New subject: Broken silence with webrtc

1 Sep 2015

      Ok so on a freshly built Debian install with FS 1.7 I get the exact same
behaviour.

I managed to stop the broken silence when using the originate command by
adding absolute_codec_string=PCMA. My thoughts were it could be a
transcoding thing but inbound calls from the Verto client (which have no
issue) result in opus > g722 so cant be.

The intermittent audio on verto calls is still a problem though. The most
common issue is one way audio from the verto client. The recording has no
audio from the verto side either so I guess the RTP stream isn't being sent.

Does anyone have any pointers to where I can look when I reproduce the
fault? It takes a while for the issue to occur but once I have the call
active I need to find where in the chain it is falling down. No errors are
reported in the Chrome or FS console.

Thanks

On Tue, Sep 1, 2015 at 4:36 PM, <
freeswitch-users-request@lists.freeswitch.org> wrote:
...
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Today's Topics:
1. Re: Broken silence with webrtc (Anthony Minessale)
---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 1 Sep 2015 10:35:30 -0500
Subject: Re: [Freeswitch-users] Broken silence with webrtc
Your findings contradict each other so much, I recommend you start over
from scratch.
Backup your configs.  update to the latest master version of FS.  If you
are on 1.4, nothing new will be done to mitigate webrtc issues.  Set up a
box with the default configurations and retest.
you may also want to try putting <X-PRE-PROCESS cmd="set" data=
"suppress_cng=true"/>  in vars.xml
On Tue, Sep 1, 2015 at 10:25 AM, Ken Rice <krice@freeswitch.org> wrote:
...
Just enable verto debugging in verto.conf.xml in your configs… it’ll
print it right to the screen
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto:
freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Gary
Foreman
*Sent:* Tuesday, September 1, 2015 10:13 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Broken silence with webrtc
Ok so the issue has been superseded by intermittent one-way / no audio.
I'm getting it very intermittently (1 in every 30 calls or so) but I'm
struggling to debug it as the traffic is encrypted and wireshark doesn't
see it as rtp stream.
Where is the best place to start debugging verto? I was previously using
sip.js without any audio issues so it seems to be verto specific.
On Tue, Sep 1, 2015 at 12:04 PM, <
freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Broken silence with webrtc (Stanislav Sinyagin)
---------- Forwarded message ----------
From: Stanislav Sinyagin <ssinyagin@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 1 Sep 2015 13:03:25 +0200
Subject: Re: [Freeswitch-users] Broken silence with webrtc
is it running on a virtual machine?
I found a strange effect that I could only reproduce in a VM, and never
on physical hardware:
https://freeswitch.org/jira/browse/FS-7805
under certain load, an originate command triggers a continuous distortion
in another, running and unrelated, channel.
It seems to be triggered by insufficient CPU resource at the moment of
the origination.
On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman@gmail.com>
wrote:
I've found that it occurs after any bridge, its not specific to the
originate command.
Would you require a wireshark trace or the output of the freeswitch
console?
The scenario below reproduces the issue ...
Test extension
<extension name="Inbound_Routing">
<condition field="destination_number" expression="^2003$">
<action application="export" data="dialed_extension=$1"/>
<action application="export" data="transfer_ringback=$${uk-ring}"/>
<action application="set" data="RECORD_TITLE=Title goes here"/>
<action application="set"
data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
<action application="export" data="send_silence_when_idle=false"/>
<action application="export" data="bridge_generate_comfort_noise=false"/>
<action application="export" data="rtp_enable_vad_in=false"/>
<action application="export" data="rtp_enable_vad_out=false"/>
<action application="export" data="fire_talk_events=true"/>
<action application="export" data="fire_not_talk_events=true"/>
<action application="export" data="recording_id=${uuid}"/>
<action application="set" data="media_bug_answer_req=true"/>
<action application="answer" />
<action application="playback"
data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
</condition>
</extension>
I originate a call from a polycom handset using g722 to the extension
above.
I originate a call using the verto client to the extension above.
I get the uuids of the channels using show channels and use uuid_bridge
[uuid1] [uuid2] to merge the channels.
On Tue, Sep 1, 2015 at 11:30 AM, <
freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. How send in To header anonymous and in P-Asserted-Identity
      caller-id-number (Alex Polischuk)
   2. Loop play while wait DTMF digit (Dmitry Mordovin)
   3. Broken silence with webrtc (Gary Foreman)
   4. Re: Broken silence with webrtc (Brian West)
   5. SIP profile not loading (Chris Young)
   6. Playing with conditions (Dmitry Mordovin)
   7. Re: Playing with conditions (Avi Marcus)
---------- Forwarded message ----------
From: Alex Polischuk <alxpol@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Mon, 31 Aug 2015 17:02:25 +0300
Subject: [Freeswitch-users] How send in To header anonymous and in
P-Asserted-Identity caller-id-number
Hi all,
How I can define different users and domains in To and P-Asserted-Identity
headers?
Thanks,
Alex
---------- Forwarded message ----------
From: Dmitry Mordovin <d.mordovin@dwide.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Mon, 31 Aug 2015 18:24:08 +0400
Subject: [Freeswitch-users] Loop play while wait DTMF digit
Hello
This example play conf-pin.wav and wait DTMF.
<extension name="play_and_get_digits example">
<condition field="destination_number" expression="^(1888)$">
<action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/>
<action application="log" data="CRIT ${foobar}"/>
</condition>
</extension>
Is it possible to play WAV file in infinity loop and wait user DTMF?
And how can I check DTMF input after user entered DTMF?
In dialpeer, like.
If ${DTMF} = 1 then bridge to XXX
If ${DTMF} = 2 then play file and finish session
Thank you.
Dmitry
---------- Forwarded message ----------
From: Gary Foreman <gaz.foreman@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Cc:
Date: Mon, 31 Aug 2015 20:36:59 +0100
Subject: [Freeswitch-users] Broken silence with webrtc
Hi all,
Hoping someone can point me in the right direction because after several
hours I'm out of ideas.
I'm having an issue where the 2nd leg of a call that is bridged to a
verto endpoint has broken silence. When there is no sound on the line the
audio goes completely silent (all background noise is dropped) but around
every third of a second it repeatedly cuts back in for a fraction, then
goes completely silent again. This only happens when the call is created
using the originate command.
Can anyone give me an idea of where to look next? I have a wireshark
trace that during playback shows the audio cutting out periodically on the
2nd leg during periods of silence. The 1st leg is using webrtc encryption
and I cant decode the stream.
Thanks in advance!
---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Mon, 31 Aug 2015 14:40:17 -0500
Subject: Re: [Freeswitch-users] Broken silence with webrtc
Please provide logs and samples of how you originate this, sounds like
Voice Activity Detection possibly.
On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman@gmail.com>
wrote:
Hi all,
Hoping someone can point me in the right direction because after several
hours I'm out of ideas.
I'm having an issue where the 2nd leg of a call that is bridged to a
verto endpoint has broken silence. When there is no sound on the line the
audio goes completely silent (all background noise is dropped) but around
every third of a second it repeatedly cuts back in for a fraction, then
goes completely silent again. This only happens when the call is created
using the originate command.
Can anyone give me an idea of where to look next? I have a wireshark
trace that during playback shows the audio cutting out periodically on the
2nd leg during periods of silence. The 1st leg is using webrtc encryption
and I cant decode the stream.
Thanks in advance!
_________________________________________________________________________
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---------- Forwarded message ----------
From: Chris Young <Chris.Young@enghouse.com>
To: "freeswitch-users@lists.freeswitch.org" <
freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 1 Sep 2015 07:31:00 +0000
Subject: [Freeswitch-users] SIP profile not loading
Hi all,
Recently, we've begun experiencing a strange problem whereby the first
SIP profile to be loaded gets 'stuck' and never actually completes its
initialisation. This always seems to affect the first profile only, so if I
have profiles named (for example):
dummy.xml
external.xml
internal.xml
then 'dummy' would fail to load but 'external' and 'internal' would be
fine. No error messages are output to the logs but preliminary
investigation suggests that the profile thread is becoming blocked for some
reason. The specified IP address is valid and available and there are no
other processes using the requested port. FreeSWITCH comes up successfully
but 'sofia status' shows only 'external' and 'internal'. At this point, I
can use 'sofia profile dummy start' and the profile loads correctly so it
appears to be valid.
Has anybody else seen this kind of behaviour or know what could be
causing it?
Many thanks,
Chris
---------- Forwarded message ----------
From: Dmitry Mordovin <d.mordovin@dwide.com>
To: "freeswitch-users@lists.freeswitch.org" <
freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 01 Sep 2015 12:56:08 +0400
Subject: [Freeswitch-users] Playing with conditions
Hello
<section name="dialplan" description="Local call">
    <context name="public">
      <extension name="local_1">
<condition field="destination_number" expression="^(\d{4})$">
      <action application="answer"/>
<action application="play_and_get_digits" data="1 1 3 7000 #
$${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
foobar \d+"/>
      <action application="log" data="CRIT ${foobar}"/>
<condition field="${foobar}" expression="1$">
            <action application="log" data="CRIT
--------------1------------------"/>
            <anti-action application="log" data="CRIT --------------not
1------------------"/>
      </condition>
<action application="log" data="CRIT ---- last ${foobar}"/>
    </condition>
</extension>
    </context>
  </section>
Why inner condition not works?
---------- Forwarded message ----------
From: Avi Marcus <avi@avimarcus.net>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Tue, 1 Sep 2015 10:30:00 +0000
Subject: Re: [Freeswitch-users] Playing with conditions
Short answer: Each extension only has 1 set of conditions.
The condition evaluating foobar is run *before* it gets set.
After the play_and_get_digits you should transfer/execute_extension to a
new extension that will evaluate ${foobar}
-Avi Marcus
BestFone
On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin@dwide.com>
wrote:
Hello
<section name="dialplan" description="Local call">
     <context name="public">
       <extension name="local_1">
<condition field="destination_number" expression="^(\d{4})$">
       <action application="answer"/>
<action application="play_and_get_digits" data="1 1 3 7000 #
$${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav
/invalid.wav foobar \d+"/>
       <action application="log" data="CRIT ${foobar}"/>
<condition field="${foobar}" expression="1$">
             <action application="log" data="CRIT
--------------1------------------"/>
             <anti-action application="log" data="CRIT --------------not
1------------------"/>
       </condition>
<action application="log" data="CRIT ---- last ${foobar}"/>
     </condition>
</extension>
     </context>
   </section>
Why inner condition not works?
_________________________________________________________________________
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_________________________________________________________________________
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Official FreeSWITCH Sites
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--
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Re: [Freeswitch-users] Broken silence with webrtc

Gary Foreman

Anthony Minessale

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participants (2)