Hi everyone!
I test a scenario where I register with:
./pjsua --id sip:myNumber at myims.com --registrar sip:myims.com --proxy sip:anIP --realm myims.com --username user --password pass
So as long as I am registered, I want to create a call to someone (it is an auto-answer number) and play a .wav file using the command:
./pjsua --id sip:myNumber at myims.com sip:NumberToCall at myims.com --proxy sip:anIP --play-file=applause.wav
After I take the tcpdump, I see only the RTP flow of the auto-answer, and the 1 sec of silence created by my own device (after that VAD voice detector takes over).
No RTPs with applause going out of my device!
I see these logs if any helpful:
pa_dev.c ..PortAudio sound library initialized, status=0
12:18:35.779 pa_dev.c ..PortAudio host api count=2
12:18:35.779 pa_dev.c ..Sound device count=4 ---> I have a dummy sound card, it should be enough!
wav_player.c .File player 'applause.wav' created: samp.rate=44100, ch=1, bufsize=4KB, filesize=534KB
12:18:35.794 pjsua_aud.c .Player created, id=0, slot=1
.Set sound device: capture=-1, playback=-2
12:18:35.796 pjsua_app.c ..Turning sound device ON
12:18:35.797 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms
12:18:35.799 ec0xa183290 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms
12:18:35.800 pjsua_media.c .Call 0: initializing media..
12:18:35.800 pjsua_media.c ..RTP socket reachable at 192.168.1.79:4000
12:18:35.800 pjsua_media.c ..RTCP socket reachable at 192.168.1.79:4001
12:18:35.800 pjsua_media.c ..Media index 0 selected for audio call 0
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