[Freeswitch-users] Broken silence with webrtc
Gary Foreman
gaz.foreman at gmail.com
Tue Sep 1 19:36:21 MSD 2015
I have that running. Unfortunately info reported in the console output
appears to be the same whether the call has audio or not.
The only message I've seen that has given a clue was "no audio stun for a
long time". The setup is purely on a LAN so I'm not sure why stun is
causing an issue.
On Tue, Sep 1, 2015 at 4:25 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Ken Rice)
>
>
> ---------- Forwarded message ----------
> From: Ken Rice <krice at freeswitch.org>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 1 Sep 2015 10:25:10 -0500
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> 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 Ok so the issue has been superseded by intermittent one-way / no audio.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
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>
> You can reach the person managing the list at
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>
> 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
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>
> FreeSWITCH-users mailing list
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> 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
>
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>
>
>
>
> _______________________________________________
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>
>
>
>
> _________________________________________________________________________
> 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
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>
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