No audio when call is established

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Hi Chris,

ok, in this case it seems to be related with your special 
audio-configuration.
Some experiments with the build might help here. On some Ubuntu-systems 
for example, I had to compile without the included 
portaudio-implementation due to some conflicts between this 
portaudio-impl & pulseaudio. Therefore I had to insert the following 
lines into the file pjlib/include/pj/config_site.h
   #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0
   #define PJMEDIA_AUDIO_DEV_HAS_ALSA 1
furthermore you may want to disable all unnecessary options (see 
./configure --help) in the build-configuration. Watch out for these 
*config*.h where You can also find several hidden switches.

But before trying out lots and lots of configurations, You can try a few 
tests, to narrow down the cause:
- run a testprogram, that uses the PJSUA-ToneGenerator - do you hear 
these local generated tones?
- make a call and measure the RTP-traffic with wireshark: is there any 
data transmitted? Empty (silent) audio-packets? Or nothing (no packets)?


Greetings
Niels Klaas


>  Thanks Niels.
> 
>  I'm running on an Arm based system running a Debian variant.? There
> is no on board audio, so I have a Sabrent USB sound card which shows
> up as a C-Media Electronics audio adapter.? I've got sound working
> and can play audio as well as record via the USB sound card.? The
> system only has one network adapter (eth0).? IPtables is not
> running.? When a call is established, no audio is heard on either
> end.? DTMF from the other end is detected by pjsip.? The call is
> routing through an Asterisk server.? Asterisk shows no errors or
> warnings.
> 
>  On Mon, Jul 27, 2015 at 3:06 AM, Niels Klaas PJSIP_ML
> <pjsip.org at nylz.de> wrote:
> 
> Hi Chris,
> 
>  I had the same problem several times, caused by various reasons.
>  Last time I had this problem, the PJSIP-RTP-routing was the cause.
>  Short description:
>  Weird behaviour: I call someone, the callee hears me, but I don't
> hear the callee. Problem not reproducable on every workstation.
>  A lot of investigation resulted in the following:
>  My RTP-Stream out (mic) went out through wlan0
>  RTP-Stream in (speaker) came in through lan0 (eth), while pjsip was
> listening on the other adapter. 8-/
>  This was related to PJSIPs way of determination of the routing. It
> wasn't compatible with our infrastructure.
>  Quick fix: pull the lan0 - plug!
>  Persistant fix - a patch is under way.
> 
>  if this was not your problem (it might help someone else), You may
> want to give some more details about your system. this audio stuff is
> a world of it's own.
> 
>  greetings
>  Niels Klaas
> 
> Date: Wed, 22 Jul 2015 08:43:33 -0500
>  From: Christopher Miller <chris.miller@xxxxxxxxx>
>  To: pjsip at lists.pjsip.org
>  Subject: No audio when call is established
>  Message-ID:
>  ? ? ? ? ? ? ? ?
> ?<CALPDPXh=ZsgZF4FsR0iWJSk2xZAUVCZZYq4E25dFO2_7Y0SpXQ at mail.gmail.com>
>  Content-Type: text/plain; charset="utf-8"
> 
>  I've verified I have working speaker and microphone levels and
> pjsystest
>  plays and records audio successfully.
> 
>  Sip registration succeeds but I get no audio when a call is
> established.
>  This is what I see:
> 
>  14:46:35.894? ? pjsua_app.c? ...Call 0 state changed to CONFIRMED
>  14:46:36.099 os_core_unix.c !Info: possibly re-registering existing
> thread
> 
>  After that, no activity until the call is ended.
> 
>  In my past experience, I've not received that Info line about
>  re-registering existing thread.? Is that telling me there's
> something
>  wrong?
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