[Freeswitch-users] Broken silence with webrtc

Tue Sep 1 19:25:10 MSD 2015

Just enable verto debugging in verto.conf.xml in your configs… it’ll print it right to the screen

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gary Foreman
Sent: Tuesday, September 1, 2015 10:13 AM
To: freeswitch-users at 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Ok so the issue has been superseded by intermittent one-way / no audio.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 at lists.freeswitch.org <mailto:freeswitch-users-request at 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 at gmail.com <mailto:ssinyagin at gmail.com> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at gmail.com <mailto:gaz.foreman at 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 at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> > wrote:

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        freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> 

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When replying, please edit your Subject line so it is more specific
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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 at gmail.com <mailto:alxpol at gmail.com> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at dwide.com <mailto:d.mordovin at dwide.com> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at gmail.com <mailto:gaz.foreman at gmail.com> >
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at freeswitch.org <mailto:brian at freeswitch.org> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at gmail.com <mailto:gaz.foreman at 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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-- 

Brian West
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  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
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Got Bugs? Report them here <https://freeswitch.org/jira> ! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> 

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iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

---------- Forwarded message ----------
From: Chris Young <Chris.Young at enghouse.com <mailto:Chris.Young at enghouse.com> >
To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> " <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at dwide.com <mailto:d.mordovin at dwide.com> >
To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> " <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at avimarcus.net <mailto:avi at avimarcus.net> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at 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 at dwide.com <mailto:d.mordovin at 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?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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