[maemo-developers] [maemo-developers] VoIP through SIP

[From: Johannes Eickhold]

On Fri, 2006-10-06 at 15:27 +0200, Fred Lefévère-Laaoide wrote:
> Frédéric Crozat wrote:  
> > Well, I want to use my ISP (Free.fr) SIP platform, which allows me to
> > call anywhere in France and various countries for free, as if I was
> > calling from home.
> > 
> > They only support g711 vocodeur ATM (because they are using Cirpack PBX)
> > and their cirpack server is also sending dummy "ping" packet (see
> > http://lists.digium.com/pipermail/asterisk-dev/2006-May/021033.html for
> > more info on this, it would be nice to integrate a similar fix in
> > Sofia-SIP).
> 
> Any news on this ?
> Or a how to get started on adding g711 support?

Sofsip-cli is already using G.711 as it's only supported codec. It makes
use of gstreamer to setup a pipeline like this:

dsppcmsrc -> rtppcmupay -> rtpbin

If you look at the produced RTP packets with e.g. wireshark and decode
the UDP packets as RTP traffic you can see they are tagged to carry
G.711 PCMU payload.

For a list of further codecs use

gst-inspect |grep rtp

on the device (you will maybe have to install the gstreamer-tools
package). Be aware that you would currently have to change the code of
sofsip-cli to enable usage of other codecs.

Greets, Jonek