Hello,
I have several soft-phone (VoIP) daemons (linphone-daemon) running at
the same time on the same machine and connecting to the same audio card
via PulseAudio Jack sink. They each connect to an Asterisk server on
different ports and do their SIP registrations with Asterisk under
different user names. While the daemons start together with no problems
and with their corresponing SIP phones registered properly, when I start
sending call messages to them, the results can be unpredictable.
Sometimes I can hear concurrent calls together, other times one call
will block the other, and other times all calls will be blocked.
If I have 2 calls running on 2 daemons with one call blocking the other
and I terminate the call that was unblocked and playing through the
speakers, the blocked calls resumes playing but at the point in time
where it should have been playing if it were not blocked. This makes me
think that there might be a problem with PulseAudio Jack sink, but I
don't know.
All concurrent daemons on a call show up in pavucontrol named
"linphone-daemon" and connected to Jack sink in the pull down menu. If
i change the device in pavucontrol of a blocked call to the internal
sound card (from the sink, which is connected to a USB card via Jack),
it does not stop the blockage.
In one attempt to fix, I tried giving each process a different name
with, for example:
bash -c "exec -a linphone-daemon1 linphone-daemon --config
/home/iain/.config/linphone/linphonerc1 --pipe linphone-daemon1.soc" )&
but this didn't change the way each process was named in pavucontrol,
with each process still named linphone-daemon rather than
linphone-daemon1, linphone-daemon2 and linphone-daemon3. The blocks
continue as well.
Might there be some port issue with the sink?
It could of course be a problem relating to my Asterisk configuration,
but I thought i'd ask this question here about the sink in case there
are issues running multiple processes of the same type. I've posted a
related question to a linphone list as well.
Hope someone can help. Thanks, all the best,
Iain
PS. Here's the message I posted to the linphone-users list. It gives
more details. I said in this message that i didn't think the problem is
related to the Jack sink, but i don't know:
Hello,
I need to run several linphone-daemons at once on the one Linux machine
with an Asterisk server. I have set this up so that each daemon loads a
separate config file with a unique UDP/TCP port and SIP username and
corresponding entries in Asterisk's sip.conf. All the daemons use the
PulseAudio Jack sink to enable audio to be played concurrently with that
of other processes.
While each daemon works successfully on its own, the daemons only
sometimes work properly when run together. For example if daemon 1 is on
a call and I send a call message through to daemon 2 to make another
call, the outcome is unpredictable, sometimes both can be heard, other
times the sound of one is blocked and yet other times both audio streams
get jittery.
There are no problems with CPU or memory. Also, i don't believe it is a
problem with the Jack sink as i can run multiple other audio programs
concurrently through the sink successfully, including ones in addition
to the linphone-daemon itself. Also, if i call in from several different
Android phones with SIP software (again Linphone) to extensions in
Asterisk (but not the server-based phones), the sound streams with no
problems.
One difference I see is that the SIP phones all have different IPs,
whereas all the server-based phones are on the same IP. Might this be it?
Can anyone please offer some suggestions?
I'll paste below the script that I use to launch the daemons, a script
to send call messages to the servers and also the entries in Asterisk's
sip.conf.
I've also uploaded a log file of one of the daemons here which includes
log info of an event where the audio stopped (towards the end):
https://ufile.io/088vjf45
Relevant looking messages include:
mediastreamer-message-MSAudio_stream_iterate[0x55d89bd96ff0], local
statistics available:
Local current jitter buffer size: 0.0ms
ortp-warning-ortp_loss_rate_estimator_process 0x55d89bd5c1d0: Suspected
discontinuity in sequence numbering from 503 to 250
ortp-warning-Receiving packet with unknown payload type 0
Hope someone can help,
Iain
#!/bin/bash
# startlinphones.sh
# Start 3 linphone-daemons
process=$(pgrep linphone-daemon)
set -- $process
process=$1
if [ -z $process ]; # to check if at least one daemon is already running
then
echo "Starting linphone-daemon 1. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc1 \
--disable-stats-events \
--log /home/iain/tmp/daemon1.log \
--pipe linphone-daemon1.soc)&
echo "Starting linphone-daemon 1. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc2 \
--disable-stats-events \
--log /home/iain/tmp/daemon2.log \
--pipe linphone-daemon2.soc)&
echo "Starting linphone-daemon 3. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc3 \
--disable-stats-events \
--log /home/iain/tmp/daemon3.log \
--pipe linphone-daemon3.soc)&
else
echo "linphone-daemons already running"
fi
exit 0
--------------------------------------------
#!/bin/bash
# callnumber.sh
# call number on a linphone-daemon
daemonnum=$1
phonenum=$2
if ( [ -z $phonenum ] || [ -z $daemonnum ] );
then
echo "Requires both daemon number and phone number as arguments"
else
process=$(pgrep linphone-daemon)
set -- $process
process=$1 # the 1st number in list
if [ -z $process ];
then
echo "Starting linphone-daemons"
startlinphones.sh
sleep 5
fi
echo "Calling"
$(echo "call sip:$phonenum@127.0.0.1" | \
socat STDIN UNIX-CONNECT:/tmp/linphone-daemon$daemonnum.soc)
fi
exit 0
--------------------------------------------------
excerpt from sip.conf
[daemon1]
type=friend
context=outgoing
host=dynamic
username=daemon1
secret=mypassword
disallow=all
;allow=opus
allow=gsm
allow=ulaw
;direct_media=true
direct_media=false
language=pt_BR
[daemon2]
type=friend
context=outgoing
host=dynamic
username=daemon2
secret=mypassword
disallow=all
;allow=opus
allow=ulaw
;direct_media=true
language=pt_BR
[daemon3]
type=friend
context=outgoing
host=dynamic
username=daemon3
secret=mypassword
disallow=all
;allow=opus
allow=gsm
allow=ulaw
;direct_media=true
direct_media=false
language=pt_BR
----------------------------------
The conf files for each daemon specify unique port numbers
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