I'm doing some more tests and found some strange
things (most probably my code...). I'm using every
media port with the same parameters (8 kHz, 16 bit,
160 samples per frame). When I connect (dial in) to
the application I hear the welcome wav with no
problems (played when call media gets active). Now if
I try to add the audio stream from one call to the
conference bridge of another call (to put them in
conference for example), the audio in the first call
(the one that I grabbed the audio stream) get really
horrible (slows down, cuts a lot of frames...), even
before I connect both calls in the conference bridge.
If I remove the port from the destination conference
bridge, the original call gets perfect again.
Thats one thing. Another one is if I hangup the call
that I got the audio stream and added to another
conference bridge, I get
../src/pj/os_core_unix.c:1048: pj_mutex_lock:
Assertion `mutex' failed and the application closes.
If I put any call on hold (destroy media), no mater if
it was added to another conference bridge or not, I'll
get a seg fault and the following stack:
8 put_frame_imp() 0x080a1f69
7 put_frame() 0x080a24a0
6 get_frame() 0x080981f3
5 clock_callback() 0x0809a52f
4 clock_thread() 0x080a7f01
3 thread_main() 0x080ebcf7
2 start_thread() 0xb7ebd240
1 clone() 0xb7b4f4ae
My application is object oriented, and I have one
object per call, and this class has one conference
bridge for each call. I'm using a null sound port. I
already tried to change the conference bridge
resampling to see if it changes the quality, but no
difference (as expected, since I'm using all sampling
rates equal). All my call are using pcmu codec. My
conference bridges have 10 max slots in each.
It seems like there is a problem to add the call audio
stream to more than one conference bridge. I may be
missing something here... not sure what.
My system is a Debian Linux 4.0 (Etch) and I'm using
pjsip from 25-10-2007 (I think its r 1522).
Thanks a lot for any help, I'm pretty lost here...
Thiago Paiva Flores
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