[Live-devel] SIP Server

[Bernhard Feiten]

> do you distinguish between audio and video ports in sdp media
> announcement (m=audio.. and m=video...) ?
> Now, with new API of ServerMediaSubsession written by Ross, is possible
> to get much information for the SDP negotiation!
>
Yes I distinguish audio and video. But I did not yet resolve how to map this
information on the tracknames of the subsession.
So far the tracks are named automatically track1, track2, ....  It is
assumed that track1 is audio, if two tracks are present.

> Sorry for my ignorance, but REINVITE and REFER are SIP commands?
> I've studied rfc3261...
I learned about SIP also very recently. To my knowledge is REFER a SIP
command which is under consideration as extension (perhaps already a RFC;
I'm not sure). REINVITE should be written reINVITE.



----- Original Message -----
From: "Emiliano Parasassi" <millallo at tiscali.it>
To: "LIVE.COM Streaming Media - development & use" <live-devel at ns.live.com>
Sent: Thursday, April 14, 2005 11:55 AM
Subject: Re: [Live-devel] SIP Server


> > Dear Emiliano, dear Ross, dear all,
> >
> > I'm also developing a SIP server for live.com (derived from the
RTSPServer.
> > First version is already running,
> > I will study your code and integrate in my code.
> > My code is also quite simple so far. It uses also the Call-ID to
initiate or
> > find the appropriate subsession.
> > The client port is parsed from the SDP from the INVITE command.
>
> do you distinguish between audio and video ports in sdp media
> announcement (m=audio.. and m=video...) ?
> Now, with new API of ServerMediaSubsession written by Ross, is possible
> to get much information for the SDP negotiation!
>
Yes I distinguish audio and video. But I did not yet

> > The stream is started in the ACK.
> >
> > One question is how to name the stream. "sip:hostname/streamname" is not
> > compliant with SIP protocol.
> > Our idea is to use sip:streamname at hostname. Each stream is handled like
a
> > user. The stream then can als be registered in a proxy. Are there any
other
> > ideas?
>
> I also think that the format sip:streamname at hostname is the better
> choice.
>
> >
> > Other problems appear when wishing  to control the stream.
>
> ...mmmmm...
>
> > E.g. to put the stream on hold. A proposal would be to do a REINVITE
with
> > client port set to zero.
> > With this also the audio and video could be paused seperately.
> > Another issue we want to study is the session transfer to another client
> > terminal.
> > The REFER commands handles the handover on terminal level. The server
will
> > get another INVITE from the other terminal. As the Call-ID is identical
(or
> > should be made identical), the server can identify the subsession. The
idea
> > is to redirect the RTP streams to the other terminal then. Do you think
this
> > is feasable?
>
> Sorry for my ignorance, but REINVITE and REFER are SIP commands?
> I've studied rfc3261...
>
>
> >
> > Best regards,
> > Bernhard
> >
>
> Greetings
> Emiliano
>
>
> >
> > ----- Original Message -----
> > From: "Emiliano Parasassi" <millallo at tiscali.it>
> > To: "LIVE.COM Streaming Media - development & use"
<live-devel at ns.live.com>
> > Sent: Wednesday, April 13, 2005 12:32 PM
> > Subject: [Live-devel] SIP Server
> >
> >
> > > Hi Ross,
> > > I've stared developing a SIPServer too, but i stopped it because i
> > > haven't enough time.
> > > It is not finished, and i copied and pasted some code from
> > > RTSPServer. ;)
> > > It behaves like a stateless Server, and also it misses SDP
negoziation.
> > > Server recognizes clientSession by Call-ID, and then passes the
> > > datagram request to the right ClientSession.
> > >
> > > I hope this 'simple' SIPServer helps you.
> > >
> > > Bye
> > > Emiliano
> > >
> > >
> >
> >
>
> --------------------------------------------------------------------------
--
> > ----
> >
> >
> > > _______________________________________________
> > > live-devel mailing list
> > > live-devel at lists.live.com
> > > http://lists.live.com/mailman/listinfo/live-devel
> > >
> >
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>
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