[Live-devel] SIP Server

[Bernhard Feiten]

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.
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?

Other problems appear when wishing  to control the stream.
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?

Best regards,
Bernhard


----- 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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