |
| |<--------------------------------------------|
|(20)SIP 200 OK | | | |
|<-----------------| | | |
|(21)STUN Req | | | |
|----------------->| | | |
| |(22) STUN Req | | |
| |-------------------------------------------->|
| | |(23)STUN Resp | |
| |<--------------------------------------------|
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|(24)STUN Resp | | | |
|<-----------------| | | |
|================================================================|
|>>>>>>>>>>>Outgoing Media sent from 10.1.1.1:5301>>>>>>>>>>>>>>>|
|================================================================|
| | |(25) STUN Req | |
| |<--------------------------------------------|
|(26)STUN Req | | | |
|<-----------------| | | |
|(27)STUN Resp | | | |
|----------------->| | | |
| | |(28)STUN Resp | |
| |-------------------------------------------->|
|================================================================|
|<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<<<<<<<<<|
|================================================================|
|(29)SIP ACK | | | |
|----------------->| | | |
| |(30) SIP ACK | | |
| |-------------------------------------------->|
| | | | |
Figure 22: Restricted NAT with ICE - Initiating
o On deciding to initiate a SIP voice session the VOIP client starts
a local STUN and TURN client. The STUN client generates a
standard STUN request as indicated in (1) from Figure 22 which
also highlights the source address and port for which the client
device wishes to obtain a mapping. The STUN request is sent
through the NAT towards the public internet.
o STUN message (2) traverses the NAT and breaks out onto the public
internet towards the public STUN server. Note that the source
address of the STUN requests now represents the public address and
port from the public side of the NAT.
o
o The STUN server receives the request and processes appropriately.
This results in a successful STUN response being generated and
returned (3). The message contains details of the mapped public
address (contained in the STUN MAPPED-ADDRESS attribute) which is
to be used by the originating client to receive media (see 'map=')
from (3)).
o The STUN response traverses back through the NAT using the binding
created by the STUN request and presents the new mapped address to
the client (4). The process is repeated and a second STUN derived
address is obtained, as illustrated in (5)-(8) in Figure 22.
While the STUN client is obtaining addresses', the TURN client
will also be attempting to obtain external representations. The
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TURN Allocate message is constructed in association with the local
IP address and port combination(9). The TURN Allocate message is
then sent from the client to the external TURN server via the
NAT(10). The TURN server processes the Allocate request and
returns an appropriate response(11). The response contains the
'Mapped-Address'(defined in STUN specification) attribute which
contains the external representation that the TURN server will
provide for the internal mapping. The TURN response then
traverses back through NAT and returns the newly allocated
external representation to the originating client(12).The process
is repeated and a second TURN derived address is obtained, as
illustrated in (13)-(16) in Figure 22. At this point the client
behind the NAT has a pair of STUN external representations and
TURN equivalents. The client would be free to gather any number
of external representations using any UNSAF[7] compliant protocol.
o The client now constructs a SIP INVITE message(17). The INVITE
request will use the addresses it has obtained in the previous
STUN/TURN interactions to populate the SDP of the SIP INVITE.
This should be carried out in accordance with the semantics
defined in the ICE specification[13], as shown below (*note - /*
signifies line continuation):
v=0
o=test 2890844526 2890842807 IN IP4 10.0.1.1
c=IN IP4 1.2.3.4
t=0 0
m=audio 5601 RTP/AVP 0
a=candidate:H83jksd 1.0 rtp_uname_frag_1 rtp_pass_1 1.2.3.4 5601
/* rtcp_uname_frag_1 rtcp_pass_1 1.2.3.4 5611
a=candidate:Hye73hd 0.8 rtp_uname_frag_2 rtp_pass_2 1.2.3.4 5608
/* rtcp_uname_frag_2 rtcp_pass_2 1.2.3.4 5618
a=candidate:H82hjjh 0.5 rtp_uname_frag_3 rtp_pass_3 1.2.3.4 5600
o The SDP has been constructed to include all the available
addresses that have been assembled. The first 'candidate' address
contains the two STUN derived addresses for both RTP and RTCP
traffic. This entry has been given the highest priority(1.0) by
the client and also inserted as the default address.
o The second 'candidate' address contains the two TURN derived
addresses for both RTP and RTCP traffic. This entry has been
given the second highest priority(0.8).
o The third and final 'candidate' address contains a local interface
address that has not been derived externally. This entry has been
given the lowest priority(0.5).
o The SIP signalling then traverses the NAT and sets up the SIP
session (18)-(20). On advertising a candidate address, the client
should have a local STUN server running on each advertised
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candidate address. This is for the purpose of responding to
incoming connectivity checks. In this example, after sending the
INVITE and receiving a 200 OK, the client initiates an outgoing
STUN connectivity check to the selected remote interfaces
(21)-(24) (*Note - this process will be repeated for every
advertised address which is not shown in the diagram for
simplicity). On receiving a STUN response, the client is able to
stream media to the remote destination(*Note - if further STUN
connectivity responses are received after the client has started
streaming media with a higher priority, it will be used instead).
The remote destination will also carry out similar STUN
connectivity checks (25)-(28) which then allows media to be
streamed to the client behind the NAT using the advertised
connections. Two way audio is now possible between the two
clients.
4.2.1.2.2 Receiving Session Invitation
This example is similar to that described in Section 4.2.1.2.1. The
client behind a NAT is receiving the incoming ICE Initiate in a SIP
INVITE request.
Client NAT STUN TURN [..]
Server Server
| | | | |
| | |(1)SIP INVITE | |
| |<--------------------------------------------|
|(2)SIP INVITE | | | |
|<-----------------| | | |
| | | | |
|(3) STUN Req | | | |
|src=10.0.1.1:5301 | | | |
|----------------->| | | |
| |(4) STUN Req | | |
| |src=1.2.3.4:5601 | | |
| |----------------->| | |
| |(5) STUN Resp | | |
| |<-----------------| | |
| |map=1.2.3.4:5601 | | |
| |dest=1.2.3.4:5601 | | |
|(6) STUN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5601 | | | |
|dest=10.0.1.1:5301| | | |
|(7) STUN Req | | | |
|src=10.0.1.1:5311 | | | |
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|----------------->| | | |
| |(8) STUN Req | | |
| |src=1.2.3.4:5611 | | |
| |----------------->| | |
| |(9) STUN Resp | | |
| |<-----------------| | |
| |map=1.2.3.4:5611 | | |
| |dest=1.2.3.4:5611 | | |
|(10) STUN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5611 | | | |
|dest=10.0.1.1:5311| | | |
|(11) TURN Allocate| | | |
|src=10.0.1.1:5302 | | | |
|----------------->| | | |
| |(12) TURN Allocate| | |
| |src=1.2.3.4:5608 | | |
| |------------------------------------>| |
| |(13) TURN Resp | | |
| |<------------------------------------| |
| |map=1.2.3.4:5608 | | |
| |dest=1.2.3.4:5608 | | |
|(14) TURN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5608 | | | |
|dest=10.0.1.1:5302| | | |
|(15) TURN Allocate| | | |
|src=10.0.1.1:5312 | | | |
|----------------->| | | |
| |(16) TURN Allocate| | |
| |src=1.2.3.4:5618 | | |
| |------------------------------------>| |
| |(17) TURN Resp | | |
| |<------------------------------------| |
| |map=1.2.3.4:5618 | | |
| |dest=1.2.3.4:5618 | | |
|(18) TURN Resp | | | |
|<-----------------| | | |
|map=1.2.3.4:5618 | | | |
|dest=10.0.1.1:5312| | | |
|(19)SIP 200 OK | | | |
|----------------->| | | |
| |(20)SIP 200 OK | | |
| |-------------------------------------------->|
|(21)STUN Req | | | |
|----------------->| | | |
| |(22) STUN Req | | |
| |-------------------------------------------->|
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| | |(23)STUN Resp | |
| |<--------------------------------------------|
|(24)STUN Resp | | | |
|<-----------------| | | |
|================================================================|
|>>>>>>>>>>>Outgoing Media sent from 10.1.1.1:5301>>>>>>>>>>>>>>>|
|================================================================|
| | |(25) STUN Req | |
| |<--------------------------------------------|
|(26)STUN Req | | | |
|<-----------------| | | |
|(27)STUN Resp | | | |
|----------------->| | | |
| | |(28)STUN Resp | |
| |-------------------------------------------->|
|================================================================|
|<<<<<<<<<<<Incoming Media sent to 1.2.3.4:5601<<<<<<<<<<<<<<<<<<|
|================================================================|
| | |(29)SIP ACK | |
| |<--------------------------------------------|
|(30)SIP ACK | | | |
|<-----------------| | | |
| | | | |
Figure 24: Restricted NAT with ICE - Receiving
o As mentioned previously, this example is similar that described in
Section 4.2.1.2.1. For this reason, some of the description may
reference the previous example. The scenario starts with the
client behind the NAT receiving a SIP INVITE(1) request(ICE
initiate message).
o On receiving the SIP INVITE the client is able to collect all
possible addresses available for media interaction (e.g. Local
addresses, STUN derived, TURN derived). See detail from
Section 4.2.1.2.1 for explanation on accumulating all possible
media addresses (Steps (3)-(18) in Figure 24).
o The client will perform connectivity checks on all addresses
received in the SIP INVITE message(21)-(24). Note that steps
(21)-(24) will be repeated for every address offered in the SIP
INVITE request. This is not shown in the diagram for simplicity.
On receiving a response to a STUN connectivity check, the client
will start streaming media(*Note - if further STUN connectivity
responses are received after the client has started streaming meda
with a higher priority, it will be used instead).
o The STUN connectivity checks will then occur in the opposite
direction, as illustrated in Section 4.2.1.2.1. A STUN server
running on each advertised address will respond to incoming STUN
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connectivity requests(25)-(28).
o Bi-directional audio can now occur between the two clients.
4.2.2 Symmetric NAT
4.2.2.1 STUN Failure
This section highlights that while STUN is the preferred mechanism
for traversal of NAT, it does not solve every cases. The use of STUN
on its own will not guarantee traversal through every NAT type, hence
the recommendation that ICE be the preffered option.
Client SYMMETRIC STUN [..]
NAT Server
| | | |
|(1) STUN Req | | |
|src=10.0.1.1:5301 | | |
|----------------->| | |
| |(2) STUN Req | |
| |src=1.2.3.4:5601 | |
| |----------------->| |
| |(3) STUN Resp | |
| |<-----------------| |
| |map=1.2.3.4:5601 | |
| |dest=1.2.3.4:5601 | |
|(4) STUN Resp | | |
|<-----------------| | |
|map=1.2.3.4:5601 | | |
|dest=10.0.1.1:5301| | |
|(5)SIP INVITE | | |
|----------------->| | |
| |(6)SIP INVITE | |
| |------------------------------------>|
| | |(7)SIP 200 OK |
| |<------------------------------------|
|(8)SIP 200 OK | | |
|<-----------------| | |
|========================================================|
|>>>>>>>>>>>Outgoing Media sent from 10.0.1.1:5301>>>>>>>|
|========================================================|
| x=====================================|
| xIncoming Media sent to 1.2.3.4:5601<<|
| x=====================================|
|(9)SIP ACK | | |
|----------------->| | |
| |(10) SIP ACK | |
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| |------------------------------------>|
| | | |
Figure 25: Symmetric NAT with STUN - Failure
The example in Figure 25 is conveyed in the context of the client
behind the Symmetric NAT initiating a call. It should be noted that
the same problem applies when a client receives a SIP invitation and
is behind a Symmetric NAT.
o In Figure 25 the client behind the NAT obtains an external
representation using standard STUN mechanisms (1)-(4) that have
been used in previous examples in this document (e.g
Section 4.2.1.1.1).
o The external mapped address obtained is also used in the outgoing
SDP contained in the SIP INVITE request(5).
o In this example the client is still able to send media to the
external client. The problem occurs when the client outside the
NAT tries to use the address supplied in the outgoing INVITE
request to traverse media back through the Symmetric NAT.
o A symmetric NAT has differing rules from the Cone variety of NAT.
For any internal IP address and port mapping, data sent to
different external addresses does not provide the same public
mapping at the NAT. In Figure 25 the STUN query produced a valid
external mapping. This mapping, however, can only be used in the
context of the original STUN request that was sent to the STUN
server. Any packets that attempt to use the mapped address, that
does not come from the STUN server IP address and port, will be
dropped at the NAT. Figure 25 shows the media being dropped at
the NAT after (8).
4.2.2.2 TURN Solution
As identified in Section 4.2.2.1, STUN provides a useful tool for the
traversal of the majority of NATs but fails with symmetric type NAT.
This led to the development of the TURN solution[11] which introdcues
a media relay in the path for NAT traversal (as described in
Section 3.2.3). The following example explains how TURN solves the
previous failure when using STUN to traverse a symmetric NAT.
Client SYMMETRIC TURN [..]
NAT Server
| | | |
|(1) TURN Allocate | | |
|src=10.0.1.1:5301 | | |
|----------------->| | |
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| |(2) TURN Allocate | |
| |src=1.2.3.4:5601 | |
| |----------------->| |
| |(3) TURN Resp | |
| |<-----------------| |
| |map=2.3.4.5:5601 | |
| |dest=1.2.3.4:5601 | |
|(4) TURN Resp | | |
|<-----------------| | |
|map=2.3.4.5:5601 | | |
|dest=10.0.1.1:5301| | |
|(5)SIP INVITE | | |
|----------------->| | |
| |(6)SIP INVITE | |
| |------------------------------------>|
| | |(7)SIP 200 OK |
| |<------------------------------------|
|(8)SIP 200 OK | | |
|<-----------------| | |
|========================================================|
|>>>>>>>>>>>Outgoing Media sent from 10.0.1.1:5301>>>>>>>|
|========================================================|
| | |==================|
| | |<<<Media Sent to<<|
| | |<<< 2.3.4.5:5601<<|
| | |==================|
|=====================================| |
|<Incoming Media Sent to 1.2.3.4:5601<| |
|=====================================| |
|(9)SIP ACK | | |
|----------------->| | |
| |(10) SIP ACK | |
| |------------------------------------>|
| | | |
Figure 26: Symmetric NAT with STUN - Failure
o The client obtains a TURN derived address by issuing TURN allocate
request(1). The request traverses through the symmetric NAT and
reaches the TURN server (2). The Turn server generates a response
that contains an external representation. The representation maps
to an address mapping on the TURN server which is bound to the
public pin hole in the NAT, opened by the TURN request. This
results in any traffic being sent to the TURN server
representation (2.3.4.5:5601) will be redirected to the external
representation of the pin hole created by the TURN
request(1.2.3.4:5601).
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o The TURN derived address (2.3.4.5:5601) arrives back at the
originating client(4). This address can then be used in the SDP
for the outgoing SIP INVITE request as shown below (note that the
RTCP attribute would have been obtained by another TURN derived
address which is not shown in the call flow for simplicity):-
o
v=0
o=test 2890844342 2890842164 IN IP4 10.0.1.1
c=IN IP4 2.3.4.5
t=0 0
m=audio 5601 RTP/AVP 0
a=rtcp:5608
o On receiving the INVITE request, the UAS is able to stream media
to the TURN derived address (2.3.4.5:5601). As shown in
Figure 26, the media from the UAS is directed to the TURN derived
address at the TURN server. The TURN server then redirects the
traffic to the open pin hole in the symmetric NAT(1.2.3.4:5601).
The media traffic is then able to traverse the symmetric NAT and
arrives back at the client.
o The TURN solution on its own will work for Symmetric and other
types of NAT mentioned in this specification but should only be
used as a last resort. The relaying of media through an external
entity is not an efficient mechanism for all NAT traversal.
4.2.2.3 ICE Solution
The previous two examples have highlighted the problem with using
STUN for all forms of NAT traversal and a solution using TURN for the
symmetric NAT case. As mentioned previously in this document, the
RECOMMENDED mechanism for traversing all varieties of NAT is using
ICE, as detailed in Section 3.2.4. Ice maked use of STUN, TURN and
any other UNSAF [7] compliant protocol to provide a list of
prioritised addresses that can be used for media traffic. Detailed
examples of ICE can be found in Section 4.2.1.2.1 and in
Section 4.2.1.2.2. These examples are associated with a 'Port
Restricted' type NAT but can be applied to any NAT type variation,
including 'Symmetric' type NAT. The procedures are the same and of
the list of candidate addresses, a client will choose where to send
media dependant on the results of the STUN connectivity checks on
each candidate address and the associated priority (highest priority
wins). For more information see the core ICE specificagtion[13]
4.3 Advanced NAT media Traversal Using ICE
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4.3.1 Full Cone --> Full Cone traversal
4.3.1.1 Without NAT
4.3.1.1.1 Initiating Session
4.3.1.1.2 Receiving Session Invitation
4.3.1.2 With NAT
4.3.1.2.1 Initiating Session
4.3.1.2.2 Receiving Session Invitation
4.3.2 Port Restricted Cone --> Port Restricted Cone traversal
4.3.2.1 Without NAT
4.3.2.1.1 Initiating Session
4.3.2.1.2 Receiving Session Invitation
4.3.2.2 With NAT
4.3.2.2.1 Initiating Session
4.3.2.2.2 Receiving Session Invitation
4.3.3 Internal TURN Server (Enterprise Deployment)
4.3.3.1 Peer in same Enterprise
4.3.3.2 Peer in same Enterprise - Separated by NAT
4.3.3.3 Peer outside Enterprise
4.4 Intercepting Intermediary (B2BUA)
4.5 IPV4/IPV6
4.6 ICE with RTP/TCP
5. References
5.1 Normative References
[1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
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Session Initiation Protocol", RFC 3261, June 2002.
[2] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications",
RFC 1889, January 1996.
[3] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[5] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing",
RFC 3581, August 2003.
[6] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Registering Non-Adjacent Contacts",
RFC 3327, December 2002.
[7] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 2002.
[8] Rosenberg, J., Weinberger, J., Huitema, C. and R. Mahy, "STUN -
Simple Traversal of User Datagram Protocol (UDP) Through
Network Address Translators (NATs)", RFC 3489, March 2003.
[9] Jennings, C. and A. Hawrylyshen, "SIP Conventions for
Connection Usage",
Internet-Draft draft-jennings-sipping-outbound-00, October
2004.
[10] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", Internet-Draft draft-ietf-sip-gruu-02, July 2004.
[11] Rosenberg, J., "Traversal Using Relay NAT (TURN)",
Internet-Draft draft-rosenberg-midcom-turn-06, October 2004.
[12] Wing, D., "Symmetric RTP and RTCP Considered Helpful",
Internet-Draft draft-wing-mmusic-symmetric-rtprtcp-01, October
2004.
[13] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Multimedia Session Establishment Protocols",
Internet-Draft draft-ietf-mmusic-ice-03, October 2004.
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[14] Rosenberg, J., "Simple Traversal of UDP Through Network Address
Translators (NAT) (STUN)",
Internet-Draft draft-ietf-behave-rfc3489bis-00, October 2004.
5.2 Informative References
Authors' Addresses
Chris Boulton
Ubiquity Software Corporation
Eastern Business Park
St Mellons
Cardiff, South Wales CF3 5EA
Email: cboulton@ubiquitysoftware.com
Jonathan Rosenberg
Cisco Systems
600 Lanidex Plaza
Parsippany, NJ 07054
Email: jdrosen@dynamicsoft.com
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