I building a small streaming application based on PJSIP 2.5 stack. At first I was little bit confused by the way of using it with Qt projects as a VC compiled library. I was able to rebuild whole library with Qt+mingw for Windows.
My app is now working fine, except RTP timing has some jitter - packets dalay vary from 10 to 30 ms. It is not bad at all, but my VoIP provider suggesting to fix it at 20ms. I found solution in your FAQ at https://trac.pjsip.org/repos/wiki/FAQ#tx-timing with help of splitcomb and master port. This solution gives perfect timing. Delays are now mostly in range 19..21 ms.
First problem was - SEGFAULT after hangup because clock thread was not closed properly. I was able to fix it by adding the following lines in the pjsua_aud.c, function close_snd_dev(void), after line #1927:
if( m )
{
pjmedia_master_port_stop( m );
pjmedia_master_port_destroy( m, PJ_TRUE );
m = NULL;}
One problem I am facing now - I got a SEGFAULT when I try to make a second call after first one is over. Here is a screenshot of the faulted call to pj_mutex_lock( conf->mutex ); in conference.c, line #765: https://s31.postimg.org/hyg27c5p7/image.png
Can somebody describe correct procedure of closing such kind of calls? How to fix this problem?
Thanks!
Kirill
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