[pjsip] Media issue with multiple (parallel) calls

Anshuman S. Rawat arawat at 3clogic.com
Fri Feb 15 02:40:35 EST 2008


Hi Benny,

> You mentioned you use splitcomb for your sound device. Any reason why?
> Are you using multiple sound devices (simultaneously)?

We are using the master clock port for driving the timing which we create in 
pjsua_set_snd_dev() which
is why splitcomb is being used. Using master clock seemed to have solved the 
problem of high source jitter
in the outgoing stream.However, we are again seeing high source jitter (as 
seen in wireshark log). The delta for
outgoing RTP stream is either about 20ms (which is should be) or 10ms. For 
both cases, the size of RTP payload
is the same (160b for G711 and 20b for G729).

>
> Actually the log above doesn't tell us if port 1 has stopped
> transmitting to port 0, so without anything else, audio from port 1
> (the first call) would still be played to speaker, hence you hear the
> interference.
>

Sorry I missed a line there.

19:28:33.171 conference.c Port 1 (sip:+18007860404 at 208.109.178.168) stop 
transmitting to port 0 (Primary Sound Capture Driver)
19:28:33.171 conference.c Port 0 (Primary Sound Capture Driver) stop 
transmitting to port 1 (sip:+18007860404 at 208.109.178.168)
19:28:33.171 strm021ADBC4 Start talksprut..
19:28:33.171 conference.c Port 4 (sip:+18009363500 at 208.109.178.168) 
transmitting to port 0 (Primary Sound Capture Driver)
19:28:33.171 conference.c Port 0 (Primary Sound Capture Driver) transmitting 
to port 4 (sip:+18009363500 at 208.109.178.168)

Will provide more info as when I find something suspicious.

Thanks,
Anshuman





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