Just a quick question:

I didn't quite glean this from the spec and am not sure if it's been discussed in this forum, but is there a way to associate two streams (or two <content /> entities)? Typically, for a video "call", there are two streams, audio and video. You want these two streams associated in the client a) so that they can be presented in an associated way (camera and speaker controls near each other), and b) so that they can be associated for lip sync. Especially if there are two video streams (for example, there's a document camera), you want to know which is the "main" stream that goes (by default) in the main window with the audio controls. Or for that matter, if you only want to allow one video stream, you know which one to do a content-remove on.

Or, is it to be inferred that for a single session, there can be at most one entry for each content type, and that any others would be yet another session (not sure I like that). I have no idea which approach maps better to SIP.

Also, it seems to me that, although "ringing" and "hold", would typically be associated with a session, I could see how "mute" would be associated with individual streams (<content/>). I may be in a voice-video session, but temporarily want to mute only video, because I need to pick my nose, or scratch an intimate area, or whatever, and then un-mute again. Otherwise, how would session-mute be different than session-hold? Perhaps <mute /> could include an optional "name" property which, if present, specified the name of a particular <content /> entity???

Thanks for listening,
Jeff

"XMPP Extensions Editor" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
Version 0.20 of XEP-0167 (Jingle RTP Sessions) has been released.

Abstract: This specification defines a Jingle application type for negotiating a session that uses 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.

Changelog: 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. (psa)

Diff: http://is.gd/r20

URL: http://www.xmpp.org/extensions/xep-0167.html




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