[I tried to send this message via gmane, but it seems it has not reached the list, so I send it again, with modifications due to new findings] Hello, I am new to VoIP and pjsip and as an exercise I try to understand and extend the pjsua sample application. I am using pjproject-1.3 on Ubuntu 9.10alpha and testing against Asterisk softswitch that is running on another PC on the local net. Now I modified pjsua to to set the playback device depending on the call state. If there is no call, the sound goes to the PC speaker, if there is an active call, it goes to an USB handset, Capture device is always the handset. I change the sound device with pjsua_set_snd_dev(). This basically works, but pjsua segfaults when a call is terminated. This only happens if I set --snd-auto-close to certain values (Bad traces below are with value 0). It seems that close_snd_dev is called from different threads concurrently, one call is coming from the auto close, the other comes from the application due to the pjsua_set_snd_dev() call. For unknown reasons, pjmedia_snd_port_destroy() which is called from close_snd_dev takes more than one second on this system, so the second thread calls close_snd_dev while first thread is still in pjmedia_snd_port_destroy and tries to close the same sound device again-> Crash I have added some additional logging to show the call sequence. The number in <> is the thread id pthread_self(). The logs are attached to this mail to keep the formatting. Are there any restrictions when pjsua_set_sound_dev resp. close_snd_dev can be called? Documentation says "Application may call this function at any time to replace current sound device". Regards, Wolfgang -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log.txt URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090730/3ff2359d/attachment.txt>
