%ents; ]>
Jingle RTP Sessions This specification defines a Jingle application type for negotiating one or more sessions that use the Real-time Transport Protocol (RTP) to exchange media such as voice or video. The application type includes a straightforward mapping to Session Description Protocol (SDP) for interworking with SIP media endpoints. &LEGALNOTICE; 0167 Experimental Standards Track Standards Council XMPP Core XEP-0166 NOT_YET_ASSIGNED &scottlu; &stpeter; &seanegan; &robmcqueen; 0.23 2008-07-31 ram/psa

Removed profile attribute; modified secure session establishment to align with SRTP usage.

0.22 2008-06-09 psa

Added name attribute to active element to mirror usage for mute element; clarified meaning of session in the context of this specification; recommended that all sessions established via the same Jingle negotiation should be treated as synchronized.

0.21 2008-06-09 psa

Added name attribute to mute element for more precise handling of informational messages.

0.20 2008-06-04 psa

In accordance with list consensus, generalized to cover all RTP media, not just audio; corrected text regarding payload types sent by responder in order to match SDP approach.

0.19 2008-05-28 psa

Specified default value for profile attribute; clarified relationship to SDP offer-answer model.

0.18 2008-05-28 psa

Removed content-replace from ICE-UDP examples per XEP-0176.

0.17 2008-02-29 psa

Corrected use of content-replace action per XEP-0166.

0.16 2008-02-28 psa

Moved profile attribute from XEP-0166 to this specification.

0.15 2008-01-11 psa

Removed content-accept after content-remove per XEP-0166.

0.14 2008-01-03 psa

Modified examples to track changes to XEP-0176.

0.13 2007-12-06 psa

To track changes to XEP-0166, modified busy scenario and removed unsupported-codecs error.

0.12 2007-11-27 psa

Further editorial review.

0.11 2007-11-15 psa

Editorial review and consistency check; moved voice chat scenarios from XEP-0166 to this specification.

0.10 2007-11-13 psa

Removed info message for busy since it is now a Jingle-specific error condition defined in XEP-0166; defined info message for active.

0.9 2007-04-17 psa

Specified Jingle conformance, including the preference for lossy transports over reliable transports and the process of sending and receiving audio content over each transport type.

0.8 2007-03-23 psa/ram

Renamed to mention RTP as the associated transport; corrected negotiation flow to be consistent with SIP/SDP (each party specifies a list of the payload types it can receive); added profile attribute to content element in order to specify RTP profile in use.

0.7 2006-12-21 psa

Modified spec to use provisional namespace before advancement to Draft (per XEP-0053).

0.6 2006-10-31 psa/se

Specified how to include SDP parameters and codec-specific parameters; clarified negotiation process; added Speex examples; removed queued info message.

0.5 2006-08-23 psa

Modified namespace to track XEP-0166.

0.4 2006-07-12 se/psa

Specified when to play received audio (early media); specified that DTMF must use in-band signalling (XEP-0181).

0.3 2006-03-20 psa

Defined info messages for hold and mute.

0.2 2006-02-13 psa

Defined info message for busy; added info message examples; recommended use of Speex; updated schema and XMPP Registrar considerations.

0.1 2005-12-15 psa

Initial version.

0.0.3 2005-12-05 psa

Described service discovery usage; defined initial informational messages.

0.0.2 2005-10-27 psa

Added SDP mapping, security considerations, IANA considerations, XMPP Registrar considerations, and XML schema.

0.0.1 2005-10-21 psa/sl

First draft.

&xep0166; can be used to initiate and negotiate a wide range of peer-to-peer sessions. One session type of interest is media such as voice or video. This document specifies an application format for negotiating Jingle media sessions, where the media is exchanged over the Realtime Transport Protocol (RTP; see &rfc3550;).

The Jingle application format defined herein is designed to meet the following requirements:

  1. Enable negotiation of parameters necessary for media sessions using the Realtime Transport Protocol (RTP).
  2. Map these parameters to Session Description Protocol (SDP; see &rfc4566;) to enable interoperability.
  3. Define informational messages related to typical RTP uses such as audio chat and video chat (e.g., ringing, on hold, on mute).

In accordance with Section 8 of XEP-0166, this document specifies the following information related to the Jingle RTP application type:

  1. The application format negotiation process is defined in the Negotiating a Jingle RTP Session section of this document.

  2. The semantics of the &DESCRIPTION; element are defined in the Application Format section of this document.

  3. A mapping of Jingle semantics to the Session Description Protocol is provided in the Mapping to Session Description Protocol section of this document.

  4. A Jingle RTP session SHOULD use a lossy transport method such as &xep0177; or the "ice-udp" method specified in &xep0176;, but MAY use a reliable transport such as "ice-tcp" if a low-bandwidth codec is employed.

  5. Content is to be sent and received as follows:

    • For lossy transports, outbound content shall be encoded into RTP packets and each packet shall be sent individually over the transport. Each inbound packet received over the transport is an RTP packet.

    • For reliable transports, outbound content shall be encoded into RTP packets and each packet data shall be sent in succession over the transport. Incoming data received over the transport shall be processed as a stream of RTP packets, where each RTP packet boundary marks the location of the next packet.

A Jingle RTP session is described by a content type that contains one application format and one transport method. Each <content/> element defines a single RTP session. A Jingle negotiation MAY result in the establishment of multiple RTP sessions (e.g., one for audio and one for video). An application SHOULD consider all of the RTP sessions that are established via the same Jingle negotiation to be synchronized for purposes of streaming, playback, recording, etc.

The application format consists of one or more encodings contained within a wrapper <description/> element qualified by the 'urn:xmpp:tmp:jingle:apps:rtp' namespace &NSNOTE;. In the language of RFC 4566 each encoding is a payload-type; therefore, each <payload-type/> element specifies an encoding that can be used for the RTP stream, as illustrated in the following example.

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The &DESCRIPTION; element is intended to be a child of a &CONTENT; element as specified in XEP-0166.

The &DESCRIPTION; element MUST possess a 'media' attribute that specifies the media type, such as "audio" or "video".

The encodings SHOULD be provided in order of preference by placing the most-preferred &PAYLOADTYPE; element as the first child of the &DESCRIPTION; element (etc.).

The allowable attributes of the &PAYLOADTYPE; element are as follows:

Attribute Description Datatype Inclusion
channels The number of channels; if omitted, it MUST be assumed to contain one channel positiveInteger (defaults to 1) RECOMMENDED
clockrate The sampling frequency in Hertz positiveInteger RECOMMENDED
id The payload identifier positiveInteger REQUIRED
maxptime Maximum packet time as specified in RFC 4566 positiveInteger OPTIONAL
name The appropriate subtype of the MIME type string RECOMMENDED for static payload types, REQUIRED for dynamic payload types
ptime Packet time as specified in RFC 4566 positiveInteger OPTIONAL

In Jingle RTP, the encodings are used in the context of RTP. The most common encodings for the Audio/Video Profile (AVP) of RTP are listed in &rfc3551; (these "static" types are reserved from payload ID 0 through payload ID 95), although other encodings are allowed (these "dynamic" types use payload IDs 96 to 127) in accordance with the dynamic assignment rules described in Section 3 of RFC 3551. The payload IDs are represented in the 'id' attribute.

Each <payload-type/> element MAY contain one or more child elements that specify particular parameters related to the payload. For example, as described in &rtpspeex;, the "cng", "mode", and "vbr" parameters may be specified in relation to usage of the Speex See <http://www.speex.org/>. codec. Where such parameters are encoded via the "fmtp" SDP attribute, they shall be represented in Jingle via the following format:

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The order of parameter elements MUST be ignored.

Parameter names MUST be treated as case-sensitive. However, parameter names are effectively guaranteed to be unique, since &IANA; maintains a registry of SDP parameters (see <http://www.iana.org/assignments/sdp-parameters>).

In general, the process for negotiating a Jingle RTP session is as follows:

| | ack | |<----------------------------| | [transport negotiation] | |<--------------------------->| | session-accept | |<----------------------------| | ack | |---------------------------->| | AUDIO (RTP) | |<===========================>| | | ]]>

When the initiator sends a session-initiate stanza to the responder, the &DESCRIPTION; element includes all of the payload types that the initiator can send and/or receive for Jingle RTP, each one encapsulated in a separate &PAYLOADTYPE; element (the rules specified in &rfc3264; SHOULD be followed regarding inclusion of payload types).

action='session-initiate' initiator='romeo@montague.net/orchard' sid='a73sjjvkla37jfea'> ]]>

Upon receiving the session-initiate stanza, the responder determines whether it can proceed with the negotiation. The general Jingle error cases are specified in XEP-0166 and illustrated in the Scenarios section of this document.

If there is no error, the responder acknowledges the session initiation request.

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After successful transport negotiation (not shown here), the responder accepts the session by sending a session-accept action to the initiator. The session-accept SHOULD include a subset of the payload types sent by the initiator, i.e., a list of the offered payload types that the responder can send and/or receive. The list that the responder sends SHOULD retain the ID numbers specified by the initiator. The order of the &PAYLOADTYPE; elements indicates the responder's preferences, with the most-preferred types first.

In the following example, we imagine that the responder supports Speex at clockrate of 8000 but not 16000, G729, and PCMU but not PMCA. Therefore the responder returns only two payload types.

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And the initiator acknowledges session acceptance:

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The initiator and responder would then exchange media using any of the codecs that meet the following criteria:

The SDP media type for Jingle RTP is "audio" (see Section 8.2.1 of RFC 4566) for audio media, "video" (see Section 8.2.1 of RFC 4566) for video media, etc.

If the payload type is static (payload-type IDs 0 through 95 inclusive), it MUST be mapped to a media field defined in RFC 4566. The generic format for the media field is as follows:

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In the context of Jingle audio sessions, the <media> is "audio" or "video" or some other media type, the <port> is the preferred port for such communications (which may be determined dynamically), and the <fmt list> is the payload-type ID.

For example, consider the following static payload-type:

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That Jingle-formatted information would be mapped to SDP as follows:

If the payload type is dynamic (payload-type IDs 96 through 127 inclusive), it SHOULD be mapped to an SDP media field plus an SDP attribute field named "rtpmap".

For example, consider a payload of 16-bit linear-encoded stereo audio sampled at 16KHz associated with dynamic payload-type 96:

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That Jingle-formatted information would be mapped to SDP as follows:

As noted, if additional parameters are to be specified, they shall be represented as attributes of the <parameter/> child of the &PAYLOADTYPE; element, as in the following example.

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That Jingle-formatted information would be mapped to SDP as follows:

The formatting is similar for video parameters, as shown in the following example.

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That Jingle-formatted information would be mapped to SDP as follows:

&rfc3711; defines the Secure Real-time Transport Protocol, and &rfc4568; defines the SDP "crypto" attribute for signalling and negotiating the use of SRTP in the context of offer-answer protocols such as SIP. To enable the use of SRTP and gatewaying to non-XMPP technologies that make use of the "crypto" SDP attribute, we define a corresponding <crypto/> element qualified by the 'urn:xmpp:tmp:jingle:apps:rtp' namespace.

If the initiator wishes to use SRTP, the session-initiate MUST include at least one <crypto/> element and MAY multiple instances of the element. The <crypto/> element MUST be a child of the <description/> element.

The XML attributes of the <crypto/> element are as follows:

An example follows.

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The mapping to SDP is as follows.

When the responder receives a session-initiate action containing one or more instances of the <crypto/> element, it MUST either accept one of the <crypto/> elements or reject the offer by sending a session-terminate action with a reason of <invalid-crypto/>.

Informational messages may be sent by either party within the context of Jingle to communicate the status of a Jingle RTP session, device, or principal. The informational message MUST be an IQ-set containing a &JINGLE; element of type "session-info", where the informational message is a payload element qualified by the 'urn:xmpp:tmp:jingle:apps:rtp:info' namespace; the following payload elements are defined: A <trying/> element (equivalent to the SIP 100 Trying response code) is not necessary, since each session-level action is acknowledged via XMPP IQ semantics.

Element Meaning
<active/> The principal or device is again actively participating in the session after having been on hold or on mute. The <active/< element MAY possess a 'name' attribute whose value specifies a particular session that is again active (e.g., activating the video aspect but not the voice aspect of a voice+video chat). If no 'name' attribute is included, the recipient MUST assume that all sessions are active.
<hold/> The principal is temporarily pausing the chat (i.e., putting the other party on hold).
<mute/> The principal is temporarily stopping media output but continues to accept media input. The <mute/< element MAY possess a 'name' attribute whose value specifies a particular session to be muted (e.g., muting the video aspect but not the voice aspect of a voice+video chat). If no 'name' attribute is included, the recipient MUST assume that all sessions are to be muted.
<ringing/> The device is ringing but the principal has not yet interacted with it to answer (this maps to the SIP 180 response code).

Note: Because the informational message is sent in an IQ-set, the receiving party MUST return either an IQ-result or an IQ-error (normally only an IQ-result to acknowledge receipt; no error flows are defined or envisioned at this time).

action='session-info' initiator='romeo@montague.net/orchard' sid='a73sjjvkla37jfea'> ]]> ]]> ]]> ]]>

If an entity supports Jingle RTP session, it MUST advertise that fact by returning a feature of "urn:xmpp:tmp:jingle:apps:rtp" &NSNOTE; in response to &xep0030; information requests.

]]> ... ... ]]>

Naturally, support MAY also be determined via the dynamic, presence-based profile of Service Discovery defined in &xep0115;.

The following sections show a number of Jingle RTP scenarios, in relative order of complexity.

In this scenario, Romeo initiates a voice chat with Juliet but she is otherwise engaged.

The session flow is as follows:

| | ack | |<----------------------------| | session-info (ringing) | |<----------------------------| | ack | |---------------------------->| | terminate | | (reason = busy) | |<----------------------------| | ack | |---------------------------->| | | ]]>

The protocol flow is as follows.

]]> ]]> ]]> ]]> No time to chat right now! ]]>

The other party then acknowledges termination of the session:

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In this scenario, Romeo initiates a voice chat with Juliet using a transport method of ICE-UDP. The parties also exchange informational messages.

The session flow is as follows:

| | ack | |<----------------------------| | session-info (ringing) | |<----------------------------| | ack | |---------------------------->| | transport-info (X times) | | (with acks) | |<--------------------------->| | STUN connectivity checks | |<===========================>| | session-accept | |<----------------------------| | ack | |---------------------------->| | AUDIO (RTP) | |<===========================>| | session-terminate | |<----------------------------| | ack | |---------------------------->| | | ]]>

The protocol flow is as follows.

]]> ]]> ]]> ]]>

Because the parties have chosen the Jingle ICE-UDP Transport Method, the initiator and responder exchange an open-ended number of possible candidate transports, perform connectivity checks, and agree upon a candidate transport as explained in XEP-0176. Once ICE negotiation is completed, the responder sends a session-accept action to the initiator.

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If the payload types and transport candidate can be successfully used by both parties, then the initiator acknowledges the session-accept action.

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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 terminates the session.

Sorry, gotta go! ]]>

The other party then acknowledges termination of the session:

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In this scenario, Romeo initiates a combined audio and video chat with Juliet using a transport method of ICE-UDP. Juliet at first refuses the video portion, then later offers to add video, which Romeo accepts. The parties also exchange various informational messages

The session flow is as follows:

| | ack | |<----------------------------| | session-info (ringing) | |<----------------------------| | ack | |---------------------------->| | content-remove | |<----------------------------| | ack | |---------------------------->| | transport-info (X times) | | (with acks) | |<--------------------------->| | STUN connectivity checks | |<===========================>| | session-accept | |<----------------------------| | ack | |---------------------------->| | AUDIO (RTP) | |<===========================>| | session-info (hold) | |<----------------------------| | ack | |---------------------------->| | session-info (active) | |<----------------------------| | ack | |---------------------------->| | content-add | |<----------------------------| | ack | |---------------------------->| | content-accept | |---------------------------->| | ack | |<----------------------------| | AUDIO + VIDEO (RTP) | |<===========================>| | session-terminate | |<----------------------------| | ack | |---------------------------->| | | ]]>

The protocol flow is as follows.

]]> ]]> ]]> ]]>

However, Juliet doesn't want to do video because she is having a bad hair day, so she sends a "content-remove" request to Romeo.

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Romeo then acknowledges the content-remove request:

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Because the parties have chosen the Jingle ICE-UDP Transport Method, the initiator and responder exchange an open-ended number of possible candidate transports, perform connectivity checks, and agree upon a candidate transport as explained in XEP-0176. Once ICE negotiation is completed, the responder sends a session-accept action to the initiator.

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As above, if the payload types and transport candidate can be successfully used by both parties, then the initiator acknowledges the session-accept action.

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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 chat for a while. Eventually Juliet wants to get her hair in order so she puts Romeo on hold.

]]> ]]>

Juliet returns so she informs Romeo that she is actively engaged in the call again.

]]> ]]>

The parties now continue the audio chat.

Finally Juliet decides that she is presentable for a video chat so she sends a content-add request to Romeo.

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The entity receiving the content-add request then acknowledges the request and, if it is acceptable, returns a content-accept action:

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The other party then acknowledges the acceptance.

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The media session proceeds. Now they would exchange both audio and video, where the audio is exchanged via the Speex codec at a clockrate of 8000 and the video is exchanged using the Theora codec with a height of 720 pixels, a width of 1280 pixels, and so on.

The parties may continue the session as long as desired.

Eventually, one of the parties terminates the session.

I'm outta here! ]]> ]]>

In this scenario, Romeo initiates a secure voice chat with Juliet using a transport method of ICE-UDP. The parties also exchange informational messages.

The session flow is as follows:

| | ack | |<----------------------------| | session-info (ringing) | |<----------------------------| | ack | |---------------------------->| | transport-info (X times) | | (with acks) | |<--------------------------->| | STUN connectivity checks | |<===========================>| | session-accept | |<----------------------------| | ack | |---------------------------->| | AUDIO (RTP) | |<===========================>| | session-terminate | |<----------------------------| | ack | |---------------------------->| | | ]]>

The protocol flow is as follows.

]]> ]]> ]]> ]]>

Because the parties have chosen the Jingle ICE-UDP Transport Method, the initiator and responder exchange an open-ended number of possible candidate transports, perform connectivity checks, and agree upon a candidate transport as explained in XEP-0176. Once ICE negotiation is completed, the responder sends a session-accept action to the initiator.

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If the payload types and transport candidate can be successfully used by both parties, then the initiator acknowledges the session-accept action.

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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 terminates the session.

Sorry, gotta go! ]]>

The other party then acknowledges termination of the session:

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For the sake of interoperability with a wide variety of free and open-source voice systems as well as deployment of patent-free technologies, support for the Speex codec is RECOMMENDED.

For the sake of interoperability with the public switched telephone network (PSTN) and most VoIP providers, support for the Pulse Code Modulation (PCM) codec defined in &ITU; recommendation G.711 is RECOMMENDED, including both the μ-law ("U-law") and A-law versions widely deployed in North America and Japan and in the rest of the world respectively.

If it is necessary to send Dual Tone Multi-Frequency (DTMF) tones in the content of audio exchanges, it is RECOMMENDED to use the XML format specified &xep0181;. However, an implementation MAY also support native RTP methods, specifically the "audio/telephone-event" and "audio/tone" media types.

When the Jingle RTP content type is accepted via a session-accept action, both initiator and responder SHOULD start listening for audio as defined by the negotiated transport method and audio application format. For interoperability with telephony systems, after the responder acknowledges the session initiation request, the responder SHOULD send a "ringing" message and both parties SHOULD play any audio received.

Support for the Theora codec is RECOMMENDED.

In order to secure the data stream, implementations SHOULD use encryption methods appropriate to the transport method and media being exchanged. Such encryption methods are out of scope for this specification.

This document requires no interaction with &IANA;.

Until this specification advances to a status of Draft, its associated namespaces shall be:

  • urn:xmpp:tmp:jingle:apps:rtp
  • urn:xmpp:tmp:jingle:apps:rtp:errors
  • urn:xmpp:tmp:jingle:apps:rtp-info

Upon advancement of this specification, the ®ISTRAR; shall issue permanent namespaces in accordance with the process defined in Section 4 of &xep0053;.

The following namespaces are requested, and are thought to be unique per the XMPP Registrar's requirements:

  • urn:xmpp:jingle:app:rtp
  • urn:xmpp:jingle:app:rtp:errors
  • urn:xmpp:jingle:app:rtp:info

For each RTP media type that an entity supports, it MUST advertise support for the "urn:xmpp:tmp:jingle:apps:rtp#[media]" feature, where the string "[media]" is replaced by the appropriate media type such as "audio" or "video".

The initial registry submission is as follows.

urn:xmpp:tmp:jingle:apps:rtp#audio Signals support for audio sessions via RTP XEP-0167 urn:xmpp:tmp:jingle:apps:rtp#video Signals support for video sessions via RTP XEP-0167 ]]>

The XMPP Registrar shall include "rtp" in its registry of Jingle application formats. The registry submission is as follows:

rtp Jingle sessions that support media exchange via the Real-time Transport Protocol lossy XEP-0167 ]]>
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Thanks to Milton Chen, Diana Cionoiu, Olivier Crête, Tim Julien, Steffen Larsen, Jeff Muller, Mike Ruprecht, and Paul Witty for their feedback.