[Freeswitch-users] Bridge to other FS server has no audio until DTMF
Avi Marcus
avi at avimarcus.net
Thu Oct 7 12:30:09 UTC 2021
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
>>>> 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
>
> --
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211007/499a1e09/attachment-0001.html>
More information about the FreeSWITCH-users
mailing list