Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions.
draft-ietf-stox-media-02
The information below is for an old version of the document.
| Document | Type |
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|---|---|---|---|
| Authors | Peter Saint-Andre , Saúl Ibarra Corretgé , Emil Ivov | ||
| Last updated | 2013-12-12 | ||
| Replaces | draft-saintandre-sip-xmpp-media | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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| Additional resources | Mailing list discussion | ||
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draft-ietf-stox-media-02
Network Working Group P. Saint-Andre
Internet-Draft Cisco Systems, Inc.
Intended status: Standards Track S. Ibarra
Expires: June 15, 2014 AG Projects
E. Ivov
Jitsi
December 12, 2013
Interworking between the Session Initiation Protocol (SIP) and the
Extensible Messaging and Presence Protocol (XMPP): Media Sessions.
draft-ietf-stox-media-02
Abstract
This document defines a bi-directional protocol mapping for use by
gateways that enable the exchange of media signalling messages
between systems that implement the Jingle extensions to the
Extensible Messaging and Presence Protocol (XMPP) and those that
implement the Session Initiation Protocol (SIP).
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on June 15, 2014.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
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to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Compatibility with Offer/Answer model . . . . . . . . . . . . 4
4. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
5. Syntax Mappings . . . . . . . . . . . . . . . . . . . . . . . 5
6. Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . 9
7. Early Media . . . . . . . . . . . . . . . . . . . . . . . . . 10
8. Detecting Endless Loops . . . . . . . . . . . . . . . . . . . 11
9. SDP Format-Specific Parameters . . . . . . . . . . . . . . . . 11
10. Dialog Forking . . . . . . . . . . . . . . . . . . . . . . . . 12
11. Sample Scenarios . . . . . . . . . . . . . . . . . . . . . . . 13
12. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 18
13. Security Considerations . . . . . . . . . . . . . . . . . . . 18
14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
14.1. Normative References . . . . . . . . . . . . . . . . . . 18
14.2. Informative References . . . . . . . . . . . . . . . . . 20
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20
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1. Introduction
The Session Initiation Protocol [RFC3261] is a widely-deployed
technology for the management of media sessions (such as voice and
video calls) over the Internet. SIP itself provides a signalling
channel (sometimes via the User Datagram Protocol [RFC0768]), over
which two or more parties can exchange messages for the purpose of
negotiating a media session that uses a dedicated media channel such
as the Real-time Transport Protocol [RFC3550].
The Extensible Messaging and Presence Protocol (XMPP) [RFC6120] also
provides a signalling channel, typically via the Transmission Control
Protocol [RFC0793]. Given the significant differences between XMPP
and SIP, it is difficult to combine the two technologies in a single
user agent. Therefore, developers wishing to add media session
capabilities to XMPP clients have defined an XMPP-specific
negotiation protocol called Jingle [XEP-0166].
However, Jingle was designed to easily map to SIP for communication
through gateways or other transformation mechanisms. Therefore,
consistent with existing specifications for mapping between SIP and
XMPP (see [I-D.ietf-stox-core] and other related specifications),
this document describes a bidirectional protocol mapping for use by
gateways that enable the exchange of media signalling messages
between systems that implement SIP and those that implement the XMPP
Jingle extensions.
It is important to note that SIP and Jingle sessions can be
gateway-ed in a rather simple fashion if all media was always routed
and potentially even transcoded in through a gateway. This
specification aims to define a mapping that goes beyond the above and
allows gateways to (wherever possible) only intervene at the
signalling level, letting user agents exchange media in an end-to-end
manner. Such gateways would likely focus on handling handling RTP
session establishment and control within the context of what users
would perceive as "calls". This document is hence primarily dealing
with calling scenarios as opposed to generic media sessions with SIP.
The discussion venue for this document is the mailing list of the
STOX WG; visit https://www.ietf.org/mailman/listinfo/stox for
subscription information and discussion archives.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
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[RFC2119].
A number of technical terms used here are defined in [RFC3261],
[RFC6120], [XEP-0166], and [XEP-0167]. The term "JID" is short for
"Jabber Identifier".
3. Compatibility with Offer/Answer model
Even if Jingle semantics have many similarities with those used in
SIP, there are some use cases that cannot be handled in exactly the
same way due to the Offer/Answer model used in SIP in conjunction
with SDP.
More specifically, mapping SIP and SDP Offer/Answer to XMPP is often
complicated due to the difference in how each handles backward
compatibility. Jingle, as most other XMPP extensions, relies heavily
on the protocol's advanced service discovery [XEP-0030] mechanisms.
In other words, XMPP entities are able to verify the capabilities of
their intended peer before actually attempting to establish a session
with it.
SDP Offer/Answer on the other hand uses a least common denominator
approach where every SDP offer has to be understandable by legacy
endpoints. Newer, unsupported aspects in this offer can therefore
only appear as optional or their use be limited to subsequent Offer/
Answer exchanges, once their support has been confirmed.
Use of "trickle ICE" (see [I-D.ietf-mmusic-trickle-ice] and
[I-D.ivov-mmusic-trickle-ice-sip]) is one example where the issue
occurs. SIP endpoints need to always behave as vanilla ICE agents
when sending their first offer and make sure they gather all
candidates before sending a SIP INVITE. This is necessary because
otherwise ICE agents with no support for trickle can prematurely
declare failure. Jingle endpoints, on the other hand can verify
support for trickle ICE prior to engaging in a session and adapt
their behaviour accordingly.
In order to work around such issues, [XEP-0176] defines an Offer/
Answer support mode through the "urn:ietf:rfc:3264" feature tag. It
indicates that a specific XMPP entity can only be contacted through
the use of Offer/Answer semantics. Implementations conforming to
this specification MUST support Offer/Answer model with Jingle. Note
that such endpoints are not required to actually declare support for
this tag because this would mean that they too would only be
reachable through Offer/Answer semantics.
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4. Overview
As mentioned, Jingle was designed in part to enable straightforward
protocol mapping between XMPP and SIP. However, given the
significantly different technology assumptions underlying XMPP and
SIP, Jingle is naturally different from SIP in several important
respects:
o Base SIP messages and headers use a plaintext format similar in
some ways to the Hypertext Transport Protocol [RFC2616], whereas
Jingle messages are pure XML. Mappings between SIP headers and
Jingle message syntax are provided below.
o The SIP payloads defining session semantics use the Session
Description Protocol [RFC4566], whereas the equivalent Jingle
payloads are defined as XML child elements of the Jingle
<content/> element. However, the Jingle specifications defining
such child elements specify mappings to SDP for all Jingle syntax,
making the mapping relatively straightforward.
o The SIP signalling channel has historically often been transported
over UDP, whereas the signalling channel for Jingle is XMPP over
TCP. Mapping between the transport layers typically happens
within a gateway using techniques below the application level, and
therefore is not addressed in this specification.
5. Syntax Mappings
5.1. Generic Jingle Syntax
Jingle is designed in a modular fashion, so that session description
data is generally carried in a payload within the generic Jingle
elements, i.e., the <jingle/> element and its <content/> child. The
following example illustrates this structure, where the XMPP stanza
is a request to initiate an audio session using RTP over a raw UDP
transport.
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<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='ne91v36s'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-initiate'
initiator='romeo@example.net/v3rsch1kk3l1jk'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'
senders='both'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='96' name='speex' clockrate='16000'/>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
<payload-type channels='2'
clockrate='16000'
id='103'
name='L16'/>
<payload-type id='98' name='x-ISAC' clockrate='8000'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='10.1.1.104' port='13540' generation='0'/>
</transport>
</content>
</jingle>
</iq>
In the foregoing example, the syntax and semantics of the <jingle/>
and <content/> elements are defined in [XEP-0166], the syntax and
semantics of the <description/> element are defined in [XEP-0167],
and the syntax and semantics of the <transport/> element are defined
in [XEP-0177]. Other <description/> elements are defined in
specifications for the appropriate application types (see for example
[XEP-0167]) and other <transport/> elements are defined in the
specifications for appropriate transport methods (see for example
[XEP-0176], which defines an XMPP profile of [RFC5245]).
At the core Jingle layer, the following mappings are defined.
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+--------------------------------+--------------------------------+
| Jingle | SIP |
+--------------------------------+--------------------------------+
| <jingle/> 'action' | [ see next table ] |
+--------------------------------+--------------------------------+
| <jingle/> 'initiator' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <jingle/> 'responder' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <jingle/> 'sid' | local-part of Dialog ID |
+--------------------------------+--------------------------------+
| local-part of 'initiator' | <username> in SDP o= line |
+--------------------------------+--------------------------------+
| <content/> 'creator' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <content/> 'name' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <content/> 'profile' | <proto> in SDP m= line |
+--------------------------------+--------------------------------+
| <content/> 'senders' value of | a= line of sendrecv, recvonly, |
| both, initiator, or responder | or sendonly |
+--------------------------------+--------------------------------+
The 'senders' attribute is optional in Jingle, thus in case it's
absent it's RECOMMENDED that the direction value is considered as
'sendrecv'.
The 'action' attribute of the <jingle/> element has nine allowable
values. In general they should be mapped as shown in the following
table, with some exceptions as described herein.
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+-------------------+-----------------+
| Jingle Action | SIP Method |
+-------------------+-----------------+
| content-accept | INVITE response |
| | (1xx or 2xx) |
+-------------------+-----------------+
| content-add | INVITE request |
+-------------------+-----------------+
| content-modify | INVITE request |
+-------------------+-----------------+
| content-remove | INVITE request |
+-------------------+-----------------+
| session-accept | INVITE response |
| | (1xx or 2xx) |
+-------------------+-----------------+
| session-info | [varies] |
+-------------------+-----------------+
| session-initiate | INVITE request |
+-------------------+-----------------+
| session-terminate | BYE |
+-------------------+-----------------+
| transport-info | unnused |
+-------------------+-----------------+
5.2. Application Formats
Jingle application formats for audio and video exchange via RTP are
specified in [XEP-0167]. These application formats effectively maps
to the "RTP/AVP" profile specified in [RFC3551] and the "RTP/SAVP"
profile specified in [RFC3711], where the media types are "audio" and
"video", and the specific mappings to SDP syntax are provided in
[XEP-0167].
As stated in [XEP-0167] future versions of this specification might
define how to use other RTP profiles such as "RTP/AVPF" and "RTP/
SAVPF" as defined in RFC4585 and RFC5124 respectively.
5.3. Raw UDP Transport Method
A basic Jingle transport method for exchanging media over UDP is
specified in [XEP-0177]. This transport method involves the
negotiation of an IP address and port only. It does not provide NAT
traversal, effectively leaving the task to intermediary entries. The
Jingle 'ip' attribute maps to the connection-address parameter of the
SDP c= line and the 'port' attribute maps to the port parameter of
the SDP m= line. Use of SIP without ICE would generally map to use
of Raw UDP on the XMPP side of a session.
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5.4. ICE-UDP Transport Method
A more advanced Jingle transport method for exchanging media over UDP
is specified in [XEP-0176]. Under ideal conditions this transport
method provides NAT traversal by following the Interactive
Connectivity Exchange methodology specified in [RFC5245].
The relevant SDP mappings are provided in [XEP-0176], however there
are a few syntax incompatibilities which need to be addressed by
gateways conforming to this specification:
o The 'foundation' attribute is defined as a number in Jingle
(unsigned byte) whereas ICE [RFC5245] defines it as a string,
which can contain letters, digits and the '+' and '/' symbols.
Gateway applications MUST therefore convert ICE originating
foundations into integer numbers and they MUST guarantee that such
a conversion preserves foundation uniqueness. The exact mechanism
for the conversion is undefined.
o Jingle defines a 'generation' attribute which is used to determine
if an ICE restart is required. Such attribute has no counterpart
in SIP as ICE restarts are detected by detecting a change in the
ICE ufrag and password. Gateways MUST therefore increase the
generation number when they detect such changes.
o The 'id' attribute defined by Jingle has no SIP counterpart thus
applications are free to choose means to generate unique
identifiers across the different candidates of an ICE generation.
o The 'network' attribute defined by Jingle has no counterpart in
SIP and SHOULD be ignored.
6. Call Hold
[RFC3264] stipulates that streams are placed on hold by setting their
direction to "sendonly". A session is placed on hold by doing this
for all the streams it contains. The same semantics are also
supported by Jingle through the "senders" element and its "initiator"
and "responder" values.
[example to follow]
In addition to these semantics however Jingle also defines a more
concise way for achieving the same, which consists in sending a
"hold" command within a "session-info" action:
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<iq from='juliet@capulet.lit/balcony'
id='xv39z423'
to='romeo@montague.lit/orchard'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-info'
initiator='romeo@montague.lit/orchard'
sid='a73sjjvkla37jfea'>
<hold xmlns='urn:xmpp:jingle:apps:rtp:info:1'/>
</jingle>
</iq>
Gateways that receive a "hold" command from their Jingle side MUST
generate a new offer on their SIP side, placing all streams in a
"sendonly" state.
When relaying offers from SIP to XMPP however, gateways are not
required to translate "sendonly" attributes into a "hold" command as
this would not always be possible (e.g. when not all streams have the
same direction). Additionally such conversions might introduce
complications in case further offers placing a session of hold also
contain other session modifications.
[[OPEN ISSUE: do we need to mention double hold here? That is, when
you put me on hold after I did it first. Direction would then be
"inactive".]]
7. Early Media
[RFC3959] and [RFC3960] describe a number of scenarios relying on
"early media". While similar attempts have also been made for XMPP
[XEP-0269] support for early media is not currently widely supported
in Jingle implementations. Therefore, gateways SHOULD NOT forward
SDP answers from SIP to Jingle until a final response has been
received, except in cases where the gateway is in a position to
confirm specific support for early media by the endpoint (one
approach to such support can be found in [XEP-0269] but it has not
yet been standardized).
Gateways MUST however store early media SDP answers when they are
sent inside a reliable provisional response. In such cases, a
subsequent final response may follow without an actual answer and the
one from the provisional response will need to be forwarded to the
Jingle endpoint.
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8. Detecting Endless Loops
[RFC3261] defines a "Max-Forwards" header that allows intermediate
entities such as SIP proxies to detect and prevent loops from
occurring. The specifics of XMPP make such a prevention mechanism
unnecessary for XMPP-only environments. With the introduction of
SIP-to-XMPP gatewaying however, it would be possible for loops to
occur where messages are being repeatedly forwarded from XMPP to SIP
to XMPP to SIP, etc.
To compensate for the lack of a "Max-Forwards" header in SIP,
gateways MUST therefore keep track of all SIP transactions and Jingle
sessions that they are currently serving and they MUST block re-
entrant messages.
[[OPEN ISSUE: In order for this to work, we need a consistent way of
translating dialog IDs into Jingle sessions, and vice versa, so that
the following can be verified: jingleSessID ==
toJingleSessID(toSipCallID( jingleSessID )). We need to mention
mention spirals here as well. Alice could call Bob, but Bob forwards
his call to Romeo. A spiral on the SIP side could end up becoming a
loop if the gateway is in between.]]
9. SDP Format-Specific Parameters
[RFC4566] defines "a=fmtp" attributes for the transmission of format
specific parameters as a single transparent string. Such strings can
be used to convey either a single value or a sequence of parameters,
separated by semi-colons, comas or whatever delimiters are chosen by
a particular payload type specification.
[XEP-0177] on the other hand defines a "<format/>" element as
follows:
<format name="paramName" value="paramValue"/>
A sequence of parameters is thus transmitted as an array of
distinctive name/value couples.
These differences make it impossible to devise a generic mechanism
that accurately translates format parameters from Jingle to SDP
without the specifics of the payload being known to the gateway.
This specification therefore makes the following recommendations for
a best-effort attempt at translation:
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1. Gateways that are aware of the formats in use SHOULD parse all
format parameters and generate "<format/>" arrays and "a=fmtp"
values accordingly.
2. When translating Jingle to SIP, gateways that have no explicit
support for the formats that are being negotiated, SHOULD convert
the list of "<format/>" elements into a single string, containing
a sequence of "name=value" pairs, separated by a semi-colon and a
space (i.e. "; ").
3. When translating SIP to Jingle, gateways that have no explicit
support for the formats that are being negotiated, SHOULD
tokenize the "a=fmtp" format string using the following list of
delimiters: ";", "; ", ",", ", ". The resulting tokens should
then be parsed as "name=value" pairs. If this process does
actually yield any such pairs, they SHOULD be used for generating
the respective "<format/>" elements. If some of the tokens
cannot be parsed into a "name=value" pair, or in case the format
string couldn't be tokenized with the above delimiters, the
remaining brute strings SHOULD be used as a value for the "name"
attribute of the "<format/>" element and their corresponding
"value" element SHOULD be left empty.
[[OPEN ISSUE: we need to add examples for these transformations.]]
10. Dialog Forking
[RFC3261] defines semantics for dialog forking. Such semantics have
not been defined for Jingle and need to be hidden from XMPP
endpoints.
To achieve this SIP-to-XMPP MUST NOT forward more than one
provisional response on their Jingle side. Typically they would do
so only for the first provisional response they receive and ignore
the rest. This provisional response SHOULD be forwarded as
originating from a bare Jabber ID (JID) corresponding to the AOR URI
found in the "From" header of the SIP provisional response. The
gateway MUST NOT attempt to translate GRUUs into full JIDs because it
cannot know at this stage, which of the dialogs established by these
provisional responses will be used for the actual session.
Likewise, gateways conforming to this specification MUST NOT forward
more than a single final response received through SIP to the Jingle
side. The gateway SHOULD terminate the SIP sessions whose received
final response wasn't forwarded to the Jingle side.
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11. Sample Scenarios
The following sections provide sample scenarios (or "call flows")
that illustrate the principles of interworking from Jingle to SIP.
These scenarios are not exhaustive.
11.1. Basic Voice Chat
The protocol flow for a basic voice chat for which an XMPP user
(juliet@example.com) is the initiator and a SIP user
(romeo@example.net) is the responder. The voice chat is consummated
through a gateway. To simplify the example, the transport method
negotiated is "raw user datagram protocol" as specified in
[XEP-0177].
INITIATOR ...XMPP... GATEWAY ...SIP... RESPONDER
| | |
| session-initiate | |
|----------------------->| |
| IQ-result (ack) | |
|<-----------------------| |
| | INVITE |
| |---------------------->|
| | 180 Ringing |
| |<----------------------|
| session-info (ringing) | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | 200 OK |
| |<----------------------|
| session-accept | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | ACK |
| |---------------------->|
| MEDIA SESSION |
|<==============================================>|
| | BYE |
| |<----------------------|
| session-terminate | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | 200 OK |
| |---------------------->|
| | |
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The packet flow is as follows.
First the XMPP user sends a Jingle session-initiation request to the
SIP user.
<iq from='juliet@example.com/t3hr0zny'
id='hu2s61f4'
from='romeo@example.net/v3rsch1kk3l1jk'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-initiate'
initiator='juliet@example.com/t3hr0zny'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='96' name='speex' clockrate='16000'/>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='192.0.2.101' port='49172' generation='0'/>
</transport>
</content>
</jingle>
</iq>
The gateway returns an XMPP IQ-result to the initiator on behalf of
the responder.
<iq from='juliet@example.com/t3hr0zny'
id='hu2s61f4'
to='romeo@example.net/v3rsch1kk3l1jk'
type='result'/>
The gateway transforms the Jingle session-initiate action into a SIP
INVITE.
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INVITE sip:romeo@example.net SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:juliet@client.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 184
v=0
o=alice 2890844526 2890844526 IN IP4 client.example.com
s=-
c=IN IP4 192.0.2.101
t=0 0
m=audio 49172 RTP/AVP 18 96 97
a=rtpmap:96 sppex/16000
a=rtpmap:97 speex/8000
a=rtpmap:18 G729
The responder returns a SIP 180 Ringing message.
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9;\
received=192.0.2.101
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>;tag=v3rsch1kk3l1jk
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:romeo@client.example.net;transport=tcp>
Content-Length: 0
The gateway transforms the ringing message into XMPP syntax.
<iq from='romeo@montague.net/v3rsch1kk3l1jk'
id='ol3ba71g'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-info'
initiator='juliet@example.com/t3hr0zny'
sid='a73sjjvkla37jfea'>
<ringing xmlns='urn:xmpp:jingle:apps:rtp:info:1'/>
</jingle>
</iq>
The initiator returns an IQ-result acknowledging receipt of the
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ringing message, which is used only by the gateway and not
transformed into SIP syntax.
<iq from='juliet@example.com/t3hr0zny'
id='ol3ba71g'
to='romeo@example.net/v3rsch1kk3l1jk'
type='result'/>
The responder sends a SIP 200 OK to the initiator.
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9;\
received=192.0.2.101
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>;tag=v3rsch1kk3l1jk
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:romeo@client.example.net;transport=tcp>
Content-Type: application/sdp
Content-Length: 147
v=0
o=romeo 2890844527 2890844527 IN IP4 client.example.net
s=-
c=IN IP4 192.0.2.201
t=0 0
m=audio 3456 RTP/AVP 97
a=rtpmap:97 speex/8000
The gateway transforms the 200 OK into a Jingle session-accept
action.
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<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='pd1bf839'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-accept'
initiator='juliet@example.com/t3hr0zny'
responder='romeo@example.net/v3rsch1kk3l1jk'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='97' name='speex' clockrate='8000'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='192.0.2.101' port='49172' generation='0'/>
</transport>
</content>
</jingle>
</iq>
If the payload types and transport candidate can be successfully used
by both parties, then the initiator acknowledges the session-accept
action.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='pd1bf839'
to='juliet@example.com/t3hr0zny'
type='result'/>
The parties now begin to exchange media. In this case they would
exchange audio using the Speex codec at a clockrate of 8000 since
that is the highest-priority codec for the responder (as determined
by the XML order of the <payloadtype/> children).
The parties may continue the session as long as desired.
Eventually, one of the parties (in this case the responder)
terminates the session.
BYE sip:juliet@client.example.com SIP/2.0
Via: SIP/2.0/TCP client.example.net:5060;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Romeo Montague <sip:romeo@example.net>;tag=8321234356
To: Juliet Capulet <sip:juliet@example.com>;tag=9fxced76sl
Call-ID: 3848276298220188511@example.com
CSeq: 1 BYE
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Content-Length: 0
The gateway transforms the SIP BYE into XMPP syntax.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='rv301b47'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-terminate'
initiator='juliet@example.com/t3hr0zny'
reasoncode='no-error'
sid='a73sjjvkla37jfea'/>
</iq>
The initiator returns an IQ-result acknowledging receipt of the
session termination, which is used only by the gateway and not
transformed into SIP syntax.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='rv301b47'
to='juliet@example.com/t3hr0zny'
type='result'/>
12. IANA Considerations
This document has no actions for the IANA.
13. Security Considerations
Detailed security considerations for session management are given for
SIP in [RFC3261] and for XMPP in [XEP-0166] (see also [RFC6120]).
The security considerations provided in [I-D.ietf-stox-core] also
apply.
14. References
14.1. Normative References
[I-D.ietf-stox-core]
Saint-Andre, P., Houri, A., and J. Hildebrand,
"Interworking between the Session Initiation Protocol
(SIP) and the Extensible Messaging and Presence Protocol
(XMPP): Core", draft-ietf-stox-core-08 (work in progress),
December 2013.
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[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, March 2011.
[XEP-0030]
Hildebrand, J., Eatmon, R., and P. Saint-Andre, "Service
Discovery", XSF XEP 0030, June 2008.
[XEP-0166]
Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
S., and J. Hildebrand, "Jingle", XSF XEP 0166, June 2007.
[XEP-0167]
Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen,
"Jingle RTP Sessions", XSF XEP 0167, February 2009.
[XEP-0176]
Beda, J., Ludwig, S., Saint-Andre, P., Hildebrand, J., and
S. Egan, "Jingle ICE-UDP Transport Method", XSF XEP 0176,
February 2009.
[XEP-0177]
Beda, J., Saint-Andre, P., Ludwig, S., Hildebrand, J., and
S. Egan, "Jingle Raw UDP Transport", XSF XEP 0177,
February 2009.
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14.2. Informative References
[I-D.ietf-mmusic-trickle-ice]
Ivov, E., Rescorla, E., and J. Uberti, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol",
draft-ietf-mmusic-trickle-ice-00 (work in progress),
October 2013.
[I-D.ivov-mmusic-trickle-ice-sip]
Ivov, E., Marocco, E., and C. Holmberg, "A Session
Initiation Protocol (SIP) usage for Trickle ICE",
draft-ivov-mmusic-trickle-ice-sip-01 (work in progress),
October 2013.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3959] Camarillo, G., "The Early Session Disposition Type for the
Session Initiation Protocol (SIP)", RFC 3959,
December 2004.
[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
Tone Generation in the Session Initiation Protocol (SIP)",
RFC 3960, December 2004.
[XEP-0269]
Cionoiu, D. and P. Saint-Andre, "Jingle Early Media", XSF
XEP 0269, May 2009.
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Authors' Addresses
Peter Saint-Andre
Cisco Systems, Inc.
1899 Wynkoop Street, Suite 600
Denver, CO 80202
USA
Phone: +1-303-308-3282
Email: psaintan@cisco.com
Saul Ibarra Corretge
AG Projects
Dr. Leijdsstraat 92
Haarlem 2021RK
The Netherlands
Email: saul@ag-projects.com
Emil Ivov
Jitsi
Strasbourg 67000
France
Phone: +33-177-624-330
Email: emcho@jitsi.org
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