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George Kiagiadakis wrote:
>> 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.
> 

Hi.. interesting! Well, SIREN is the proprietary audio codec used by
MSN. You have it installed on your system (comes with gst-plugins-bad)
so it takes it, you can give a different priority to your codecs by
using the fs_set_codec_preferences API (iirc). Anyways, SIREN is not
broken, it works quite good (it's used by aMSN with farsight2 and it
works nicely), the sound can become muted or contain noise if the volume
is too high, so maybe try to lower the volume and see if it fixes it.

Hope that helps.
KaKaRoTo

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