On 31/07/14 20:05, Ernesto Celis wrote: > It looks like the audio backend is unable to set the appropriate bit > rate, try building pjsip disabling PortAudio and enabling Alsa > backend instead. Thx Ernesto! I'm looking on ./configure and don't see how to disable portaudio and enable alsa May You hint me how to do this? I tried current pjproject-svn (r4883) build with external portaudio (svn1919). There seems to be small progress, as now log looks following: --------------------------- You can talk! 15:32:28.116 pjsua_aud.c ...Set sound device: capture=0, playback=0 15:32:28.116 pjsua_aud.c ....Opening sound device PCM at 16000/1/20ms Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2714 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838 Expression 'paInvalidSampleRate' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2043 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2714 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838 ---------------------------- I understand above set of "Expression....failed...." is because psjip tries to set sound-dev to PCM at 16000 - and this isn't supported on my hardware. Next I see: ---------------------------- 15:32:28.117 pjsua_aud.c ....Opening sound device PCM at 44100/1/20ms 15:32:28.193 ec0x1a853f0 .....AEC created, clock_rate=44100, channel=1, samples per frame=882, tail length=200 ms, latency=0 ms 15:32:28.198 pjsua_aud.c ...Conf connect: 1 --> 0 15:32:28.198 conference.c ....Port 1 (sip:508047160 at 192.168.1.254) transmitting to port 0 (HDA NVidia: ALC889A Analog (hw:0,0) (44KHz)) 15:32:28.198 pjsua_aud.c ...Conf connect: 0 --> 1 15:32:28.198 conference.c ....Port 0 (HDA NVidia: ALC889A Analog (hw:0,0) (44KHz)) transmitting to port 1 (sip:508047160 at 192.168.1.254) 15:32:28.198 pjsua_core.c ....TX 838 bytes Response msg 200/INVITE/cseq=102 (tdta0x1a72540) to UDP 192.168.1.254:5060: SIP/2.0 200 OK ---------------------------- All above looks all OK for me. Unfortunately local->remote audio still isn't working. Instead of, I have local MIC looped to local speakers. Where issue might be? BTW: Infact I hear local MIC all the time in local speakers - no difference is SIP app launched or not. Is this normal? br
