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[Sip] Re: [Sipping] RE: [Sip-implementors] "Early Media"
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: 'sipping@ietf.org'; sip-implementors@cs.columbia.edu
> > 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
> > Sip-implementors@cs.columbia.edu
> > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >
> _______________________________________________
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--
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 : Gonzalo.Camarillo@ericsson.com
Finland http://www.hut.fi/~gonzalo
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