From 0635a307b2bc0a87147c682b3b5e51ae641e3756 Mon Sep 17 00:00:00 2001 From: Peter Saint-Andre Date: Wed, 24 Sep 2008 22:35:09 +0000 Subject: [PATCH] 0.24rc1 git-svn-id: file:///home/ksmith/gitmigration/svn/xmpp/trunk@2259 4b5297f7-1745-476d-ba37-a9c6900126ab --- xep-0167.xml | 716 +++++++++++++++++++++++++++++---------------------- 1 file changed, 412 insertions(+), 304 deletions(-) diff --git a/xep-0167.xml b/xep-0167.xml index 8a4af162..7a54464a 100644 --- a/xep-0167.xml +++ b/xep-0167.xml @@ -25,6 +25,22 @@ &stpeter; &seanegan; &robmcqueen; + &diana; + + 0.24 + 2008-09-24 + psa/dc + +
    +
  • Defined handling of early media.
    • +
  • Clarified handling of SRTP negotiation.
    • +
  • Defined invalid-crypto error condition.
    • +
  • Changed DTMF text to prefer native RTP methods and not recommend sending of DTMF in the XMPP signalling channel.
    • +
  • Modified namespaces to incorporate namespace versioning.
    • +
  • Cleaned up XML schemas.
    • +
+ + 0.23 2008-07-31 @@ -202,7 +218,7 @@

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

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

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

    • -

  • 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.

    • +

  • 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 and the media negotiated is not unduly heavy (e.g., it might be possible to use a reliable transport for audio, but not for video).

  • Content is to be sent and received as follows:

      @@ -215,9 +231,9 @@

      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.

      - +

      The application format consists of one or more encodings contained within a wrapper <description/> element qualified by the 'urn:xmpp:jingle:apps:rtp:0' namespace &VNOTE;. 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.

      + @@ -229,10 +245,10 @@ - ]]>       -

      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 &DESCRIPTION; element is intended to be a child of a Jingle &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", where the media type SHOULD be as registered at &ianamedia;.

      +

      The encodings SHOULD be provided in order of preference by placing the most-preferred payload type as the first &PAYLOADTYPE; child of the &DESCRIPTION; element and the least-preferred payload type as the last child.

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

      @@ -279,7 +295,7 @@

      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:

      +

      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 can 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:

      ]]> @@ -312,12 +328,12 @@ Initiator Responder id='jingle1' to='juliet@capulet.com/balcony' type='set'> - + action='session-initiate' initiator='romeo@montague.net/orchard' sid='a73sjjvkla37jfea'> - + @@ -325,7 +341,7 @@ Initiator Responder - + @@ -339,23 +355,23 @@ Initiator Responder type='result'/> ]]>

      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.

      +

      In the following example, we imagine that the responder supports Speex at clockrate of 8000 but not 16000, G729, and PCMA but not PMCU. Therefore the responder returns only two payload types (since PMCA was not offered).

      - - + - +

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

        -
      • If the value of the 'senders' attribute is "initiator" then the initiator may use any codec that it can send and the responder can receive.
        • -
      • If the value of the 'senders' attribute is "responder" then the responder may use any codec that it can send and the initiator can receive.
        • -
      • If the value of the 'senders' attribute is "both" then the parties may use any codec that both parties can send and receive.
        • +
      • If the value of the 'senders' attribute is "initiator" then the initiator MAY use any codec that it can send and the responder can receive.
        • +
      • If the value of the 'senders' attribute is "responder" then the responder MAY use any codec that it can send and the initiator can receive.
        • +
      • If the value of the 'senders' attribute is "both" then the parties MAY use any codec that both parties can send and receive.
       @@ -393,61 +409,105 @@ Initiator Responder ]]> -

      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.

      +

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

      For example, consider the following static payload-type:

      - - ]]> + + + + ]]>

      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:

      - - ]]> + + + + ]]>

      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.

      - - - - - ]]> + + + + + + + ]]>

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

      - + ]]>

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

      - - - - - - - - ]]> + + + + + + + + + + ]]>

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

      - + ]]> + + + +

      The term "early media" refers to media that is exchanged before a responder has definitively accepted a session request generated by an initiator. Early media is typically used to send ringing tones and announcements, either via audio streams or Dual Tone Multi-Frequency (DTMF) events.

      +

      In Jingle, the exchange of early media is established through use of the "content-add" action. The session flow is as follows.

      + | + | ack | + |<----------------------------| + | session-info (ringing) | + |<----------------------------| + | ack | + |---------------------------->| + | content-add | + | (early media) | + |<----------------------------| + | ack | + |---------------------------->| + | EARLY MEDIA (RTP) | + |<===========================>| + | session-accept | + |<----------------------------| + | ack | + |---------------------------->| + | AUDIO (RTP) | + |<===========================>| + | session-terminate | + |<----------------------------| + | ack | + |---------------------------->| + | | + ]]>       -

      &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.

      +

      &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:jingle:apps:rtp:0' namespace.

      +

      If the initiator wishes to use SRTP, the session-initiate MUST include at least one <crypto/> element and MAY include 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:

      • crypto-suite -- this maps to the SDP "crypto-suite" parameter and has the same semantics (i.e., it is an identifier that describes the encryption and authentication algorithms).
        • @@ -456,24 +516,24 @@ delivery-method=inline; configuration=somebase16string;
      • tag -- this maps to the SDP "tag" parameter and has the same semantics (i.e., it is a decimal number used as an identifier for a particular crypto element).

      An example follows.

      - - ]]> + ]]>

      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/>.

      + ]]> +

      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 Jingle reason of <general-error/> and an RTP-specific condition of <invalid-crypto/>; see the Scenarios section of this document for examples.

       -

      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.

      +

      Informational messages can 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:jingle:apps:rtp:info:0' 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.

      @@ -481,7 +541,7 @@ delivery-method=inline; configuration=somebase16string; - + @@ -489,7 +549,7 @@ delivery-method=inline; configuration=somebase16string; - + @@ -504,11 +564,12 @@ delivery-method=inline; configuration=somebase16string; id='active1' to='romeo@montague.net/orchard' type='set'> - + action='session-info' initiator='romeo@montague.net/orchard' sid='a73sjjvkla37jfea'> - + ]]> @@ -517,11 +578,11 @@ delivery-method=inline; configuration=somebase16string; id='hold1' to='romeo@montague.net/orchard' type='set'> - - + ]]> @@ -530,11 +591,12 @@ delivery-method=inline; configuration=somebase16string; id='mute1' to='romeo@montague.net/orchard' type='set'> - - + ]]> @@ -543,11 +605,11 @@ delivery-method=inline; configuration=somebase16string; id='ringing1' to='romeo@montague.net/orchard' type='set'> - - + ]]> @@ -555,7 +617,7 @@ delivery-method=inline; configuration=somebase16string; -

      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.

      +

      If an entity supports Jingle RTP session, it MUST advertise that fact by returning a feature of "urn:xmpp:jingle:apps:rtp:0" &VNOTE; in response to &xep0030; information requests.

      ... - - + + ... ]]> -

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

      +

      In order for an application to determine whether an entity supports this protocol, where possible it SHOULD use the dynamic, presence-based profile of service discovery defined in &xep0115;. However, if an application has not received entity capabilities information from an entity, it SHOULD use explicit service discovery instead.

       -

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

      +

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

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

      The session flow is as follows:

      @@ -609,24 +671,24 @@ Romeo Juliet id='jingle1' to='juliet@capulet.lit/balcony' type='set'> - - + - + ]]> - - - + ]]> @@ -651,23 +713,23 @@ Romeo Juliet to='juliet@capulet.lit/balcony' type='result'/> ]]> +

      Now the responder immediately terminates the session.

      +

      Note: It may be wondered why the responder does not accept the session and then terminate. That order would be acceptable, too, but here we assume that the responder's client has immediate information about the responder's free/busy status (e.g., because the responder is on the phone) and therefore returns an automated busy signal without requiring user interaction.

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

      The other party then acknowledges termination of the session:

      - - + - + ]]> - - - + ]]> @@ -760,17 +822,196 @@ Romeo Juliet id='accept1' to='romeo@montague.lit/orchard' type='set'> - - + - + + + + + + + ]]> +

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

      + + ]]> +

      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 terminates the session.

      + + + + + Sorry, gotta go! + + + + ]]> +

      The other party then acknowledges termination of the session:

      + + ]]> +       + +

      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.

      + + + + + + + + + + + + + + + + ]]> +

      To signal that the initiator wishes to use SRTP, the initiator's client includes keying material.

      +

      The responder immediately acknowledges the session initiation request.

      + + ]]> +

      If the keying material is acceptable, the responder's continues with the negotiation. If the keying material is not acceptable, the responder's client terminates the session (and can do so at any time).

      + + + + + + + + + ]]> + + + + + + ]]> + + ]]> +

      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.

      + + + + + + + + + ]]>

      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.

      +

      The parties can continue the session as long as desired.

      Eventually, one of the parties terminates the session.

      - @@ -884,23 +1125,25 @@ Romeo Juliet id='jingle1' to='juliet@capulet.lit/balcony' type='set'> - - + - + - - + + + + @@ -909,7 +1152,7 @@ Romeo Juliet - + @@ -925,11 +1168,11 @@ Romeo Juliet id='ringing1' to='romeo@montague.net/orchard' type='set'> - - + ]]> @@ -945,11 +1188,11 @@ Romeo Juliet id='remove1' to='romeo@montague.lit/orchard' type='set'> - - + ]]> @@ -966,17 +1209,17 @@ Romeo Juliet id='accept1' to='romeo@montague.lit/orchard' type='set'> - - + - + - - + ]]> @@ -1031,11 +1274,11 @@ Romeo Juliet id='active1' to='romeo@montague.net/orchard' type='set'> - - + ]]> @@ -1052,13 +1295,15 @@ Romeo Juliet id='add1' to='romeo@montague.lit/orchard' type='set'> - - - + + + + @@ -1066,7 +1311,7 @@ Romeo Juliet - + @@ -1083,20 +1328,22 @@ Romeo Juliet id='add2' to='juliet@capulet.lit/balcony' type='set'> - - - + + + + - + @@ -1109,19 +1356,19 @@ Romeo Juliet type='result'/> ]]>

      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.

      +

      The parties can continue the session as long as desired.

      Eventually, one of the parties terminates the session.

      - - + I'm outta here! @@ -1134,166 +1381,6 @@ Romeo Juliet type='result'/> ]]>       - -

      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.

      - - - - - - - - - - - - - - - ]]> -

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

      - - ]]> -

      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:

      - - ]]> -       @@ -1303,11 +1390,11 @@ Romeo Juliet

      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.

      +

      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") version deployed in North America and in Japan, and the A-law version deployed in the rest of the world.

       -

      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.

      +

      XMPP applications that use Jingle RTP sessions for voice chat MUST support and prefer native RTP methods of communicating DTMF information, in particular the "audio/telephone-event" and "audio/tone" media types. It is NOT RECOMMENDED to use the protocol described in &xep0181; for communicating DTMF information with RTP-aware endpoints.

      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.

      @@ -1329,32 +1416,29 @@ Romeo Juliet       - -

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

      + +

      This specification defines the following XML namespaces:

        -
      • 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
        • +
      • urn:xmpp:jingle:apps:rtp:0
        • +
      • urn:xmpp:jingle:apps:rtp:errors:0
        • +
      • urn:xmpp:jingle:apps:rtp:info:0
      +

      Upon advancement of this specification from a status of Experimental to a status of Draft, the ®ISTRAR; shall add the foregoing namespaces to the registry located at &NAMESPACES;, as described in Section 4 of &xep0053;.

      +       + +

      If the protocol defined in this specification undergoes a major revision that is not fully backward-compatible with an older version, or that contains significant new features, the XMPP Registrar shall increment the protocol version number found at the end of the XML namespaces defined herein, as described in Section 4 of XEP-0053.

       -

      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".

      +

      For each RTP media type that an entity supports, it MUST advertise support for the "urn:xmpp: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 + urn:xmpp:jingle:apps:rtp:audio Signals support for audio sessions via RTP XEP-0167 - urn:xmpp:tmp:jingle:apps:rtp#video + urn:xmpp:jingle:apps:rtp:video Signals support for video sessions via RTP XEP-0167 @@ -1365,7 +1449,10 @@ Romeo Juliet rtp - Jingle sessions that support media exchange via the Real-time Transport Protocol + + Jingle sessions that support media exchange + via the Real-time Transport Protocol. + lossy XEP-0167 @@ -1380,8 +1467,8 @@ Romeo Juliet @@ -1429,6 +1516,27 @@ Romeo Juliet + + ]]> +       + + + + + + + + + + + + + ]]> @@ -1438,8 +1546,8 @@ Romeo Juliet @@ -1477,6 +1585,6 @@ Romeo Juliet       -

      Thanks to Milton Chen, Diana Cionoiu, Olivier Crête, Tim Julien, Steffen Larsen, Jeff Muller, Mike Ruprecht, and Paul Witty for their feedback.

      +

      Thanks to Milton Chen, Olivier Crête, Tim Julien, Steffen Larsen, Jeff Muller, Mike Ruprecht, and Paul Witty for their feedback.
      Element
      <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.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 audio aspect of a voice+video chat). If no 'name' attribute is included, the recipient MUST assume that all sessions are active.
      <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.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 audio aspect but not the video aspect of a voice+video chat). If no 'name' attribute is included, the recipient MUST assume that all sessions are to be muted.
      <ringing/>