Internet DRAFT - draft-ietf-stox-media
draft-ietf-stox-media
Network Working Group P. Saint-Andre
Internet-Draft &yet
Intended status: Standards Track S. Ibarra
Expires: January 21, 2016 AG Projects
E. Ivov
Jitsi
July 20, 2015
Interworking between the Session Initiation Protocol (SIP) and the
Extensible Messaging and Presence Protocol (XMPP): Media Sessions
draft-ietf-stox-media-07
Abstract
This document defines a bidirectional protocol mapping for use by
gateways that enable the exchange of media signaling messages between
systems that implement the Session Initiation Protocol (SIP) and
systems that implement the Jingle extensions to the Extensible
Messaging and Presence Protocol (XMPP).
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
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This Internet-Draft will expire on January 21, 2016.
Copyright Notice
Copyright (c) 2015 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
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to this document. Code Components extracted from this document must
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Intended Audience . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Compatibility with Offer/Answer Model and Interactive
Connectivity Establishment . . . . . . . . . . . . . . . . . 5
5. Syntax Mappings . . . . . . . . . . . . . . . . . . . . . . . 6
5.1. Generic Jingle Syntax . . . . . . . . . . . . . . . . . . 6
5.2. Application Formats . . . . . . . . . . . . . . . . . . . 10
5.3. Raw UDP Transport Method . . . . . . . . . . . . . . . . 10
5.4. ICE-UDP Transport Method . . . . . . . . . . . . . . . . 10
6. Transport Fallback . . . . . . . . . . . . . . . . . . . . . 11
7. Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . 12
8. Early Media . . . . . . . . . . . . . . . . . . . . . . . . . 13
9. Detecting Endless Loops . . . . . . . . . . . . . . . . . . . 14
10. SDP Format-Specific Parameters . . . . . . . . . . . . . . . 14
11. Dialog Forking . . . . . . . . . . . . . . . . . . . . . . . 16
12. Sample Call Flow . . . . . . . . . . . . . . . . . . . . . . 17
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
14. Security Considerations . . . . . . . . . . . . . . . . . . . 24
15. References . . . . . . . . . . . . . . . . . . . . . . . . . 24
15.1. Normative References . . . . . . . . . . . . . . . . . . 24
15.2. Informative References . . . . . . . . . . . . . . . . . 25
Appendix A. Acknowledgements . . . . . . . . . . . . . . . . . . 28
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 28
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 signaling
channel via TCP [RFC0793] or UDP [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 (RTP) [RFC3550]. The Extensible Messaging and
Presence Protocol (XMPP) [RFC6120] also provides a signaling channel,
typically via TCP (although bindings for HTTP [XEP-0124] and
WebSocket [RFC7395] also exist).
Given the significant differences between XMPP and SIP, traditionally
it was difficult to combine the two technologies in a single user
agent (although nowadays such implementations are not uncommon, as
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described in [RFC7081]). Thus in 2005 some developers wishing to add
media session capabilities to XMPP clients defined a set of XMPP-
specific session negotiation protocol extensions called Jingle (see
especially [XEP-0166], [XEP-0167], and [XEP-0176]).
Jingle was designed to easily map to SIP for communication through
gateways or other transformation mechanisms. Nevertheless, given the
significantly different technology assumptions underlying XMPP and
SIP, Jingle is 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 [RFC7230], whereas
Jingle messages are pure XML. Mappings between SIP headers and
Jingle message syntax are provided below.
o SIP payloads for session semantics use the Session Description
Protocol [RFC4566], whereas the equivalent Jingle payloads use 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 SIP messages have historically often been transported over UDP,
whereas the signaling 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.
Consistent with existing specifications for mapping between SIP and
XMPP (see [RFC7247]), this document describes a bidirectional
protocol mapping for use by gateways that enable the exchange of
media signaling messages between systems that implement SIP and
systems that implement the XMPP Jingle extensions.
It is important to note that SIP and Jingle sessions could be
gatewayed in a very simple way if all media were always routed and
potentially even transcoded through the same gateway used for
signaling. By contrast, this specification defines a mapping that
allows gateways to only intervene at the signaling level, thus
letting user agents exchange media in an end-to-end or peer-to-peer
manner without intervention by a specialized gateway (naturally, a
media relay that supports TURN [RFC5766] might be used). Such
signaling-only gateways focus on handling 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 or specialized sessions for
functionality such as file transfer (see [RFC5547] and [XEP-0234]).
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2. Intended Audience
The documents in this series are intended for use by software
developers who have an existing system based on one of these
technologies (e.g., SIP), and would like to enable communication from
that existing system to systems based on the other technology (e.g.,
XMPP). We assume that readers are familiar with the core
specifications for both SIP [RFC3261] and XMPP [RFC6120], with the
base document for this series [RFC7247], and with the following
media-related specifications:
o RTP Profile for Audio and Video Conferences with Minimal Control
[RFC3551]
o The Secure Real-time Transport Protocol (SRTP) [RFC3711]
o SDP: Session Description Protocol [RFC4566]
o Interactive Connectivity Establishment (ICE): A Protocol for
Network Address Translator (NAT) Traversal for Offer/Answer
Protocols [RFC5245]
o Jingle [XEP-0166]
o Jingle RTP Sessions [XEP-0167]
o Jingle ICE-UDP Transport Method [XEP-0176]
o Jingle Raw UDP Transport Method [XEP-0177]
3. Terminology
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".
In flow diagrams, SIP traffic is shown using arrows such as "***>"
whereas XMPP traffic is shown using arrows such as "...>".
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
[RFC2119].
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4. Compatibility with Offer/Answer Model and Interactive Connectivity
Establishment
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 XMPP extension for service discovery [XEP-0030], which implies
that XMPP entities are able to verify the capabilities of their
intended peer before attempting to establish a session with it.
SDP Offer/Answer, on the other hand, uses a "least common
denominator" approach where every SDP offer needs to be
comprehensible by legacy endpoints. Newer, unsupported aspects in
this offer can therefore only appear as optional, or their use needs
to be limited to subsequent Offer/Answer exchanges once their support
has been confirmed.
In particular, many older SIP endpoints do not support Interactive
Connectivity Establishmen (ICE) [RFC5245]. A signaling gateway from
Jingle to SIP has two primary alternatives for dealing with such
endpoints on the SIP side:
o Require the use of ICE and otherwise fail the call by including
the "Require: ice" SIP option tag [RFC5768] in the SIP INVITE that
it sends on behalf of the Jingle initiator.
o Send an initial SIP INVITE for an ICE connection and, if the SIP
endpoint indicates that it cannot handle ICE, send a re-INVITE for
a non-ICE connection to the SIP endpoint and a Jingle transport-
replace for a Raw UDP connection to the Jingle endpoint. (This
will introduce a potentialy large delay and might not result in a
much higher percentage of calls succeeding unless the signaling
gateway also offers a TURN [RFC5766] service for NAT traversal.)
See Section 6 for further discussion.
Use of "Trickle ICE" is one significant example where this issue
occurs. From the beginning, Jingle supported the trickling of
candidates (via Jingle messages of type 'transport-info'), and only
years later was this behavior generalized
[I-D.ietf-mmusic-trickle-ice] and then ported to SIP
[I-D.ietf-mmusic-trickle-ice-sip]. Therefore SIP endpoints need to
always behave like so-called "vanilla ICE" agents when sending their
first offer and make sure they gather all candidates before sending a
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SIP INVITE. This is necessary because otherwise ICE agents with no
support for trickling of candidates can prematurely declare failure.
Jingle endpoints, on the other hand, can verify support for trickling
of candidates prior to engaging in a session and adapt their behavior
accordingly (and, as noted, trickling of candidates is standard
operating procedure in Jingle).
In order to work around this disparity in relation to communication
of transport candidates, the Jingle RTP transport method [XEP-0176]
defines a mode for supporting traditional Offer/Answer interactions
through the "urn:ietf:rfc:3264" feature tag. When an XMPP entity
such as a client (or, significantly, a gateway to a SIP system)
advertises support for this feature, the entity indicates that it
needs to receive multiple transport candidates in the initial offer,
instead of receiving them trickled over time. Although
implementations conforming to this specification MUST support the
Offer/Answer model with Jingle, such endpoints SHOULD NOT actually
declare support for the "urn:ietf:rfc:3264" service discovery feature
since this would mean that they too would be reachable only through
Offer/Answer semantics and not also through trickle-ICE semantics.
The difference in handling of transport candidates also has an impact
on ICE restarts (see Section 9.1.1.1 of [RFC5245]). Because Jingle
endpoints can send candidates at any time, when communicating
directly with other Jingle endpoints they would not initiate an ICE
restart simply in order to send a candidate that, for example,
changes the media target. However, as part of support for the Offer/
Answer model a Jingle endpoint would instead need to initiate an ICE
restart when communicating with a SIP endpoint or gateway that does
not support trickle ICE. Similarly, a Jingle endpoint needs to
support the 'generation' attribute (used to signal an ICE restart)
when communicating with a SIP endpoint or gateway that does not
support trickle ICE. See also the syntax discussion under
Section 5.4.
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 high-level 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 (via the <content/> and
<description/> elements) using a transport of RTP over raw UDP (via
the <transport/> element).
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Example 1: Structure of a Jingle session initiation request
| <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='101' name='opus' clockrate='48000'/>
| <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 id='v3c18fgg'
| ip='10.1.1.104'
| port='13540'
| generation='0'/>
| </transport>
| </content>
| </jingle>
| </iq>
The syntax and semantics of the <jingle/> and <content/> elements are
defined in the core Jingle specification [XEP-0166], the syntax and
semantics of the <description/> element qualified by the
'urn:xmpp:jingle:app:rtp:1' namespace are defined in the Jingle RTP
specification [XEP-0167], and the syntax and semantics of the
<transport/> element qualified by the 'urn:xmpp:jingle:transport:raw-
udp' namespace are defined in the Jingle Raw UDP specification
[XEP-0177]. Other <description/> elements are defined in
specifications for the appropriate application types (see for example
[XEP-0234] for file transfer) and other <transport/> elements are
defined in the specifications for appropriate transport methods (see
for example [XEP-0176], which defines an XMPP profile of ICE
[RFC5245]).
At the core Jingle layer, the following mappings are defined.
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Table 1: High-Level Mapping from XMPP to SIP
+--------------------------------+--------------------------------+
| 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 mandatory mapping (1) |
+--------------------------------+--------------------------------+
| <content/> 'senders' value of | a= line of sendrecv, recvonly, |
| both, initiator, responder, or | sendonly, or inactive |
| none | |
+--------------------------------+--------------------------------+
1. In can be appropriate to map to the a=mid value defined in
[RFC5888].
The 'senders' attribute is optional in Jingle, with a default value
of "both"; thus in case the attribute is absent the SDP direction
value MUST be considered as 'sendrecv'.
The 'action' attribute of the <jingle/> element has 15 allowable
values. In general they should be mapped as shown in the following
table, with some exceptions as described below.
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Table 2: Mapping of Jingle Actions to SIP Methods
+-------------------+------------------------------+
| Jingle Action | SIP Method |
+-------------------+------------------------------+
| content-accept | INVITE response (1xx or 2xx) |
+-------------------+------------------------------+
| content-add | INVITE request |
+-------------------+------------------------------+
| content-modify | re-INVITE request |
+-------------------+------------------------------+
| content-reject | unused in this mapping |
+-------------------+------------------------------+
| content-remove | INVITE request |
+-------------------+------------------------------+
| description-info | unused in this mapping |
+-------------------+------------------------------+
| security-info | unused in this mapping |
+-------------------+------------------------------+
| session-accept | INVITE response (1xx or 2xx) |
+-------------------+------------------------------+
| session-info | see note (1) below |
+-------------------+------------------------------+
| session-initiate | INVITE request |
+-------------------+------------------------------+
| session-terminate | BYE |
+-------------------+------------------------------+
| transport-accept | unused in this mapping |
+-------------------+------------------------------+
| transport-info | see note (2) below |
+-------------------+------------------------------+
| transport-reject | unused in this mapping |
+-------------------+------------------------------+
| transport-replace | unused in this mapping |
+-------------------+------------------------------+
1. The Jingle session-info action can be used for multiple purposes,
such as putting the session on hold or sending a ringing
indication. In particular, a session-info action of type
'ringing' SHOULD be mapped to a 180 SIP provisional response.
The use of session-info for the purpose of session hold is
described in Section 7.
2. In Jingle the transport-info action is used to exchange transport
candidates after the initial offer, as documented in [XEP-0176].
This usage has been generalized as "Trickle ICE"
[I-D.ietf-mmusic-trickle-ice] and has also been extended to SIP
[I-D.ietf-mmusic-trickle-ice-sip]. Therefore a Jingle action of
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transport-info SHOULD be mapped to a SIP INFO request, but only
in cases where it is reasonable to assume that the SIP endpoint
or gateway supports trickle ICE. See Section 4 for further
discussion.
5.2. Application Formats
Jingle application formats for audio and video exchange via RTP are
specified in [XEP-0167]. These application formats effectively map
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 that 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 "Raw UDP" transport method involves
the negotiation of an IP address and port only. It does not provide
NAT traversal, effectively leaving the task to intermediary entities
(which might be a media relay associated with but functionally
independent of a signaling gateway). 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.
5.4. ICE-UDP Transport Method
A more advanced Jingle transport method for exchanging media over UDP
uses Interactive Connectivity Establishment and is specified in
[XEP-0176]. By following the ICE methodology specified in [RFC5245],
ideally this transport method provides NAT traversal for media.
The relevant SDP mappings are provided in [XEP-0176]. However, those
who implement signaling gateways need to be aware of a few syntax
incompatibilities that 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
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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. This attribute has no counterpart
in SIP because ICE restarts are initiated by detecting a change in
the ICE 'ufrag' and 'pwd' (see Section 9.1.1.1 of [RFC5245]).
Gateways MUST therefore increase the generation number when they
detect such a change.
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. Transport Fallback
Most Jingle endpoints will first attempt to use ICE as specified for
Jingle in [XEP-0176] (since that is most likely to result in NAT
traversal) and only if that does not succeed will they fall back to
raw UDP [XEP-0177]. This fallback approach is described in the
Jingle ICE specification [XEP-0176].
However, that approach depends on the use of XMPP service discovery
[XEP-0030]. Because SIP does not have a method for determining
endpoint capabilities, SIP endpoints use what can be termed "single-
exchange fallback": they first try one method and if that fails they
then send a re-INVITE with the second method.
One way to map single-exchange fallback to Jingle is for the Jingle
endpoint to attempt ICE first and send a transport-replace if the SIP
answer indicates no support for ICE, then send a SIP re-INVITE with
the addresses in the transport-accept. Unfortunately, this approach
will result a fairly substantial post-answer delay before media can
flow.
Because such delays usually result in an unacceptable user
experience, the trend for many calling applications is to first send
only a candidate that is known beforehand to be highly likely to
result in NAT traversal, which is almost always a candidate at a
media relay (i.e., an ICE candidate of type "relay"). Such
applications will then offer and perhaps switch to a host candidate,
peer reflexive candidate, or server reflexive candidate only after
media is flowing via the relayed candidate. This approach obviates
the need for transport fallback from ICE to raw UDP during call
setup, and instead works around the problem by using trickle ICE (for
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those endpoints that support it) or re-INVITEs with updated transport
candidates after call setup has been completed.
7. Call Hold
The Offer/Answer model [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 (the Jingle
specification also defines a value of "none", which maps to an a=
value of "inactive", and a default value of "both", which maps to an
a= value of "sendrecv").
The following example shows how the responder would put the call on
hold (i.e., temporarily stop listening to media sent by the
initiator) using a Jingle content-modify action and a modified value
for the 'senders' attribute (here a value of "responder" is used to
indicate that the responder might continue to send media, such as
hold music).
Example 2: Call hold via 'senders' attribute
| <iq from='juliet@capulet.lit/balcony'
| id='hz73n2l9'
| to='romeo@montague.lit/orchard'
| type='set'>
| <jingle xmlns='urn:xmpp:jingle:1'
| action='content-modify'
| initiator='romeo@montague.lit/orchard'
| sid='a73sjjvkla37jfea'>
| <content creator='initiator'
| media='audio'
| name='this-is-the-audio-content'
| senders='responder'/>
| </jingle>
| </iq>
In addition to these semantics, however, the Jingle RTP Sessions
specification [XEP-0167] also defines a more concise way for
achieving the same end, which consists in sending a "hold" command
within a "session-info" action, as shown in the following example.
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Example 3: Call hold via session-info action
| <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 either of the foregoing hold notifications 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, 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 on hold also
contain other session modifications.
It is possible that, after one entity has put the other on hold, the
second entity might put the first entity on hold. In this case, the
effective direction would then be "inactive" in SDP and "none" in
Jingle.
8. Early Media
[RFC3959] and [RFC3960] describe a number of scenarios relying on
"early media". While similar attempts have also been made for XMPP,
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 can 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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9. 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. This can happen not only between two endpoints, but
also with the addition of a third endpoint into the mix (e.g.,
because one of the two original endpoints forwards a call to a third
endpoint, thus converting a "spiral" into a loop).
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. Although the specifics of such tracking are a
matter of implementation, the broad requirements is to consistently
translate SIP dialog IDs into Jingle session ID, and vice versa, or
generate an internal identifier for each session (e.g., by
concatenating or hashing the combination of the SIP dialog ID and the
Jingle session ID).
10. SDP Format-Specific Parameters
The SDP specification [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, commas, or
whatever delimiters are chosen by a particular payload type
specification.
The Jingle RTP application format [XEP-0167], on the other hand,
defines a <parameter/> element as follows:
<parameter name="paramName" value="paramValue"/>
A sequence of parameters is thus transmitted as an array of distinct
name/value pairs, at least in the context of the Jingle RTP
extension.
These differences make it difficult to devise a generic mechanism
that accurately translates format parameters from Jingle RTP to SDP
without the specifics of the payload being known to the gateway. In
general this is not a major problem because most of the media type
definitions supported in existing Jingle implementations follow the
semicolon-separated parameter model (e.g., typical audio and video
codecs). Possible exceptions include:
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o The RTP Payload for DTMF Digits, Telephony Tones, and Telephony
Signals (i.e., the "audio/telephone-event" payload type). As
noted in Section 2.5.1.1 of [RFC4733], in SDP the "events"
parameter is assumed to indicate support for DTMF events 0-15 even
if the parameter is not included; a gateway SHOULD explicitly
indicate this support in a Jingle parameter with name='events' and
value='0-15'.
o The RTP Payload for Redundant Audio Data (i.e., the "audio/red"
payload type). Although this payload type is defined in
[RFC2198], the SDP representation is specified in Section 4.1.21
of [RFC3555] (note that this representation was not updated by
[RFC4856]). In particular, the "pt" parameter can be mapped to
a=fmtp lines as described in the payload type registration.
For implementations that wish to provide a general-purpose
translation method, this specification makes the following
recommendations:
1. Gateways that are aware of the formats in use SHOULD parse all
format parameters and generate <parameter/> arrays and "a=fmtp"
values accordingly.
2. When translating Jingle RTP to SIP, gateways that have no
explicit support for the formats that are being negotiated SHOULD
convert the list of <parameter/> 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 RTP, gateways that have no
explicit support for the formats that are being negotiated SHOULD
tokenize the "a=fmtp" format string using one delimiter from the
following list: ";", "; ", ",", ", ". 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 <parameter/> elements. If some of the
tokens cannot be parsed into a "name=value" pair because they do
not conform to the convention suggested in [RFC4855], or in case
the format string could not be tokenized with the above
delimiters, the remaining strings SHOULD be used as a value for
the "value" attribute of the <parameter/> element and the
corresponding "name" attribute SHOULD be left empty.
Here is a relatively simple example of the foregoing transformations,
using the aforementioned example of the "audio/telephone-event"
payload type (wherein the "events" parameter is implicitly named in
the SDP):
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SDP with format data (audio/telephone-event)
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15,66,70
Jingle transformation (audio/telephone-event)
<parameter name="events" value="0-15,66,70"/>
A more complicated example would be handling of the "audio/red"
payload type (wherein the "pt" parameter can be mapped to a=fmtp
lines as described in [RFC3555]):
SDP with format data (audio/red)
m=audio 49170 RTP/AVP 99 0 103
a=rtpmap:99 RED/8000
a=fmtp:99 0/103
a=rtpmap:103 G729D/8000
a=fmtp:103 annexb=yes
Jingle transformation (audio/red)
<parameter name="pt" value="0,103"/>
<parameter name="annexb" value="yes"/>
11. Dialog Forking
The core SIP specification [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, a SIP-to-XMPP gateway MUST NOT forward more than one
provisional response on the 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 if it
originated from a "user@host" address (i.e., a "bare 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, a gateway 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 was not forwarded to the Jingle side.
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12. Sample Call Flow
The section illustrates the protocol flow of a basic voice chat
session in which an XMPP user (juliet@example.com) is the initiator
and a SIP user (romeo@example.net) is the responder. The signaling
is communicated through a gateway. To simplify the example, the
Jingle transport method negotiated is "raw UDP" as specified in
[XEP-0177].
XMPP XMPP SIP SIP
User Server Server User
| + X2S GW | |
| | | |
| (F1) XMPP | | |
| session- | | |
| initiate | | |
|...........>| | |
| (F2) XMPP | | |
| IQ-result | | |
|<...........| | |
| | (F3) SIP | |
| | INVITE | |
| |***********>| |
| | | (F4) SIP |
| | | INVITE |
| | |**********>|
| | | (F5) SIP |
| | | 180 |
| | | ringing |
| | |<**********|
| | (F6) SIP | |
| | 180 ringing| |
| |<***********| |
| (F7) XMPP | | |
| session- | | |
| info | | |
| (ringing) | | |
|<...........| | |
| (F8) XMPP | | |
| IQ-result | | |
|...........>| | |
| | | (F9) SIP |
| | | 200 OK |
| | |<**********|
| | (F10) SIP | |
| | 200 OK | |
| |<***********| |
| (F11) XMPP | | |
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| session- | | |
| accept | | |
|<...........| | |
| (F12) XMPP | | |
| IQ-result | | |
|...........>| | |
| | (F13) SIP | |
| | ACK | |
| |***********>| |
| | | (F14) SIP |
| | | ACK |
| | |**********>|
| | | |
|<=======MEDIA SESSION OVER RTP======>|
| | | |
| | | (F15) SIP |
| | | BYE |
| | |<**********|
| | (F16) SIP | |
| | BYE | |
| |<***********| |
| (F17) XMPP | | |
| session- | | |
| terminate | | |
|<...........| | |
| (F18) XMPP | | |
| IQ-result | | |
|...........>| | |
The packet flow is as follows.
First the XMPP user sends a Jingle session-initiation request to the
SIP user.
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Example 4: Jingle session-initiate (F1)
| <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 component='1' generation='0' id='u3gscv289p'
| ip='192.0.2.101' port='49172'/>
| </transport>
| </content>
| </jingle>
| </iq>
The gateway returns an XMPP IQ-result to the initiator on behalf of
the responder.
Example 5: XMPP IQ-result from gateway (F2)
| <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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Example 6: SIP transformation of Jingle session-initiate (F3)
| 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.
Example 7: SIP 180 Ringing message (F5)
| 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.
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Example 8: XMPP transformation of SIP 180 Ringing message (F7)
| <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
ringing message, which is used only by the gateway and not
transformed into SIP syntax.
Example 9: XMPP entity acknowledges ringing message (F8)
| <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 in order to accept
the session initiation request.
Example 10: SIP user accepts session request (F9)
| 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
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The gateway transforms the 200 OK into a Jingle session-accept
action.
Example 11: XMPP transformation of SIP 200 OK (F11)
| <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 id='9eg13am7' 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.
Example 12: XMPP user acknowledges session-accept (F12)
| <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 can continue the session as long as desired.
Eventually, one of the parties (in this case the responder)
terminates the session.
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Example 13: SIP user ends session (F15)
| 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: 4 BYE
| Content-Length: 0
The gateway transforms the SIP BYE into XMPP syntax.
Example 14: XMPP transformation of SIP BYE (F17)
| <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'
| sid='a73sjjvkla37jfea'/>
| <reason>
| <success/>
| </reason>
| </jingle>
| </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.
Example 15: XMPP user acknowledges end of session (F18)
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='rv301b47'
to='juliet@example.com/t3hr0zny'
type='result'/>
13. IANA Considerations
This document requests no actions by the IANA.
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14. 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 [RFC7247] also apply.
The addition of gateways to the security model of media signaling
introduces some new risks. In particular, end-to-end security
properties (especially confidentiality and integrity) between media
endpoints that interface through a gateway can be provided only if
common formats are supported. Specification of those common formats
is out of scope for this document and, unfortunately, no generalized
method for end-to-end encryption of signaling messages has yet been
defined, even outside of recognized standards development
organizations (e.g., [RFC3862] and [RFC3923] are not widely
implemented and Off-the-Record Messaging [OTR] can handle only human-
readable chat messages, not structured signaling payloads).
15. References
15.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[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.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
December 2006.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
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[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.
[RFC7247] Saint-Andre, P., Houri, A., and J. Hildebrand,
"Interworking between the Session Initiation Protocol
(SIP) and the Extensible Messaging and Presence Protocol
(XMPP): Architecture, Addresses, and Error Handling", RFC
7247, May 2014.
[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.
15.2. Informative References
[I-D.ietf-mmusic-trickle-ice-sip]
Ivov, E., Marocco, E., and C. Holmberg, "A Session
Initiation Protocol (SIP) usage for Trickle ICE", draft-
ietf-mmusic-trickle-ice-sip-02 (work in progress), July
2015.
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[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-02 (work in progress), January 2015.
[OTR] Ian Goldberg, , "Off-the-Record Messaging", <https://
otr.cypherpunks.ca/>.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
September 1997.
[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.
[RFC3555] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003.
[RFC3862] Klyne, G. and D. Atkins, "Common Presence and Instant
Messaging (CPIM): Message Format", RFC 3862, August 2004.
[RFC3923] Saint-Andre, P., "End-to-End Signing and Object Encryption
for the Extensible Messaging and Presence Protocol
(XMPP)", RFC 3923, October 2004.
[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.
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[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC4856] Casner, S., "Media Type Registration of Payload Formats in
the RTP Profile for Audio and Video Conferences", RFC
4856, DOI 10.17487/RFC4856, February 2007,
<http://www.rfc-editor.org/info/rfc4856>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
[RFC5547] Garcia-Martin, M., Isomaki, M., Camarillo, G., Loreto, S.,
and P. Kyzivat, "A Session Description Protocol (SDP)
Offer/Answer Mechanism to Enable File Transfer", RFC 5547,
May 2009.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC5768] Rosenberg, J., "Indicating Support for Interactive
Connectivity Establishment (ICE) in the Session Initiation
Protocol (SIP)", RFC 5768, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC7081] Ivov, E., Saint-Andre, P., and E. Marocco, "CUSAX:
Combined Use of the Session Initiation Protocol (SIP) and
the Extensible Messaging and Presence Protocol (XMPP)",
RFC 7081, November 2013.
[RFC7230] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Message Syntax and Routing", RFC 7230, June
2014.
[RFC7395] Stout, L., Moffitt, J., and E. Cestari, "An Extensible
Messaging and Presence Protocol (XMPP) Subprotocol for
WebSocket", RFC 7395, October 2014.
[XEP-0124]
Paterson, I., Smith, D., Saint-Andre, P., Moffitt, J., and
L. Stout, "Bidirectional-streams Over Synchronous HTTP
(BOSH)", XSF XEP 0124, November 2013.
Saint-Andre, et al. Expires January 21, 2016 [Page 27]
Internet-Draft SIP-XMPP Interworking: Media Sessions July 2015
[XEP-0234]
Saint-Andre, P., "Jingle File Transfer", XSF XEP 0234,
August 2014.
[XEP-0269]
Cionoiu, D. and P. Saint-Andre, "Jingle Early Media", XSF
XEP 0269, May 2009.
Appendix A. Acknowledgements
Thanks to Dave Crocker, Philipp Hancke, Paul Kyzivat, and Jonathan
Lennox for their feedback. Jonathan in particular provided helpful
suggestions regarding the transport fallback section.
The authors gratefully acknowledge the assistance of Markus Isomaki
and Yana Stamcheva as the working group chairs and Ben Campbell as
the sponsoring Area Director.
Peter Saint-Andre wishes to acknowledge Cisco Systems, Inc., for
employing him during his work on earlier draft versions of this
document.
Authors' Addresses
Peter Saint-Andre
&yet
Email: peter@andyet.com
URI: https://andyet.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
Saint-Andre, et al. Expires January 21, 2016 [Page 28]
