[Adium-devl] about SIP plugin

[Alvaro Saurin]

Hi,

I think a SIP plugin could easily use the new QuickTime plugin. I  
also think that Gstreamer should be avoided for the RTP stuff:  
instead of using it, a SIP framework should be used for the  
negotiation and then everything else should be done at the Adium  
level when the SDP negotiation has been established.

I think the best solution would be to develop a SIP service that  
launches the negotiation with the SIP code for Gaim (or any other SIP  
framework) and, when the payload and transport ends are obtained, it  
asks for a RTP connection to the Adium videoconferencing controller.  
This is what I'm doing with Jingle: Smack handles the negotiation,  
and the result is used by the "Smack service" (in adium/Plugins/Smack  
Service/SmackJinglePlugin.[hm]) for requesting a RTP audio connection  
to the controller (in adium/Source/AIVideoConfController.[hm]).

Alvaro


On 27 Aug 2006, at 15:40, Evan Schoenberg wrote:

> Martti,
>
> This is very exciting :)
>
> If you have further questions, don't hesitate to drop us an email  
> or find someone via IM or IRC.  As part of the Google Summer of  
> Code, Alvaro Saurin is creating an RTP Quicktime-based framework  
> within Adium which is to be used by audio/video supporting  
> protocols.  It's stil in -progress.  I'm cc'ing him on this email;  
> the two of you should chat about your SIP plugin's RTP needs and,  
> perhaps, how existing work could help in its implementation.
>
> Cheers,
> Evan
>
> On Aug 25, 2006, at 5:34 AM, Martti Mela wrote:

-- 
Alvaro Saurin <alvaro.saurin at gmail.com> <saurin at dcs.gla.ac.uk>