Hi Chris,
ok, in this case it seems to be related with your special
system-configuration. AFAIK Audio on ARM / Debian works in most cases,
because I tried already some Raspbian-Experiments. So 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 portaudio & pulseaudio. So there are many switches in
the make-step:
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
and furthermore my build-command looked finally like that:
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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