[gst-devel] sip implementation in gst?

[XuYong]

Yesterday I discussed the same topic with my friend who indicated that the model is wrong, as you mentioned in letter. What I  thought was to design a " sip plug-in" under gst framework working together with other existing plug-ins. However, it is not a feasible way to achieve  the target .
Your advice help me a lot, thank you!
----- Original Message ----- 
From: "Kai Vehmanen" <kv2004 at eca.cx>
To: "XuYong" <xuyong_xs at neusoft.com>
Cc: <gstreamer-devel at lists.sourceforge.net>
Sent: Wednesday, December 13, 2006 6:49 PM
Subject: Re: [gst-devel] sip implementation in gst?


> 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


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