[gst-devel] sip implementation in gst?

[Kai Vehmanen]

Hi,

On Wed, 13 Dec 2006, XuYong wrote:

> Thanks a lot for your reply. I have noticed the libs and 
> clients(sofia,Tapioca,etc). and they are very good indeed. But what I am 
> more interested in is a plug-in support SIP. Browsing the plug-ins list 
> (http://gstreamer.freedesktop.org/documentation/plugins.html ), I did 
> not found a plug-in providing the SIP implementation,while some other 
> network protocols ,such as http, tcp, udp, rtp, rtsp, are already there. 
> Is it a good idea to design a sip plug-in? or any body is already 
> working on it ?

I don't think there are any. And really, what would a SIP gstreamer 
element do? I guess in theory you could have an element that takes a SIP 
address as an object property, and when in PLAYING state, the element 
would establish a call with that address, but how would you handle 
incoming calls (you should ask the user whether to accept or not), deliver 
state changes (the remote is "ringing" now), modify call state (your 
presence info, send DTMF tones, etc)...?

I believe that in majority, if not all, of the cases, the right model is 
to keep the SIP signaling separate, and use the gstreamer 
rtp/tcp/udp/http/etc transport plugins to realize the negotiated media 
streams.

-- 
  links, my public keys, etc at http://eca.cx/kv