The audio problem - a total lack of sound - occurs when attempting to
use pjsip to connect two Beaglebone Black Rev-C boxes on a LAN, running
Debian from microSD cards. There is no NAT issue since both are inside
the same LAN.
Orange Box:
===========
Linux cargo-42-4 3.8.13-bone50 #1 SMP Tue May 13 13:24:52 UTC 2014
armv7l GNU/Linux
Debian GNU/Linux 7
The IP Address for eth0 is: 192.168.1.32
C-Media USB audio (nameless cheapo)
Transparent Box:
================
Linux cargo-159 3.8.13-bone79 #1 SMP Tue Oct 13 20:44:55 UTC 2015 armv7l
GNU/Linux
BeagleBoard.org Debian 7.9 Image 2015-11-12
The IP Address for eth0 is: 192.168.1.15
C-Media USB audio (Syba)
Both boxes can record ('arecord') and play ('aplay') audio properly over
the USB dongle, which is the default audio device (.asoundrc ==
'pcm.!default sysdefault:Device'). HDMI audio is fully disabled via
uEnv.txt.
The problem is that when connected via pjsip 2.4.5, no sound is
communicated, and no port 4000 UDP packets are transferred. A pjsua
recording test yields an empty wav file.
Test:
=====
I run pjsua on both boxes, then initiate a call from Orange Box to
Transparent Box. Some errors occur when opening the sound device (but
maybe they're legitimate codec-match failures). The next console
snippet shows all the errors that occur (captured from Orange Box, but
both boxes behave the same).
=====
11:06:39.410 pjsua_aud.c .Set sound device: capture=-1, playback=-2
11:06:39.411 pjsua_app.c ..Turning sound device ON
11:06:39.412 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, &hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
Expression 'PaAlsaStream_Configure( stream, inputParameters,
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
&outputLatency, &hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
11:06:39.430 pjsua_app.c ..Turning sound device ON
11:06:39.432 pjsua_aud.c ..Opening sound device PCM at 44100/1/20ms
11:06:39.489 ec0x236770 ...AEC created, clock_rate=44100,
channel=1, samples per frame=882, tail length=200 ms, latency=0 ms
=====
Some INVITES later, the media negotiation looks good:
=====
11:06:45.382 pjsua_core.c .RX 885 bytes Response msg
200/INVITE/cseq=22258 (rdata0x24cebc) from UDP 192.168.1.15:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.32:5060;rport=5060;received=192.168.1.32;branch=z9hG4bKPjl0nsfaVcpIDXP9f9jS1bzxPnA.zg2QjS
Call-ID: Odd58LgI.B2F2WUfW.RmbTox2CXzpAK7
From: <sip:192.168.1.32>;tag=-HgPab8XCkf0D3cnQxggXsmhX-IJaFur
To: <sip:192.168.1.15>;tag=BV9GhxaJGo5ihz.yt5SbQHkX3cbMmCJV
CSeq: 22258 INVITE
Contact: <sip:192.168.1.15:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 275
v=0
o=- 3660836799 3660836800 IN IP4 192.168.1.15
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 96
c=IN IP4 192.168.1.15
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.1.15
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
--end msg--
11:06:45.407 pjsua_app.c .....Call 0 state changed to CONNECTING
11:06:45.410 pjsua_media.c .....Call 0: updating media..
11:06:45.411 pjsua_aud.c ......Audio channel update..
11:06:45.414 strm0x2d6034 .......VAD temporarily disabled
11:06:45.419 strm0x2d6034 .......Encoder stream started
11:06:45.420 strm0x2d6034 .......Decoder stream started
11:06:45.424 pjsua_media.c ......Audio updated, stream #0: speex
(sendrecv)
11:06:45.426 pjsua_app.c .....Call 0 media 0 [type=audio], status is
Active
11:06:45.428 pjsua_aud.c .....Conf connect: 4 --> 0
11:06:45.429 conference.c ......Port 4 (sip:192.168.1.15)
transmitting to port 0 (USB PnP Sound Device: USB Audio (hw:0,0) (44KHz))
11:06:45.431 pjsua_aud.c .....Conf connect: 0 --> 4
11:06:45.432 conference.c ......Port 0 (USB PnP Sound Device: USB
Audio (hw:0,0) (44KHz)) transmitting to port 4 (sip:192.168.1.15)
=====
The boxes are connected but no sound can be heard at either end. Having
started pjsua with '--rec-file mic2.wav', we now connect the microphone
input to the wav file output.
=====
>>> cl
Conference ports:
Port #00[16KHz/20ms/1] USB PnP Sound Device: USB Audio (hw:0,0) (44KHz)
transmitting to: #4
Port #01[16KHz/20ms/1] mic2.wav transmitting to:
Port #02[16KHz/20ms/1] ringback transmitting to:
Port #03[16KHz/20ms/1] ring transmitting to:
Port #04[16KHz/20ms/1] sip:192.168.1.15 transmitting to: #0
>>> cc 0 1
11:07:26.206 pjsua_aud.c !Conf connect: 0 --> 1
11:07:26.207 conference.c .Port 0 (USB PnP Sound Device: USB Audio
(hw:0,0) (44KHz)) transmitting to port 1 (mic2.wav)
Success
=====
We say a few words, then end the recording using 'cd'.
=====
>>> cd
[...]
Disconnect src port # (empty to cancel): 1
From dst port # (empty to cancel):
>>> cd
Conference ports:
[...]
Disconnect src port # (empty to cancel): 0
From dst port # (empty to cancel): 1
11:08:06.483 pjsua_aud.c Conf disconnect: 0 -x- 1
11:08:06.484 conference.c .Port 0 (USB PnP Sound Device: USB Audio
(hw:0,0) (44KHz)) stop transmitting to port 1 (mic2.wav)
Success
=====
The freshly recorded wav file is unexpectedly empty.
=====
-rw-r--r-- 1 bear bear 0 Jan 3 11:06 mic2.wav
A possibly related issue: Subsequently, when attempting to hang up the
call, pjsua hangs (freezes) when closing the USB sound device, so ctrl-C
must be used to terminate it. This also happens when pjsua is run with
no recording option. Note as well that no RTP packets were ever sent.
=====
11:08:30.165 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200
(Normal call clearing)]
11:08:30.167 pjsua_app_comm ......
[DISCONNCTD] To: sip:192.168.1.15;tag=BV9GhxaJGo5ihz.yt5SbQHkX3cbMmCJV
Call time: 00h:01m:44s, 1st res in 5912 ms, conn in 5952ms
#0 audio speex @16kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
RX pt=98, last update:00h:01m:44.744s ago
total 0pkt 0B (0B +IP hdr) @avg=0bps/0bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
TX pt=98, ptime=20, last update:never
total 0pkt 0B (0B +IP hdr) @avg=0bps/0bps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
11:08:30.173 pjsua_media.c ......Call 0: deinitializing media..
11:08:30.179 pjsua_media.c ........Media stream call00:0 is destroyed
11:08:31.176 pjsua_aud.c Closing sound device after idle for 1 second(s)
11:08:31.178 pjsua_app.c .Turning sound device OFF
11:08:31.180 pjsua_aud.c .Closing USB PnP Sound Device: USB Audio
(hw:0,0) sound playback device and USB PnP Sound Device: USB Audio
(hw:0,0) sound capture device
>>>>
^C
=====
The empty recording appears to indicate that audio is not being
received, or not being processed, by pjsip. But normal alsa aplay and
arecord work fine on both boxes.
Any suggestions on how to resolve or further troubleshoot this would be
appreciated.
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