[Live-devel] SIP Server

[Emiliano Parasassi]

> 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!

> 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
> >
> >
> 
> 
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> ----
> 
> 
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