> 2009/7/10 Olivier Crête <olivier.cr...@collabora.co.uk>: >> This is strange, all that FsRtpConference does to audio is to >> encode/decode it. Which codec is used?
Hi, I did some more investigation. It seems that in calls between my client and empathy (or with my client on both sides) over jabber, the codec that is used is "SIREN". On the other hand, calls with empathy on both sides use the "speex" codec, which sounds much more reasonable, and calls between my client and google talk use the "PCMU" codec. I also tried to do some sip calls between my client and ekiga and it works quite well, also using the "PCMU" codec. So, if I understand it right, my problem is that the SIREN codec does not work... I still don't understand though why this codec is used with my client only. What determines the codec that is used and why empathy to empathy calls use speex? Best regards, George PS: Also, for the first problem I mentioned (that calls between my client and gtalk don't last much because audio is cut from my client), it seems that this is a gstreamer bug with alsasrc. I can reproduce it with "gst-launch alsasrc ! alsasink" and it works fine when I use "osssrc". I am going to ask the gstreamer guys about it. Sorry for bugging you about that. ------------------------------------------------------------------------------ Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge _______________________________________________ Farsight-devel mailing list Farsight-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/farsight-devel
