On Jul 28, 2006, at 8:51 PM, ext Ethan Blanton wrote:
> Mark Doliner spake unto us the following wisdom:
>> I'm not very familiar with gstreamer... I thought we are calling
>> from the Gaim core? If all the audio/video stuff goes through the
>> Gaim core,
>> does a plugin still need to be calling gst_init()?
Nope, if USE_GSTREAMER is turned on a plugin doesn't need to call
gst_init(). Otherwise Gst needs to be initialized later (in a plugin).
>> Also, we currently have a simple plugin... I think if we're going
>> to add voice
>> or video for sip it's more likely that we would add it to the
>> protocol than completely replace our existing protocol
> SIP voice and video are completely unrelated to SIMPLE except in that
> they are both negotiated over SIP. While it may make sense to roll
> all SIP communications together into one prpl to take advantage of as
> much shared SIP code as possible, SIMPLE and voice are really two
> different things, and as such two different prpls isn't out of line.
Hmmm.. I should have been more precise :) The point was that we
(Sofia-SIP team) needed a GUI for our stack and we chose to develop
one by utilizing Gaim. As there is no complete media support in Gaim,
yet, we wrote one for our purposes -- without the need of touching or
extending any Gaim APIs. But as/if there is a need for a separate RTP
media plugin I'd be happy to contribute to it.