On 10/26/05, Greg Herlein <[EMAIL PROTECTED]> wrote:
> > Hi, please have a look at http://tapioca-voip.sourceforge.net. It is a
> > work in progress.
> > And also a short description of how it works at:
> > http://marciom.blogspot.com/2005/10/voip-with-tapioca.html
>
> I like Tapioca's distributed approach - nice use of DBus IMHO.
> :)
>
> I'm really curious though, why you would use GStreamer for VoIP -
> you say on your website that:
>
> "Many of the existent open source VoIP has a media subsystem of
> its own, which means that transport protocols and codec
> implementation are inside the application's code. That' s why I
> believe that a great feature in Tapioca is that it uses gstreamer
> to provide audio codecs, device handling and RTP streaming. This
> gives a lot of extensibility to the multimedia layer."
>
> I'm not sure that makes sense to me since on the n770 the DSPs
> are sinks themselves - at least if this page:
>
> http://maemo.org/pipermail/maemo-developers/2005-September/001217.html
>
> is correct. No other open source telephony apps use GStreamer
> (that I know of) and it looks like you'd have to hack it to make
> it work on an n770 and a desktop from the same code base...
> assuming you had software plugins for the codecs for GStreamer
> (do those exist?).
>
> I do like getting the RTP framework of GStreamer, but for VoIP
> that seems like a heavy-weight option just for RTP.
>
> Thoughts?
I'm not sure, but seems that they use Gstreamer's pipeline, so you can
do something like:
source -> decoders={ speex, h323, ... } -> sink={alsa, oss, ... }
AFAIU Tapioca would do the SIP connection setup and then assemble this
pipeline for you.
Maemo have a special sink called mp3sink that short circuit
decoder=mp3 and sink=alsa(?), that avoids calls back to main ARM cpu,
consuming less cycles.
--
Gustavo Sverzut Barbieri
---------------------------------------
Computer Engineer 2001 - UNICAMP
GPSL - Grupo Pro Software Livre
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