SIPPING Working Group C. Boulton, Ed.
Internet-Draft Ubiquity Software Corporation
Expires: December 28, 2006 J. Rosenberg
Cisco Systems
G. Camarillo
Ericsson
June 26, 2006
Best Current Practices for NAT Traversal for SIP
draft-ietf-sipping-nat-scenarios-05
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Copyright (C) The Internet Society (2006).
Abstract
Traversal of the Session Initiation Protocol (SIP) and the sessions
it establishes through Network Address Translators (NAT) is a complex
problem. Currently there are many deployment scenarios and traversal
mechanisms for media traffic. This document aims to provide concrete
recommendations and a unified method for NAT traversal as well as
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documenting corresponding call flows.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Problem Statement . . . . . . . . . . . . . . . . . . . . . . 3
3. Solution Technology Outline Description . . . . . . . . . . . 6
3.1. SIP Signaling . . . . . . . . . . . . . . . . . . . . . . 7
3.1.1. Symmetric Response . . . . . . . . . . . . . . . . . . 7
3.1.2. Connection Re-use . . . . . . . . . . . . . . . . . . 8
3.2. Media Traversal . . . . . . . . . . . . . . . . . . . . . 8
3.2.1. Symmetric RTP . . . . . . . . . . . . . . . . . . . . 8
3.2.2. STUN . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2.3. TURN . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.2.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.2.5. 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 . . . . . . . . . . . . . . . . . 18
4.1.4. Receiving an Invitation to a Session . . . . . . . . . 20
4.2. Basic NAT Media Traversal . . . . . . . . . . . . . . . . 23
4.2.1. Endpoint independent NAT . . . . . . . . . . . . . . . 24
4.2.2. Port and Address Dependant NAT . . . . . . . . . . . . 40
4.3. Address independent Port Restricted NAT --> Address
independent Port Restricted NAT traversal . . . . . . . . 46
4.4. Internal TURN Usage (Enterprise Deployment) . . . . . . . 46
5. Intercepting Intermediary (B2BUA) . . . . . . . . . . . . . . 46
6. IPv4-IPv6 Transition . . . . . . . . . . . . . . . . . . . . . 47
6.1. IPv4-IPv6 Transition for SIP Signalling . . . . . . . . . 47
6.2. IPv4-IPv6 Transition for Media . . . . . . . . . . . . . . 47
7. ICE with RTP/TCP . . . . . . . . . . . . . . . . . . . . . . . 50
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 50
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 50
9.1. Normative References . . . . . . . . . . . . . . . . . . . 50
9.2. Informative References . . . . . . . . . . . . . . . . . . 52
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 53
Intellectual Property and Copyright Statements . . . . . . . . . . 54
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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 behaviors can be found in 'NAT Behavioral Requirements for
Unicast UDP' [11].
The IETF has produced many specifications for the traversal of NAT,
including STUN, ICE, rport, symmetric RTP, TURN, SIP Outbound, SDP
attribute for RTCP, and others. These each represent a part of the
solution, but none of them gives the overall context for how the NAT
traversal problem is decomposed and solved through this collection of
specifications. This document serves to meet that need.
This document attempts to provide a definitive set of 'Best Common
Practices' to demonstrate the traversal of SIP and its associated
media through NAT devices. The document does not propose any new
functionality but does draw on existing solutions for both core SIP
signaling and media traversal (as defined in Section 3).
The draft will be split into distinct sections as follows:
1. A clear definition of the problem statement
2. Description of proposed solutions for both SIP protocol signaling
and media signaling
3. A set of basic and advanced call flow scenarios
2. Problem Statement
The traversal of SIP through NAT can be split into two categories
that both require attention - The core SIP signaling and associated
media traversal.
The core SIP signaling has a number of issues when traversing through
NATs.
Firstly, the default operation for SIP response generation using
unreliable protocols such as the Unicast Datagram Protocol (UDP)
results in responses 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
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represents the listening port of the originating client and not the
port representing the open pin hole on the NAT. This results in
responses being sent back to the NAT but to a port that is likely not
open for SIP traffic. The SIP response will then be dropped at the
NAT. This is illustrated in Figure 1 which depicts a SIP response
being returned to port 5060.
Private NAT Public
Network | Network
|
|
-------- SIP Request |open port 5650 --------
| |-------------------->--->-----------------------| |
| | | | |
| Client | |port 5060 SIP Response | Proxy |
| | x<------------------------| |
| | | | |
-------- | --------
|
|
|
Figure 1
Secondly, when using a reliable, connection orientated transport
protocol such as TCP, SIP has an inherent mechanism that results in
SIP responses reusing the connection that was created/used for the
corresponding transactional request. The SIP protocol does not
provide a mechanism that allows new requests generated in the reverse
direction of the originating client to use the existing TCP
connection created between the client and the server during
registration. This results in the registered contact address not
being bound to the "connection" in the case of TCP. Requests are
then blocked at the NAT, as illustrated in Figure 2. This problem
also exists for unreliable transport protocols such as UDP where
external NAT mappings need to be re-used to reach a SIP entity on the
private side of the network.
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Private NAT Public
Network | Network
|
|
-------- (UAC 8023) REGISTER/Response (UAS 5060) --------
| |-------------------->---<-----------------------| |
| | | | |
| Client | |5060 INVITE (UAC 8015)| Proxy |
| | x<------------------------| |
| | | | |
-------- | --------
|
|
|
Figure 2
In Figure 2 the original REGISTER request is sent from the client on
port 8023 and received on port 5060, establishing a reliable
connection and opening a pin-hole in the NAT. The generation of a
new request from the proxy results in a request destined for the
registered entity (Contact IP address) which is not reachable from
the public network. This results in the new SIP request attempting
to create a connection to a private network address. This problem
would be solved if the original connection was re-used. While this
problem has been discussed in the context of connection orientated
protocols such as TCP, the problem exists for SIP signaling using any
transport protocol. The solution proposed for this problem in
section 3 of this document is relevant for all SIP signaling,
regardless of the transport protocol.
NAT policy can dictate that connections should be closed after a
period of inactivity. This period of inactivity can range
drastically from a number seconds to hours. Pure SIP signaling can
not be relied upon to keep alive connections for a number of reasons.
Firstly, SIP entities can sometimes have no signaling traffic for
long periods of time which has the potential to exceed the inactivity
timer, and this can lead to problems where endpoints are not
available to receive incoming requests as the connection has been
closed. Secondly, if a low inactivity timer is specified, SIP
signaling is not appropriate as a keep-alive mechanism as it has the
potential to add a large amount of traffic to the network which uses
up valuable resource and also requires processing at a SIP stack,
which is also a waste of processing resources.
Media associated with SIP calls also has problems traversing NAT.
RTP [2]] is one of the most common media transport types used in SIP
signaling. Negotiation of RTP occurs with a SIP session
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establishment using the Session Description Protocol(SDP) [3] and a
SIP offer/answer exchange[5]. During a SIP offer/answer exchange an
IP address and port combination are specified by each client in a
session as a means of receiving media such as RTP. The problem
arises when a client advertises its address to receive media and it
exists in a private network that is not accessible from outside the
NAT. Figure 3 illustrates this problem.
NAT Public Network NAT
| |
| |
| |
-------- | SIP Signaling Session | --------
| |----------------------->---<--------------------| |
| | | | | |
| Client | | | | Client |
| A |>=====>RTP>==Unknown Address==>X | | B |
| | | X<==Unknown Address==<RTP<===<| |
-------- | | --------
| |
| |
| |
Figure 3
The connection address representing both clients are not available on
the public internet and traffic can be sent from both clients through
their NATs. The problem occurs when the traffic reaches the public
internet and is not resolvable. The media traffic fails. The
connection address extracted from the SDP payload is that of an
internal address, and so not resolvable from the public side of the
NAT. To complicate the problem further, a number of different NAT
topologies with different default behaviors increase the difficulty
of proposing a single solution.
3. Solution Technology Outline Description
When analyzing issues associated with traversal of SIP through
existing NAT, it has been identified that the problem can be split
into two clear solution areas as defined in section 2 of this
document. The traversal of the core protocol signaling and the
traversal of the associated media as specified in the Session
Description Payload (SDP) of a SIP offer/answer exchange[5]. The
following sub-sections outline solutions that enable core SIP
signaling and its associated media to traverse NATs.
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3.1. SIP Signaling
SIP signaling has two areas that result in transactional failure when
traversing through NAT, as described in section 2 of this document.
The remaining sub-sections describe appropriate solutions that result
in SIP signalling traversal through NAT, regardless of transport
protocol. IT is RECOMMEDED that SIP compliant entities follow the
guidelines presented in this section to enable traversal of SIP
signaling through NATs.
3.1.1. Symmetric Response
As described in section 2 of this document, when using an unreliable
transport protocol such as UDP, SIP responses are sent to the IP
address and port combination contained in the SIP 'Via' header field
(or default port for the appropriate transport protocol if not
present). This can result in responses being blocked at a NAT. In
such circumstances, SIP signaling requires a mechanism that will
allow entities to override the basic response generation mechanism in
RFC 3261 [1]. Once the SIP response is constructed, the destination
is still derived using the mechanisms described in RFC 3261 [1]. The
port (to which the response will be sent), however, will not equal
that specified in the SIP 'Via' header field but will be the port
from which the original request was sent. This results in the pin-
hole opened for the requests traversal of the NAT being reused, in a
similar manner to that of reliable connection orientated transport
protocols such as TCP. Figure 4 illustrates the response traversal
through the open pin hole using this method.
Private NAT Public
Network | Network
|
|
-------- | --------
| | | | |
| |send/receive | send/receive| |
| Client |port 5060-----<<->>---------<<->>-----port 5060| Client |
| A | | | B |
| | | | |
-------- | --------
|
|
|
Figure 4
The exact functionality for this method of response traversal is
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called 'Symmetric Response' and the details are documented in RFC
3581 [6]. Additional requirements are imposed on SIP entities in
this specification such as listening and sending SIP requests/
responses from the same port.
3.1.2. Connection Re-use
The second problem with sip signaling, as defined in Section 2 and
illustrated in Figure 2, is to allow incoming requests to be properly
routed.
Guidelines for devices such as User Agents that can only generate
outbound connections through a NAT are documented in 'SIP Conventions
for UAs with Outbound Only Connections'[13]. The document provides
techniques that use a unique User Agent instance identifier
(instance-id) in association with a flow identifier (Reg-id). The
combination of the two identifiers provides a key to a particular
connections (both UDP and TCP) that are stored in association with
registration bindings. On receiving an incoming request to a SIP
Address-Of-Record (AOR), a proxy routes to the associated flow
created by the registration and thus a route through a NAT. It also
provides a keepalive mechanism for clients to keep NAT bindings
alive. This is achieved using peer-to-peer STUN multiplexed over the
SIP signaling connection. Usage of this specification is
RECOMMENDED. This mechanism is not transport specific and should be
used for any transport protocol.
Even if the SIP Outbound draft is not used, clients generating SIP
requests SHOULD use the same IP address and port (i.e., socket) for
both transmission and receipt of SIP messages. Doing so allows for
the vast majority of industry provided solutions to properly
function.
3.2. Media Traversal
This document has already provided guidelines that recommend using
extensions to the core SIP protocol to enable traversal of NATs.
While ultimately not desirable, the additions are relatively straight
forward and provide a simple, universal solution for varying types of
NAT deployment. The issues of media traversal through NATs is not
straight forward and requires the combination of a number of
traversal methodologies. The technologies outlined in the remainder
of this section provide the required solution set.
3.2.1. Symmetric RTP
The primary problem identified in section 2 of this document is that
internal IP address/port combinations can not be reached from the
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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 [15]. 'Symmetric RTP' SHOULD only be used for
traversal of RTP through NAT when one of the participants in a media
session definitively knows that it is on the public network.
3.2.1.1. RTCP Attribute
Normal practice when selecting a port for defining Real Time Control
Protocol(RTCP) [2] is for consecutive order numbering (i.e select an
incremented port for RTCP from that used for RTP). This assumption
causes RTCP traffic to break when traversing many NATs due to blocked
ports. To combat this problem a specific address and port need to be
specified in the SDP rather than relying on such assumptions. RFC
3605 [6] defines an SDP attribute that is included to explicitly
specify transport connection information for RTCP. The address
details can be obtained using any appropriate method including those
detailed previously in this section (e.g. STUN, TURN).
3.2.2. STUN
Simple Traversal of User Datagram Protocol (UDP) through Network
Address Translators(NAT) or STUN is defined in RFC 3489bis [10].
STUN is a lightweight tool kit and protocol that provides details of
the external IP address/port combination used by the NAT device to
represent the internal entity on the public facing side of a NAT. On
learning of such an external representation, a client can use it
accordingly as the connection address in SDP to provide NAT
traversal. Using terminology defined in the draft 'NAT Behavioral
Requirements for Unicast UDP' [11], STUN does work with 'Endpoint
Independent Mapping' but does not work with either 'Address Dependent
Mapping' or 'Address Dependent and Port Mapping' type NATs. Using
STUN with either of the previous two NAT mappings to probe for the
external IP address/port representation will provide a different
result to that required for traversal by an alternative SIP entity.
The IP address/port combination deduced for the STUN server would be
blocked for incoming packets from an alterative SIP entity.
As mentioned in Section 3.1.2, STUN is also used as a peer-to-peer
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keep-alive mechanism.
3.2.3. TURN
As described in the previous section, the STUN protocol does not work
for UDP traversal through certain identified NAT mappings.
'Obtaining Relay Addresses from Simple Traversal of UDP Through NAT
(known as TURN)' is a usage of the STUN protocol for deriving (from a
STUN server) an address that will be used to relay packet towards a
client. TURN provides an external address (globally routable) at a
STUN server that will act as a media relay which guarantees traffic
will reach the associated internal address. The full details of the
TURN specification are defined in [12]. A TURN service will almost
always provide media traffic to a SIP entity but it is RECOMMENDED
that this method only be used as a last resort and not as a general
mechanism for NAT traversal. This is because using TURN has high
performance costs when relaying media traffic and can lead to
unwanted latency.
3.2.4. ICE
Interactive Connectivity Establishment (ICE) is the RECOMMENDED
method for traversal of existing NAT if Symmetric RTP is not
appropriate. ICE is a methodology for using existing technologies
such as STUN, TURN and any other UNSAF[9] compliant protocol to
provide a unified solution. This is achieved by obtaining as many
representative IP address/port combinations as possible using
technologies such as STUN/TURN etc. Once the addresses are
accumulated, they are all included in the SDP exchange in a new media
attribute called 'candidate'. Each 'candidate' SDP attribute entry
has detailed connection information including a media addresses
(including optional RTCP information), priority, username, password
and a unique session ID. The appropriate IP address/port
combinations are used in the correct order depending on the specified
priority. A client compliant to the ICE specification will then
locally run instances of STUN servers on all addresses being
advertised using ICE. Each instance will undertake connectivity
checks to ensure that a client can successfully receive media on the
advertised address. Only connections that pass the relevant
connectivity checks are used for media exchange. The full details of
the ICE methodology are contained in [16].
3.2.5. Solution Profiles
This draft has documented a number of technology solutions for the
traversal of media through differing NAT deployments. A number of
'profiles' will now be defined that categorize varying levels of
support for the technologies described.
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3.2.5.1. Primary Profile
A client falling into the 'Primary' profile supports ICE in
conjunction with STUN, TURN and RFC 3605 [6] for RTCP. ICE is used
in all cases and falls back to standard operation when dealing with
non-ICE clients. A client which falls into the 'Primary' profile
will be maximally interoperable and function in a rich variety of
environments including enterprise, consumer and behind all varieties
of NAT.
3.2.5.2. Consumer Profile
A client falling into the 'Consumer' profile supports STUN and RFC
3605 [6] for RTCP. It uses STUN to allocate bindings, and can also
detect when it is in the unfortunate situation of being behind a
'Symmetric' NAT, although it simply cannot function in this case.
These clients will only work in deployment situations where the
access is sufficiently controlled to know definitively that there
won't be Symmetric NAT. This is hard to guarantee as users can
always pick up their client and connect via a different access
network.
3.2.5.3. 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.
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.
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4.1.1.1. UDP
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
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:
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REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK
Max-Forwards: 70
Supported: path,gruu
From: Client <sip:client@example.com>;tag=djks8732
To: Client <sip:client@example.com>
Call-ID: 763hdc73y7dkb37@example.com
CSeq: 1 REGISTER
Contact: <sip:client@client.example.com>;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E120>"
Content-Length: 0
This proxy now generates a SIP 401 response to challenge for
authentication, as depicted in (2) from Figure 5:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP client.example.com:5060
;rport=8050;branch=z9hG4bK;received=192.0.1.2
From: Client <sip:client@example.com>;tag=djks8732
To: Client <sip:client@example.com>;tag=876877
Call-ID: 763hdc73y7dkb37@example.com
CSeq: 1 REGISTER
WWW-Authenticate: [not shown]
Content-Length: 0
The response will be sent to the address appearing in the 'received'
parameter of the SIP 'Via' header (address 192.0.1.2). The response
will not be sent to the port deduced from the SIP 'Via' header, as
per standard SIP operation but will be sent to the value that has
been stamped in the 'rport' parameter of the SIP 'Via' header (port
8050). For the response to successfully traverse the NAT, all of the
conventions defined in RFC 3581 [6] MUST be obeyed. Make note of the
both the 'connectionID' and 'sip.instance' contact header parameters.
They are used to establish a connection re-use tuple as defined in
[13]. 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]
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4.1.1.2. Reliable Transport
Client NAT Registrar
| | |
|(1) REGISTER | |
|----------------->| |
| | |
| |(1) REGISTER |
| |----------------->|
| | |
| |(2) 401 Unauth |
| |<-----------------|
| | |
|(2) 401 Unauth | |
|<-----------------| |
| | |
|(3) REGISTER | |
|----------------->| |
| | |
| |(3) REGISTER |
| |----------------->|
| | |
|*************************************|
| Create Connection Re-use Tuple |
|*************************************|
| | |
| |(4) 200 OK |
| |<-----------------|
| | |
|(4) 200 OK | |
|<-----------------| |
| | |
Figure 6.
Traversal of SIP REGISTER requests/responses using a reliable,
connection orientated protocol such as TCP does not require any
additional core SIP signaling extensions. SIP responses will re-use
the connection created for the initial REGISTER request, (1) from
Figure 6:
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REGISTER sip:proxy.example.com SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bKyilassjdshfu
Max-Forwards: 70
Supported: path,gruu
From: Client <sip:client@example.com>;tag=djks809834
To: Client <sip:client@example.com>
Call-ID: 763hdc783hcnam73@example.com
CSeq: 1 REGISTER
Contact: <sip:client@client.example.com;transport=tcp>;reg-id=1
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E121>"
Content-Length: 0
This example was included to show the inclusion of the connection re-
use Contact header parameters as defined in the Connection Re-use
draft [13]. 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.
[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.
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Client NAT Proxy Registrar
| | | |
|(1) REGISTER | | |
|----------------->| | |
| | | |
| |(1) REGISTER | |
| |----------------->| |
| | | |
| | |(2) REGISTER |
| | |----------------->|
| | | |
| | |(3) 401 Unauth |
| | |<-----------------|
| | | |
| |(4) 401 Unauth | |
| |<-----------------| |
| | | |
|(4)401 Unauth | | |
|<-----------------| | |
| | | |
|(5)REGISTER | | |
|----------------->| | |
| | | |
| |(5)REGISTER | |
| |----------------->| |
| | | |
| | |(6)REGISTER |
| | |----------------->|
| | | |
| | |(7)200 OK |
| | |<-----------------|
| | | |
|********************************************************|
| Create Connection Re-use Tuple |
|********************************************************|
| | | |
| |(8)200 OK | |
| |<-----------------| |
| | | |
|(8)200 OK | | |
|<-----------------| | |
| | | |
Figure 7.
This scenario builds on the previous example contained in
Section 4.1.1.2. The primary difference being that the REGISTER
request is routed onwards from a Proxy Server to a separated
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Registrar. The important message to note is (6) 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 [13] . At this point, the proxy server routes
the SIP REGISTER message to the Registrar. The proxy will create the
connection tuple as described in SIP Outbound at the same moment as
the co-located example, but for subsequent messages to arrive at the
Proxy, the element needs to request to remain in the signaling path.
To achieve this the proxy inserts to REGISTER message (5) a SIP PATH
extension header, as defined in RFC 3327 [7]. The previously created
flow 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. 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: path,gruu
From: Client <sip:client@example.com>;tag=djks809834
To: Client <sip:client@example.com>
Call-ID: 763hdc783hcnam73@example.com
CSeq: 1 REGISTER
Path: <sip:3HS28o8HAKJSH&&U@proxy.example.com;lr>
Contact: <sip:client@client.example.com;transport=tcp>;
;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00A95A0E121>";reg-id=1
Content-Length: 0
This REGISTER request results in the Path header being stored along
with the AOR and it's associated binding at the Registrar. The URI
contained in the Path header will be inserted as a pre-loaded SIP
'Route' header into any request that arrives at the Registrar and is
directed towards the associated binding. This guarantees that all
requests for the new Registration will be forwarded to the Edge
Proxy. In our example, the user part of the SIP 'Path' header URI
that was inserted by the Edge Proxy contains the unique token
identifying the flow to the client. On receiving subsequent
requests, the edge proxy will examine the user part of the pre-loaded
SIP 'route' header and extract the unique flow token for use in its
connection tuple comparison, as defined in the SIP Outbound
specification [13]. An example which builds on this scenario
(showing an inbound request to the AOR) is detailed in section
4.1.4.2 of this document.
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4.1.3. Initiating a Session
This section covers basic SIP signaling when initiating a call from
behind a NAT.
4.1.3.1. UDP
Initiating a call using UDP.
Client NAT Proxy [..]
| | |
|(1) INVITE | | |
|----------------->| | |
| | | |
| |(1) INVITE | |
| |----------------->| |
| | | |
| |(2) 407 Unauth | |
| |<-----------------| |
| | | |
|(2) 407 Unauth | | |
|<-----------------| | |
| | | |
|(3) INVITE | | |
|----------------->| | |
| | | |
| |(3) INVITE | |
| |----------------->| |
| | | |
| | |(4) INVITE |
| | |---------------->|
| | | |
| | |(5)180 RINGING |
| | |<----------------|
| | | |
| |(6)180 RINGING | |
| |<-----------------| |
| | | |
|(6)180 RINGING | | |
|<-----------------| | |
| | | |
| | |(7)200 OK |
| | |<----------------|
| | | |
| |(8)200 OK | |
| |<-----------------| |
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| | | |
|(8)200 OK | | |
|<-----------------| | |
| | | |
|(9)ACK | | |
|----------------->| | |
| | | |
| |(9)ACK | |
| |----------------->| |
| | | |
| | |(10) ACK |
| | |---------------->|
| | | |
Figure 8.
The initiating client generates an INVITE request that is to be sent
through the NAT to a Proxy server. The INVITE message is represented
in Figure 8 by (1) and is as follows:
INVITE sip:clientB@example.com SIP/2.0
Via: SIP/2.0/UDP client.example.com:5060;rport;branch=z9hG4bK74husdHG
Max-Forwards: 70
Route: <sip:proxy.example.com;lr>
From: clientA <sip:clientA@example.com>;tag=7skjdf38l
To: clientB <sip:clientB@example.com>
Call-ID: 8327468763423@example.com
CSeq: 1 INVITE
Contact:<sip:clientA@example.com;gruu
;opaque=urn:uuid:ijed7ush-4jan-53120-aee5-e0aecwee6wef;grid=45a>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
There are a number of points to note with this message:
1. Firstly, as with the registration example in Section 4.1.1.1,
responses to this request will not automatically pass back
through a NAT and so the SIP 'Via' header 'rport' is included as
described in the 'Symmetric response' Section 3.1.1 and defined
in RFC 3581 [6].
2. Secondly, the contact inserted contains the GRUU previously
obtained from the SIP 200 OK response to the registration. Use
of the GRUU ensures that any SIP requests within the dialog that
in the opposite direction will be able to traverse the NAT. This
occurs using the mechanisms defined in the SIP Outbound
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specification[13]. A request arriving at the entity which
resolves to the GRUU is then able to determine a previously
registered connection that will allow the request to traverse the
NAT and reach the intended endpoint.
4.1.3.2. Reliable Transport
When using a reliable transport such as TCP the call flow and
procedures for traversing a NAT are almost identical to those
described in Section 4.1.3.1. The primary difference when using
reliable transport protocols is that Symmetric response[6] are not
required for SIP responses to traverse a NAT. RFC 3261[1] defines
procedures for SIP response messages to be sent back on the same
connection on which the request arrived.
4.1.4. Receiving an Invitation to a Session
This section details scenarios where a client behind a NAT receives
an inbound request through a NAT. These scenarios build on the
previous registration scenario from Section 4.1.1 and Section 4.1.2
in this document.
4.1.4.1. Registrar/Proxy Co-located
The core SIP signaling associated with this call flow is not impacted
directly by the transport protocol and so only one example scenario
is necessary. The example uses UDP and follows on from the
registration installed in the example from Section 4.1.1.1.
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Client NAT Registrar/Proxy SIP Entity
| | | |
|*******************************************************|
| Registration Binding Installed in |
| section 4.1.1.1 |
|*******************************************************|
| | | |
| | |(1)INVITE |
| | |<----------------|
| | | |
| |(2)INVITE | |
| |<-----------------| |
| | | |
|(2)INVITE | | |
|<-----------------| | |
| | | |
| | | |
Figure 9.
An INVITE request arrives at the Registrar with a destination
pointing to the AOR of that inserted in section 4.1.1.1. The message
is illustrated by (1) in Figure 9 and looks as follows:
INVITE sip:client@example.com SIP/2.0
Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: External <sip:External@external.example.com>;tag=7893hd
To: client <sip:client@example.com>
Call-ID: 8793478934897@external.example.com
CSeq: 1 INVITE
Contact: <sip:external@192.0.1.4>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
The INVITE request matches the registration binding previously
installed at the Registrar and the INVITE request-URI is re-written
to the selected onward address. The proxy then examines the request
URI of the INVITE and compares with its list of current open flows.
It uses the incoming AOR to commence the check for associated open
connections/mappings. Once matched, the proxy checks to see if the
unique instance identifier (+sip.instance) associated with the
binding equals the same instance identifier associated with the flow.
The request is then dispatched on the appropriate flow. This is
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message (2) from Figure 9 and is as follows:
INVITE sip:sip:client@client.example.com SIP/2.0
Via: SIP/2.0/UDP proxy.example.com;branch=z9hG4kmlds893jhsd
Via: SIP/2.0/UDP external.example.com;branch=z9hG4bK74huHJ37d
Max-Forwards: 70
From: External <sip:External@external.example.com>;tag=7893hd
To: client <sip:client@example.com>
Call-ID: 8793478934897@external.example.com
CSeq: 1 INVITE
Contact: <sip:external@192.0.1.4>
Content-Type: application/sdp
Content-Length: ..
[SDP not shown]
It is a standard SIP INVITE request with no additional functionality.
The major difference being that this request will not follow the
address specified in the Request-URI, as standard SIP rules would
enforce but will be sent on the flow associated with the registration
binding (look-up procedures in RFC 3263 [6] are overridden). This
then allows the original connection/mapping from the initial
registration process to be re-used.
4.1.4.2. Registrar/Proxy Not Co-located
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Client NAT Proxy Registrar SIP Entity
| | | | |
|***********************************************************|
| Registration Binding Installed in |
| section 4.1.2 |
|***********************************************************|
| | | | |
| | | |(1)INVITE |
| | | |<-------------|
| | | | |
| | |(2)INVITE | |
| | |<-------------| |
| | | | |
| |(3)INVITE | | |
| |<-------------| | |
| | | | |
|(3)INVITE | | | |
|<-------------| | | |
| | | | |
| | | | |
Figure 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.
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 reflexive transport address.
o The "Rel" attribute represents the relayed transport address.
The meaning of each STUN attribute is extensively explained in the
core STUN[10] and TURN usage[12] drafts.
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 a
representation of the client responsible for the address. For
example in the majority of the examples "L" (left client), "R" (right
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client), NAT-PUB" (NAT public), PRIV (Private), and "STUN-PUB" (STUN
Public) are used. To allow for multiple addresses from the same
network element, each representation can also be followed by a
number. These can also be used in combination. For example "L-NAT-
PUB-1" would represent a public network address on the left hand side
NAT while "R-NAT-PUB-1" would represent a public network address in
the right hand side of the NAT. "L-PRIV-1" would represent a private
network address on the left hand side of the NAT while "R-PRIV-1"
represents a private address on the right hand side of the NAT.
It should also be noted that during the examples it might be
appropriate to signify an explicit part of a transport address. This
is achieved by adding either the '.address' or '.port' tag on the end
of the representation. For example, 'L-PRIV-1.address' and 'L-PRIV-
1.port'.
4.2.1. Endpoint independent NAT
This section demonstrates an example of a client both initiating and
receiving calls behind an 'Endpoint independent' NAT. An example is
included for both STUN and ICE with ICE being the RECOMMENDED
mechanism for media traversal.
4.2.1.1. STUN Solution
It is possible to traverse media through an 'Endpoint Independent NAT
using STUN. The remainder of this section provides a simplified
examples of the 'Binding Discovery' STUN usage as defined in [10].
The STUN messages have been simplified and do not include 'Shared
Secret' requests used to obtain the temporary username and password.
[Editors Note: Expand to show full flow in including Auth?.]
4.2.1.1.1. Initiating Session
The following example demonstrates media traversal through a NAT
demonstrating 'Address Independent' NAT behavior 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=L-PRIV-1 | | |
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|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(2) STUN Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(3) STUN Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
|(4) STUN Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|Map=NAT-PUB-1 | | |
| | | |
|(5) STUN Req | | |
|Src=L-PRIV-2 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(6) STUN Req | |
| |Src=NAT-PUB-2 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(7) STUN Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-2 | |
| |Map=NAT-PUB-2 | |
| | | |
|(8) STUN Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-2 | | |
|Map=NAT-PUB-2 | | |
| | | |
|(9)SIP INVITE | | |
|----------------->| | |
| | | |
| |(10)SIP INVITE | |
| |------------------------------------>|
| | | |
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| | |(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 18: Address and Port Dependant NAT with STUN - 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 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 and STUN server.
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 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
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receive media (see 'Map=NAT-PUB-1' 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 (Address has changed from "L-PRIV-1" to "L-PRIV-2").
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 $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 mapped address obtained from the STUN 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 STUN binding is placed in the RTCP attribute (as
discussed in Section 3.2.1.1) for traversal of RTCP (a=rtcp:NAT-
PUB-2.port).
o The SIP 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 and RTCP will not pass through the
NAT as no outbound association has been created with the far end
client. Two way media communication has now been established.
4.2.1.1.2. Receiving Session Invitation
Receiving a session for an 'Address and Port dependant' 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.
Client NAT STUN [..]
Server
| | | (1)SIP INVITE |
| |<-----------------|------------------|
| | | |
|(2) SIP INVITE | | |
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|<-----------------| | |
| | | |
|(3) STUN Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(4) STUN Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(5) STUN Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
|(6) STUN Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-1 | | |
|Map=NAT-PUB-1 | | |
| | | |
|(7) STUN Req | | |
|Src=L-PRIV-2 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(8) STUN Req | |
| |Src=NAT-PUB-2 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(9) STUN Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-2 | |
| |Map=NAT-PUB-2 | |
| | | |
|(10) STUN Resp | | |
|<-----------------| | |
|Src=STUN-PUB | | |
|Dest=L-PRIV-2 | | |
|Map=NAT-PUB-2 | | |
| | | |
|(11)SIP 200 OK | | |
|----------------->| | |
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| |(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 20: Restricted NAT with STUN - 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 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 and
STUN Server.
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 The STUN server receives the request and processes it
appropriately. This results in a successful STUN response being
generated and returned (5). The message contains details of the
mapped public address (contained in the STUN MAPPED-ADDRESS
attribute) which is to be used by the originating client to
receive media (see 'Map=NAT-PUB-1' from (5)).
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o The STUN response traverses back through the NAT using the binding
created by the outgoing 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 and is
represented by L-PRIV-2 in (7)).
o The client now constructs a SIP 200 OK message (11) in response to
the original SIP INVITE requests. Note that traversal of SIP is
not covered in this example and is discussed in earlier sections
of the document. SIP Provisional responses are also left out for
simplicity. The 200 OK response will use the addresses it has
obtained in the previous STUN transactions to populate the SDP of
the SIP 200 OK as shown below:
v=0
o=test 2890844526 2890842807 IN IP4 $L-PRIV-1.address
c=IN IP4 $NAT-PUB-1.address
t=0 0
m=audio $NAT-PUB-1.port RTP/AVP 0
a=rtcp:$NAT-PUB-2.port
o Note that the mapped address obtained from the initial STUN
transaction is inserted as the connection address for the SDP
(c=NAT-PUB-1.address). The Primary port for RTP is also inserted
in the SDP (m=audio NAT-PUB-1.port RTP/AVP 0). Finally, the port
gained from the additional binding is placed in the RTCP attribute
(as discussed in Section 3.2.1.1) for traversal of RTCP (a=rtcp:
NAT-PUB-2.port).
o The SIP 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, regardless of the NAT type. The
following examples illustrate the traversal of an 'Endpoint
independent' NAT for both an initiating. The example only covers ICE
in association with STUN and TURN usage.
4.2.1.2.1. Initiating Session
The following example demonstrates an initiating traversal through an
'Endpoint independent' NAT using ICE.
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[Editors Note: Example needs to be expanded to include more ICE
detail e.g. timers etc.]
L NAT STUN NAT R
Server
| | | | |
|(1) Alloc Req | | | |
|Src=L-PRIV-1 | | | |
|Dest=STUN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(2) Alloc Req | | |
| |Src=L-NAT-PUB-1 | | |
| |Des=STUN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(3) Alloc Resp | | |
| |<---------------| | |
| |Src=STUN-PUB-1 | | |
| |Dest=L-NAT-PUB-1| | |
| |Map=L-NAT-PUB-1 | | |
| |Rel=STUN-PUB-2 | | |
| | | | |
|(4) Alloc Resp | | | |
|<---------------| | | |
|Src=STUN-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
|Map=L-NAT-PUB-1 | | | |
|Rel=STUN-PUB-2 | | | |
| | | | |
|(5) STUN Req | | | |
|Src=L-PRIV-2 | | | |
|Dest=STUN-PUB-1 | | | |
|--------------->| | | |
| | | | |
| |(6) Alloc Req | | |
| |Src=L-NAT-PUB-2 | | |
| |Dest=STUN-PUB-1 | | |
| |--------------->| | |
| | | | |
| |(7) Alloc Resp | | |
| |<---------------| | |
| |Src=STUN-PUB-1 | | |
| |Dest=NAT-PUB-2 | | |
| |Map=NAT-PUB-2 | | |
| |Rel=STUN-PUB-3 | | |
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| | | | |
|(8) Alloc Resp | | | |
|<---------------| | | |
|Src=STUN-PUB-1 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
|Rel=STUN-PUB-3 | | | |
| | | | |
|(9) SIP INVITE | | | |
|------------------------------------------------->| |
| | | | |
| | | |(10) SIP INVITE |
| | | |--------------->|
| | | | |
| | | |(11) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=STUN-PUB-1 |
| | | | |
| | |(12) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=STUN-PUB-1 | |
| | | | |
| | |(13) Alloc Res | |
| | |--------------->| |
| | |Src=STUN-PUB-1 | |
| | |Dest=R-NAT-PUB-1| |
| | |Map=R-NAT-PUB-1 | |
| | |Rel=STUN-PUB-4 | |
| | | | |
| | | |(14) Alloc Res |
| | | |--------------->|
| | | |Src=STUN-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | |Map=R-NAT-PUB-1 |
| | | |Rel=STUN-PUB-4 |
| | | | |
| | | |(15) Alloc Req |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=STUN-PUB-1 |
| | | | |
| | |(16) Alloc Req | |
| | |<---------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=STUN-PUB-1 | |
| | | | |
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| | |(17) Alloc Res | |
| | |--------------->| |
| | |Src=STUN-PUB-1 | |
| | |Dest=R-NAT-PUB-2| |
| | |Map=R-NAT-PUB-2 | |
| | |Rel=STUN-PUB-5 | |
| | | | |
| | | |(18) Alloc Res |
| | | |--------------->|
| | | |Src=STUN-PUB-1 |
| | | |Dest=R-PRIV-2 |
| | | |Map=R-NAT-PUB-2 |
| | | |Rel=STUN-PUB-5 |
| | | | |
| | | |(19) SIP 200 OK |
| |<-------------------------------------------------|
| | | | |
|(20) SIP 200 OK | | | |
|<---------------| | | |
| | | | |
|(21) SIP ACK | | | |
|------------------------------------------------->| |
| | | | |
| | | |(22) SIP ACK |
| | | |--------------->|
| | | | |
|(23) Bind Req | | | |
|------------------------>x | | |
|Src=L-PRIV-1 | | | |
|Dest=R-PRIV-1 | | | |
| | | | |
|(24) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-1 | | | |
|Dest=R-NAT-PUB-1| | | |
| | | | |
| |(25) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-1 | | |
| |Dest=R-NAT-PUB-1| | |
| | | | |
| | | |(26) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-1 |
| | | |Dest=R-PRIV-1 |
| | | | |
| | | |(27) Bind Res |
| | | |<---------------|
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| | | |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 >>>>>>>>>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(30) Bind Req |
| | | x<-----------------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-PRIV-1 |
| | | | |
| | | |(31) Bind Req |
| | | |<---------------|
| | | |Src=R-PRIV-1 |
| | | |Dest=L-NAT-PUB-1|
| | | | |
| | |(32) Bind Req | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-1 | |
| | |Dest=L-NAT-PUB-1| |
| | | | |
|(33) Bind Req | | | |
|<---------------| | | |
|Src=R-NAT-PUB-1 | | | |
|Dest=L-PRIV-1 | | | |
| | | | |
|(34) Bind Res | | | |
|--------------->| | | |
|Src=L-PRIV-1 | | | |
|Dest=R-NAT-PUB-1| | | |
|Map=R-NAT-PUB-1 | | | |
| | | | |
| |(35) Bind Res | | |
| |-------------------------------->| |
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| |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 <<<<<<<<<<<<<<<<<<<<<<<|
|===================================================================|
| | | | |
|(37) Bind Req | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
| | | | |
| |(38) Bind Req | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-2 | | |
| |Dest=R-NAT-PUB-2| | |
| | | | |
| | | |(39) Bind Req |
| | | |--------------->|
| | | |Src=L-NAT-PUB-2 |
| | | |Dest=R-PRIV-2 |
| | | | |
| | | |(40) Bind Res |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=L-NAT-PUB-2|
| | | |Map=L-NAT-PUB-2 |
| | | | |
| | |(41) Bind Res | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=L-NAT-PUB-2| |
| | |Map=L-NAT-PUB-2 | |
| | | | |
|(42) Bind Res | | | |
|<---------------| | | |
|Src=R-NAT-PUB-2 | | | |
|Dest=L-PRIV-2 | | | |
|Map=L-NAT-PUB-2 | | | |
| | | | |
|===================================================================|
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|>>>>>>>>>>>>>>>>>>>>>Outgoing RTCP sent from >>>>>>>>>>>>>>>>>>>>>>|
|===================================================================|
| | | | |
| | | |(43) Bind Req |
| | | |<---------------|
| | | |Src=R-PRIV-2 |
| | | |Dest=L-NAT-PUB-2|
| | | | |
| | |(44) Bind Req | |
| |<--------------------------------| |
| | |Src=R-NAT-PUB-2 | |
| | |Dest=L-NAT-PUB-2| |
| | | | |
|(45) Bind Req | | | |
|<---------------| | | |
|Src=R-NAT-PUB-2 | | | |
|Dest=L-PRIV-2 | | | |
| | | | |
|(46) Bind Res | | | |
|--------------->| | | |
|Src=L-PRIV-2 | | | |
|Dest=R-NAT-PUB-2| | | |
|Map=R-NAT-PUB-2 | | | |
| | | | |
| |(47) Bind Res | | |
| |-------------------------------->| |
| |Src=L-NAT-PUB-2 | | |
| |Dest=R-NAT-PUB-2| | |
| |Map=R-NAT-PUB-2 | | |
| | | | |
| | | |(48) Bind Res |
| | | |--------------->|
| | | |Src=L-NAT-PUB-2 |
| | | |Dest=R-PRIV-2 |
| | | |Map=R-NAT-PUB-2 |
| | | | |
|===================================================================|
|<<<<<<<<<<<<<<<<<<<<<Outgoing RTCP sent from <<<<<<<<<<<<<<<<<<<<<<|
|===================================================================|
| | | | |
|(49) SIP INVITE | | | |
|------------------------------------------------->| |
| | | | |
| | | |(50) SIP INVITE |
| | | |--------------->|
| | | | |
| | | |(51) SIP 200 OK |
| |<-------------------------------------------------|
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| | | | |
|(52) SIP 200 OK | | | |
|<---------------| | | |
| | | | |
|(53) SIP ACK | | | |
|------------------------------------------------->| |
| | | | |
| | | |(54) SIP ACK |
| | | |--------------->|
| | | | |
Figure 22: Endpoint Independent NAT with ICE
o On deciding to initiate a SIP voice session the SIP client 'L'
starts a local STUN client. The STUN client generates a standard
Allocate request as indicated in (1) from Figure 22 which also
highlights the source address and port combination for which the
client device wishes to obtain a mapping. The STUN request is
sent through the NAT towards the public internet.
o Allocate message (2) traverses the NAT and breaks out onto the
public internet towards the public STUN server. Note that the
source address of the Allocate request now represents the public
address and port from the public side of the NAT (L-NAT-PUB-1).
o The STUN server receives the Allocate request and processes
appropriately. This results in a successful Allocate response
being generated and returned (3). The message contains details of
the 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 relay address that can be used at the
STUN server (see 'Rel=STUN-PUB-2').
o The STUN 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 22. At this point the User Agent behind the
NAT has pairs of derived external reflexive and relayed
representations. The client would be free to gather any number of
external representations using any UNSAF[9] compliant protocol.
o The client now constructs a SIP INVITE message (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[16], as shown below in Figure 23
(*note - /* signifies line continuation):
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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
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 1.0 $L-PRIV-1.address $L-PRIV-1.port
a=candidate:$L1 2 UDP 1.0 $L-PRIV-2.address $L-PRIV-2.port
a=candidate:$L2 1 UDP 0.7 $L-NAT-PUB-1.address $L-NAT-PUB-1.port
a=candidate:$L2 2 UDP 0.7 $L-NAT-PUB-2.address $L-NAT-PUB-2.port
a=candidate:$L3 1 UDP 0.3 $STUN-PUB-2.address $STUN-PUB-2.port
a=candidate:$L3 2 UDP 0.3 $STUN-PUB-3.address $STUN-PUB-3.port
Figure 23: ICE SDP Offer
o The SDP has been constructed to include all the available
candidate pairs that have been assembled. The first candidate
pair (as identified by $L1) contain two local addresses that have
the highest priority (1.0). They are also encoded into the
connection (c=) and media (m=) lines of the SDP. The second
'candidate' address pair, as identified by the component-id,
contains the two NAT 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 (0.7) by the client.
The third and final candidate pair represents the relayed
addresses (as identified by $L3) obtained from the STUN server.
This pair has the lowest priority (0.3) and will be used as a last
resort.
o The SIP signalling 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 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.
Steps (11-18) involve client 'R' carrying out the same steps as
client 'L'. This involves obtaining local, 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 23 (*note - /* signifies line continuation):
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v=0
o=test 3890844516 3890842803 IN IP4 $R-PRIV-1
c=IN IP4 $R-PRIV-1.address
t=0 0
a=ice-pwd:$RPASS
m=audio $R-PRIV-1.port RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:$R-PRIV-2.port
a=candidate:$L1 1 UDP 1.0 $R-PRIV-1.address $R-PRIV-1.port
a=candidate:$L1 2 UDP 1.0 $R-PRIV-2.address $R-PRIV-2.port
a=candidate:$L2 1 UDP 0.7 $R-NAT-PUB-1.address $R-NAT-PUB-1.port
a=candidate:$L2 2 UDP 0.7 $R-NAT-PUB-2.address $R-NAT-PUB-2.port
a=candidate:$L3 1 UDP 0.3 $STUN-PUB-2.address $STUN-PUB-4.port
a=candidate:$L3 2 UDP 0.3 $STUN-PUB-3.address $STUN-PUB-5.port
Figure 24: ICE SDP Answer
o The two clients will now form candidate pairs and the transport
address check list as specified in ICE. Both 'L' and 'R' will
start the check list with the currently active component pair
(contained in the 'c=' and 'm=' of the SDP). As illustrated in
(23), client 'L' constructs a STUN Bind request in an attempt to
validate the connection address received in the SDP of the 200 OK
(20). As a private address was specified in the active address in
the SDP, the Stun Bind request fails to reach its destination
causing a bind failure. Client 'L' now attempts a STUN Bind
request for the first candidate pair in the returned SDP with the
highest priority (24). As can be seen from messages (24) to (29),
the STUN Bind request is successful and returns a positive outcome
for the connectivity check. Client 'L' is now free to steam media
to the candidate pair. Client 'R' also carries out the same set
of binding requests. It firstly (in parallel) tries to contact
the active address contained in the SDP (30). Client 'R' now
attempts a STUN Bind request for the first candidate pair in the
returned SDP with the highest priority (31). As can be seen from
messages (31) to (36), the STUN bind request is successful and
returns a positive outcome for the connectivity check. The
previously described check confirm on both sides that connectivity
can be achieved through appropriate candidates. 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
each candidate for use with RTCP. The connectivity checks for
part '2' of the candidate component are shown in 'L'(37-42) and
'R'(43-48).
o The candidates have now been fully verified (Valid status) and as
they are the highest priority, an updated offer (49-50) is now
sent from the offerer (client 'L') to the answerer (client 'R'
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representing the new active candidates. The new offer would look
as follows:
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
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 0.7 $L-NAT-PUB-1.address $L-NAT-PUB-1.port
a=candidate:$L2 2 UDP 0.7 $L-NAT-PUB-2.address $L-NAT-PUB-2.port
Figure 25: ICE SDP Updated Offer
o The resulting answer (51-52) for 'R' would look as follows:
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:$L2 1 UDP 0.7 $R-NAT-PUB-1.address $R-NAT-PUB-1.port
a=candidate:$L2 2 UDP 0.7 $R-NAT-PUB-2.address $R-NAT-PUB-2.port
Figure 26: ICE SDP Updated Answer
4.2.2. Port and Address Dependant NAT
4.2.2.1. STUN Failure
This section highlights that while STUN is the preferred mechanism
for traversal of NAT, it does not solve every case. The use of basic
STUN on its own will not guarantee traversal through every NAT type,
hence the recommendation that ICE is the preferred option.
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Client PORT/ADDRESS Dependant STUN [..]
NAT Server
| | | |
|(1) STUN Req | | |
|Src=L-PRIV-1 | | |
|Dest=STUN-PUB | | |
|----------------->| | |
| | | |
| |(2) STUN Req | |
| |Src=NAT-PUB-1 | |
| |Dest=STUN-PUB | |
| |----------------->| |
| | | |
| |(3) STUN Resp | |
| |<-----------------| |
| |Src=STUN-PUB | |
| |Dest=NAT-PUB-1 | |
| |Map=NAT-PUB-1 | |
| | | |
|(4) STUN 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 27: Port/Address Dependant NAT with STUN - Failure
The example in Figure 27 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 27 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 (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 [11]). 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 27 the STUN query produced a valid external
mapping or 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 does not come from the STUN server IP address and
optionally port, will be dropped at the NAT. Figure 27 shows the
media being dropped at the NAT after (7) and before (8). This
then leads to one way audio.
4.2.2.2. TURN Usage Solution
As identified in Section 4.2.2.1, STUN provides a useful tool kit for
the traversal of the majority of NATs but fails with Port/Address
Dependant NAT. This led to the development of the TURN usage
solution [12] which uses the STUN toolkit. It allows a client to
request a relayed address at the STUN server rather than a reflexive
representation. This then introduces a media relay in the path for
NAT traversal (as described in Section 3.2.3). The following example
explains how the TURN usage solves the previous failure when using
STUN to traverse a 'Port/Address Dependant' type NAT.
L Port/Address Dependant STUN [..]
NAT Server
| | | |
|(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 28: Port/Address Dependant NAT with TURN Usage - Success
o In this example, client 'L' issues a TURN allocate request(1) to
obtained a relay address at the STUN server. The request
traverses through the 'Port/Address Dependant' NAT and reaches the
STUN server (2). The STUN server generates an Allocate response
(3) that contains both a reflexive address (Map=NAT-PUB-1) of the
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client and also a relayed address (Rel=STUN-PUB-2). The relayed
address maps to an address mapping on the STUN server which is
bound to the public pin hole that has been opened on the NAT by
the Allocate request. This results in any traffic sent to the
STUN server relayed address (Rel=STUN-PUB-2) being forwarded to
the external representation of the pin hole created by the
Allocate request(NAT-PUB-1).
o The TURN derived address (STUN-PUB-2) arrives back at the
originating client(4) in an Allocate response. This address can
then be used in the SDP for the outgoing SIP INVITE request as
shown in the following example (note that the example also
includes client 'L' obtaining a second relay address for use in
the RTCP attribute (5-8)):
o
v=0
o=test 2890844342 2890842164 IN IP4 $L-PRIV-1
c=IN IP4 $STUN-PUB-2.address
t=0 0
m=audio $STUN-PUB-2.port RTP/AVP 0
a=rtcp:$STUN-PUB-3.port
o On receiving the INVITE request, the UAS is able to stream media
and RTCP to the relay address (STUN-PUB-2 and STUN-PUB-3) at the
STUN server. As shown in Figure 28 (between messages (12) and
(13), the media from the UAS is directed to the relayed address at
the STUN server. The STUN server then forwards the traffic to the
open pin holes in the Port/Address Dependant NAT (NAT-PUB-1 and
NAT-PUB-2). The media traffic is then able to traverse the 'Port/
Address Dependant' NAT and arrives back at client 'L'.
o The TURN usage of STUN on its own will work for 'Port/Address
Dependent' and other types of NAT mentioned in this specification
but should only be used as a last resort. The relaying of media
through an external entity is not an efficient mechanism for NAT
traversal and comes at a high processing cost.
4.2.2.3. ICE Solution
The previous two examples have highlighted the problem with using
core STUN usage for all forms of NAT traversal and a solution using
TURN usage for the Port/Address Dependant NAT case. As mentioned
previously in this document, the RECOMMENDED mechanism for traversing
all varieties of NAT is using ICE, as detailed in Section 3.2.4. ICE
makes use of core STUN, TURN usage and any other UNSAF[9] compliant
protocol to provide a list of prioritised addresses that can be used
for media traffic. Detailed examples of ICE can be found in
Section 4.2.1.2.1. These examples are associated with an 'Endpoint
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Independent' type NAT but can be applied to any NAT type variation,
including 'Port/Address Dependant' 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
specification[16]
[Editors Note: TODO - a detailed example will be included here which
includes promotion of a TURN relayed address to the active candidate
to traverse a 'Port/Address Dependent' type NAT.]
4.3. Address independent Port Restricted NAT --> Address independent
Port Restricted NAT traversal
[Editors Note: TODO - a detailed example will be included where User
A and B are both behind Address Independent NATs that have Port
restricted properties. This means that the stun-derived addresses
will work, but each side must send a 'suicide' or 'primer' STUN
packet that creates a permission in the NAT. So, the main thing to
show here is how the first packet from B to A will create a
permission in B's NAT but gets dropped at A. When A gets the answer
it starts its STUN checks and the packet from A to B creates a
permission in A's NAT and gets through B's NAT because of the
previously installed permission. This now triggers B to resend its
stun request which now works.]
4.4. Internal TURN Usage (Enterprise Deployment)
[Editors Note: TODO - a detailed example will be included for User A
and User B. User A is in an enterprise, which has a address and port
restricted NAT. User B is on the public internet. There is a TURN+
STUN server deployed INSIDE the enterprise NAT. The NAT has a static
set of ports forwarded to the internal TURN server (say, 100 ports).
The TURN server is configured with those ports. So, when user A
talks to the TURN server it gets an address and port on the *public*
side of the NAT, with a preconfigured port forwarding rule. Indeed,
the client is configured with two TURN servers. Both are physically
the same TURN server. However, when talking to one instance the
client gets the public address. When talking to the other instance
it gets a private address inside the NAT. The ice process ends up
selecting the public address given out by the TURN server usage.]
5. Intercepting Intermediary (B2BUA)
[Editors Note: TODO - a detailed example demonstrating how a B2BUA
can obtain STUN/TURN addresses for the purpose of allocating to
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Clients. This example shows how intermediaries can control the flow
of media without having to directly access SDP on the signalling
plane.
6. IPv4-IPv6 Transition
This section describes how IPv6-only SIP user agents can communicate
with IPv4-only SIP user agents.
6.1. IPv4-IPv6 Transition for SIP Signalling
IPv4-IPv6 translations at the SIP level usually take place at dual-
stack proxies that have both IPv4 and IPv6 DNS entries. Since this
translations do not involve NATs that are placed in the middle of two
SIP entities, they fall outside the scope of this document. A
detailed description of this type of translation can be found in [19]
6.2. IPv4-IPv6 Transition for Media
Figure 30 shows a network of IPv6 SIP user agents that has a relay
with a pool of public IPv4 addresses. The IPv6 SIP user agents of
this IPv6 network need to communicate with users on the IPv4
Internet. To do so, the IPv6 SIP user agents use TURN to obtain a
public IPv4 address from the relay. The mechanism that an IPv6 SIP
user agent follows to obtain a public IPv4 address from a relay using
TURN is the same as the one followed by a user agent with a private
IPv4 address to obtain a public IPv4 address. The example in
Figure 31 explains how to use TURN to obtain an IPv4 address and how
to use the ANAT semantics [17] of the SDP grouping framework [8] to
provide both IPv4 and IPv6 addresses for a particular media stream.
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+----------+
| / \ |
/SIP \
/Phone \
/ \
------------
IPv4 Network
192.0.2.0/8
+---------+
| |
----------------------| NAT |--------------------------
| |
+---------+
IPv6 Network
++
||
+-----++
| IPv6 |
| SIP |
| user |
| agent|
+------+
Figure 30: IPv6-IPv4 transition scenario
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IPv6 SIP TURN IPv4 SIP
User Agent Server User Agent
| | |
| (1) TURN Allocate | |
| src=[2001:DB8::1]:30000 | |
|------------------------------------>| |
| (2) TURN Resp | |
| map=192.0.2.2:25000 | |
| dest=[2001:DB8::1]:30000 | |
|<------------------------------------| |
| (3) SIP INVITE | |
|------------------------------------------------------->|
| (4) SIP 200 OK | |
|<-------------------------------------------------------|
| | |
|=====================================| |
|>>>>>>>>>> Outgoing Media >>>>>>>>>>>| |
|=====================================| |
| |==================|
| |>>>>>> Media >>>>>|
| |==================|
| | |
| |==================|
| |<<<<<< Media <<<<<|
| |==================|
|=====================================| |
|<<<<<<<<<< Outgoing Media <<<<<<<<<<<| |
|=====================================| |
| | |
| (5) SIP ACK | |
|------------------------------------------------------->|
| | |
Figure 31: IPv6-IPv4 translation with TURN
o The IPv6 SIP user agent obtains a TURN-derived IPv4 address by
issuing a TURN allocate request (1). The TURN server generates a
response that contains the public IPv4 address. This IPv4 address
maps to the IPv6 source address of the TURN allocate request,
which the IPv6 address of the SIP user agent. This results in any
traffic being sent to the IPv4 address provided by TURN server
(192.0.2.2:25000) will be redirected to the IPv6 address of the
SIP user agent ([2001:DB8::1]:30000).
o The TURN-derived address (192.0.2.2:25000) arrives back at the
originating user agent (2). This address can then be used in the
SDP for the outgoing SIP INVITE request. The user agent builds
two media lines, one with its IPv6 address and the other with the
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IPv4 address that was just obtained. The user agent groups both
media lines using the ANAT semantics as shown below (note that the
RTCP attribute in the IPv4 media line would have been obtained by
another TURN-derived address which is not shown in the call flow
for simplicity).
v=0
o=test 2890844342 2890842164 IN IP6 2001:DB8::1
t=0 0
a=group:ANAT 1 2
m=audio 20000 RTP/AVP 0
c=IN IP6 2001:DB8::1
a=mid:1
m=audio 25000 RTP/AVP 0
c=IN IP4 192.0.2.2
a=rtcp:25001
a=mid:2
o On receiving the INVITE request, the user agent server rejects the
IPv6 media line by setting its port to zero in the answer and
starts sending media to the IPv4 address in the offer. The IPv6
user agent sends media through the relay as well, as shown in
Figure 31.
7. ICE with RTP/TCP
[Editors Note: TODO - a detailed example will be included on using
ICE with RTP/TCP - as define in [18]
8. Acknowledgments
The authors would like to thank the members of the IETF SIPPING WG
for their comments and suggestions. Detailed comments were provided
by Francois Audet, kaiduan xie and Hans Persson.
9. References
9.1. Normative References
[1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[2] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
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"RTP: A Transport Protocol for Real-Time Applications",
RFC 1889, January 1996.
[3] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[4] Tsirtsis, G. and P. Srisuresh, "Network Address Translation -
Protocol Translation (NAT-PT)", RFC 2766, February 2000.
[5] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[6] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing",
RFC 3581, August 2003.
[7] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP)
Extension Header Field for Registering Non-Adjacent Contacts",
RFC 3327, December 2002.
[8] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne,
"Grouping of Media Lines in the Session Description Protocol
(SDP)", RFC 3388, December 2002.
[9] Daigle, L. and IAB, "IAB Considerations for UNilateral Self-
Address Fixing (UNSAF) Across Network Address Translation",
RFC 3424, November 2002.
[10] Rosenberg, J., "Simple Traversal of UDP Through Network Address
Translators (NAT) (STUN)", draft-ietf-behave-rfc3489bis-03
(work in progress), March 2006.
[11] Audet, F. and C. Jennings, "NAT Behavioral Requirements for
Unicast UDP", draft-ietf-behave-nat-udp-07 (work in progress),
June 2006.
[12] Rosenberg, J., "Obtaining Relay Addresses from Simple Traversal
of UDP Through NAT (STUN)", draft-ietf-behave-turn-00 (work in
progress), March 2006.
[13] Jennings, C. and A. Hawrylyshen, "SIP Conventions for UAs with
Outbound Only Connections", draft-jennings-sipping-outbound-01
(work in progress), February 2005.
[14] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-09 (work in progress), June 2006.
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[15] Wing, D., "Symmetric RTP and RTCP Considered Helpful",
draft-wing-mmusic-symmetric-rtprtcp-01 (work in progress),
October 2004.
[16] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols", draft-ietf-mmusic-ice-08 (work in
progress), March 2006.
[17] Camarillo, G., "The Alternative Network Address Types Semantics
(ANAT) for theSession Description Protocol (SDP) Grouping
Framework", draft-ietf-mmusic-anat-02 (work in progress),
October 2004.
[18] Rosenberg, J., "TCP Candidates with Interactive Connectivity
Establishment (ICE)", draft-ietf-mmusic-ice-tcp-00 (work in
progress), March 2006.
9.2. Informative References
[19] Camarillo, G., "IPv6 Transcition in the Session Initiation
Protocol (SIP)", draft-camarillo-sipping-v6-transition-00 (work
in progress), February 2005.
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Authors' Addresses
Chris Boulton
Ubiquity Software Corporation
Eastern Business Park
St Mellons
Cardiff, South Wales CF3 5EA
Email: cboulton@ubiquitysoftware.com
Jonathan Rosenberg
Cisco Systems
600 Lanidex Plaza
Parsippany, NJ 07054
Email: jdrosen@cisco.com
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: Gonzalo.Camarillo@ericsson.com
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