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 _______________________________________________ maemo-developers mailing list maemo-developers@maemo.org https://maemo.org/mailman/listinfo/maemo-developers
