Error using with Echo Cancellation Module with VoIP

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Thank Russell for your fast reply,


I use Baresip. As you can check in the following link, Baresip uses the EC module of Pulseaudio


https://github.com/alfredh/baresip/wiki/Using-PulseAudio:-module-echo-cancel


Do you know any other good VoIP client for linux with a low CPU consumption and Echo Canceller inside ?

And what do you think about the error i get ?

Thank you very much in advance



________________________________
De: pulseaudio-discuss <pulseaudio-discuss-bounces at lists.freedesktop.org> en nombre de Dferop Aero <Dferop_Aero at hotmail.es>
Enviado: lunes, 11 de junio de 2018 16:43
Para: pulseaudio-discuss at lists.freedesktop.org
Asunto: [pulseaudio-discuss] Error using with Echo Cancellation Module with VoIP


Hello,


I have always the same error when i use the Echo Cancellation Module.


I get the same error all the time:

May 31 09:30:54 localhost pulseaudio[753]: [alsa-source-CDC PCM Codec-0] module-echo-cancel.c: Playback after capture (-87178), drop sink 1414
May 31 09:30:55 localhost pulseaudio[753]: [alsa-source-CDC PCM Codec-0] module-echo-cancel.c: Playback after capture (-87698), drop sink 1422
May 31 09:30:56 localhost pulseaudio[753]: [alsa-source-CDC PCM Codec-0] module-echo-cancel.c: Playback after capture (-87148), drop sink 1414
May 31 09:30:57 localhost pulseaudio[753]: [alsa-source-CDC PCM Codec-0] module-echo-cancel.c: Playback after capture (-87279), drop sink 1416

It doesn't matter if i change the configuration parameters, i always get the same error.

I thouhgt it was a problem of CPU (A20 dual core 950MHz, performance) so i have one isolated core running only Pulseaudio. In this case, the CPU consumption is only 20%, so now i don't think it is a problem of CPU

As I said, It doesn't matter if i change the configuration parameters, i always get the same error... "drop sink 14xx" (more or less)


I use tsched=0 in default.pa, if i set tsched=1 the error increases

For the rest of parameters, same error. I have checked it with webrtc, adrian and speex.

Besides I have set different values for the buffer (size and mseconds) in daemon.conf


Because of this error, it seems that the audio is not fluid, the audio is choppy


I hope you can help me,


Thank you very much in advance,


Paul
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