Hi, It has not yet been incorporated to amy draft, but it will be as soon as we find some spare cycles... we are working on it.
Thanks, Gonzalo "Yadavalli, Satyamurthy" wrote: > > I figure out that a discussion on this issue occured earlier with proposals > to use 183 Session Progress etc., > (Subject: "[Sip] Local vs. Remote ringback on 180") > Was a conclusion reached and was it incorporated into any (draft) RFC? > -Satya > > > -----Original Message----- > > From: Yadavalli, Satyamurthy [mailto:syadavalli@;telogy.com] > > Sent: Sunday, November 03, 2002 12:56 AM > > To: '[EMAIL PROTECTED]'; [EMAIL PROTECTED] > > Subject: [Sip-implementors] "Early Media" > > > > > > Hi, > > Would like to know how to achieve the following using any of > > the SIP related > > RFCs: > > > > UA1 invites UA2. Say UA2 generates a local ringing tone (or > > some audio) to > > alert its user. However, UA2 woud like to feed some (ringing) > > audio back to > > UA1 (in addition to the 180 RINGING response). And also say > > UA2 may like its > > early media to be overriding any locally generated ringback > > that UA1 may be > > generating to its user about the call in progress. > > > > This 'early media' may have its own characteristics such as codec > > requirements etc., different from those of the final > > dialogue. How (ofcourse > > via an SDP but attached to which SIP response: 180?) are these to be > > described by UA2 to UA1 so that UA1 may load the necessary > > codecs etc., so > > as to be able to present the early media (to be received from > > UA2) to its > > user. And also how to indicate when the audio path (RTP) for > > this early > > media is to be established/released. > > > > Is there any RFC that describes how to provide this kind of > > 'subdialogue' > > before the actual SIP dialogue is established? A quick glance > > at the UPDATE > > RFC appeared to somewhat support this kind of requirement but > > appeared (at > > first glance) to specify the main dialogue itself and didnot appear to > > clearly say when to establish/release this 'subdialogue'. > > > > Appreciate any help in this regard, > > Thanks, > > Satya > > _______________________________________________ > > Sip-implementors mailing list > > [EMAIL PROTECTED] > > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use [EMAIL PROTECTED] for questions on current sip > Use [EMAIL PROTECTED] for new developments of core SIP -- Gonzalo Camarillo Phone : +358 9 299 33 71 Oy L M Ericsson Ab Mobile: +358 40 702 35 35 Telecom R&D Fax : +358 9 299 30 52 FIN-02420 Jorvas Email : [EMAIL PROTECTED] Finland http://www.hut.fi/~gonzalo _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
