[Freeswitch-users] Broken silence with webrtc

Gary Foreman gaz.foreman at gmail.com
Wed Sep 2 00:05:54 MSD 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 at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>         freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
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>
> You can reach the person managing the list at
>         freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>    1. Re: Broken silence with webrtc (Anthony Minessale)
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 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> wrote:
>>
>> Send FreeSWITCH-users mailing list submissions to
>>         freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>>         freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>>         freeswitch-users-owner at lists.freeswitch.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>> Today's Topics:
>>
>>    1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>>
>>
>> ---------- Forwarded message ----------
>> From: Stanislav Sinyagin <ssinyagin at gmail.com>
>> To: FreeSWITCH Users Help <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>
>> 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> wrote:
>>
>> Send FreeSWITCH-users mailing list submissions to
>>         freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>>         freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>>         freeswitch-users-owner at lists.freeswitch.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>> 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>
>> To: FreeSWITCH Users Help <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>
>> To: FreeSWITCH Users Help <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>
>> To: 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>
>> To: FreeSWITCH Users Help <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>
>> 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!
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Chris Young <Chris.Young at enghouse.com>
>> To: "freeswitch-users at lists.freeswitch.org" <
>> 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>
>> To: "freeswitch-users at lists.freeswitch.org" <
>> 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 #
>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
>> foobar \d+"/>
>>       <action application="log" data="CRIT ${foobar}"/>
>>
>>       <condition field="${foobar}" expression="1$">
>>             <action application="log" data="CRIT
>> --------------1------------------"/>
>>             <anti-action application="log" data="CRIT --------------not
>> 1------------------"/>
>>       </condition>
>>
>>       <action application="log" data="CRIT ---- last ${foobar}"/>
>>     </condition>
>>
>>       </extension>
>>     </context>
>>   </section>
>>
>> Why inner condition not works?
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Avi Marcus <avi at avimarcus.net>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>> Subject: Re: [Freeswitch-users] Playing with conditions
>>
>> Short answer: Each extension only has 1 set of conditions.
>>
>> The condition evaluating foobar is run *before* it gets set.
>>
>> After the play_and_get_digits you should transfer/execute_extension to a
>> new extension that will evaluate ${foobar}
>>
>>
>> -Avi Marcus
>> BestFone
>>
>>
>>
>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin at dwide.com>
>> wrote:
>>
>> Hello
>>
>> <section name="dialplan" description="Local call">
>>      <context name="public">
>>        <extension name="local_1">
>>
>>      <condition field="destination_number" expression="^(\d{4})$">
>>        <action application="answer"/>
>>
>>        <action application="play_and_get_digits" data="1 1 3 7000 #
>> $${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
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
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>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _______________________________________________
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