[Freeswitch-users] Bridge to other FS server has no audio until DTMF

Brian West brian at freeswitch.com
Thu Oct 7 15:01:32 UTC 2021 

execure_on_answer=playback::silence_stream://100 should solve it.

/b
PS, the non pc term that this has been said to be is
https://en.wikipedia.org/wiki/Mexican_standoff

On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi at avimarcus.net> wrote:

> I meant there's audio from pstn to fs1, but indeed I'm observing no audio
> between fs1 and fs2.
>
> What api should I call with api on answer..?
>
> On Thu, Oct 7, 2021, 3:19 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> If you see rtp glowing both ways, then this is not the stalemate I was
>> talking about. The scenario I’m referring to is about FS not starting
>> sending rtp waiting for the other side to start sending, and the other side
>> doing the same thing, thus going into a stalemate. This is solved by
>> injecting a silence (I would do api_on_answer).
>>
>> What you’re describing seems different to me.
>>
>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I'm using dialplan bridge, so then the dialplan is over. How do I send
>>> silence after the bridge...? An api_on_answer with a uuid_broadcast..
>>> seems overly complicated.
>>>
>>> <action application="bridge" data="sofia/external/
>>> number at yyy.bestfone.com"/>
>>>
>>>
>>> (And I don't know why there isn't audio - I had to set up an audio to
>>> get to this options in the IVR... so there's already audio. And Server B
>>> also started a file playback so should have initiated audio.)
>>>
>>>
>>> -Avi Marcus
>>>
>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> I seem to remember Brian saying this was because FS is waiting for the
>>>> remote end to send audio before starting itself. I believe he recommended
>>>> sending an empty (silence) to force the audio stream to be sent even if fs
>>>> hasn’t received anything.
>>>>
>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> I started a new thread in case anyone muted it... it wasn't simply a
>>>>> network issue.
>>>>>
>>>>> It seems the bridging occurs and dialplan processes, but no media
>>>>> flows - until DTMF from the A-leg.
>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>>> freeswitch B.
>>>>>
>>>>> Calls directly from carrier to Freeswitch B are fine.
>>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>>> Freeswitch B are also fine.
>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in
>>>>> the media path, it's an FS issue...
>>>>>
>>>>>
>>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>>> bridge, to force the audio stream to update.
>>>>>
>>>>> Here's the log on freeswitch B:
>>>>>
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  log(DEBUG class chosen: 1234567)
>>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  playback(class/hold-wait-teacher.wav)
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>  entering state [completed][200]
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>  entering state [ready][200]
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz
>>>>>  1 channels 20ms
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>>> 2021-10-07 09:16:34.923190 [INFO
>>>>> ] switch_channel.c:522 RECV DTMF 3:2080
>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>>
>>>>>
>>>>> You can see a 10 second gap between call ready 200 and correct
>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't
>>>>> happen automatically, only when I choose to press something.
>>>>>
>>>>>
>>>>> Any ideas as to the root cause of this?
>>>>>
>>>>>
>>>>> -Avi Marcus
>>>>>
>>>>> ---------- Forwarded message ---------
>>>>> From: Avi Marcus <avi at avimarcus.net>
>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>>> Subject: Bridge to other FS server has no audio ???
>>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>>
>>>>>
>>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>>> think my upstream changed his SDP again.
>>>>>
>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>>>> server B. Media works.
>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not
>>>>> on bypass). Media works.
>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to
>>>>> FS server B. Call looks OK, but no media is flowing (I don't hear anything,
>>>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>>>> codecs are set in the json cdrs (PCMU).
>>>>>
>>>>> FS server B is to join a conference if that matters.
>>>>>
>>>>> I was assuming it had to do with codecs, but setting
>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>>> it's already always PCMU.
>>>>>
>>>>> I have NO clue what further could cause this other than codecs, which
>>>>> seem to be fine. Any ideas please?
>>>>>
>>>>>
>>>>> -Avi Marcus
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
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>>>>
>>>> --
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
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>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://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
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://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
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://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
> https://freeswitch.com



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