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[avi at avimarcus.net]

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@172.123.123.123 (972581234567@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@172.123.123.123 (972581234567@172.123.123.123) javascript(conference/lookupAndJoinConference.js 1234567)
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 (972581234567@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@172.123.123.123 (972581234567@172.123.123.123) entering state [completed][200]
2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 (972581234567@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@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@avimarcus.net (avi@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@lists.freeswitch.org (FreeSWITCH-users@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

[david.villasmil.work a...]

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@avimarcus.net (avi@avimarcus.net)> wrote:

[avi at avimarcus.net]

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@yyy.bestfone.com (number@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@gmail.com (david.villasmil.work@gmail.com)> wrote:

[david.villasmil.work a...]

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@avimarcus.net (avi@avimarcus.net)> wrote:

[avi at avimarcus.net]

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@gmail.com (david.villasmil.work@gmail.com)> wrote:

[brian at freeswitch.com]

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@avimarcus.net (avi@avimarcus.net)> wrote:

[david.villasmil.work a...]

That’s the one!

On Thu, 7 Oct 2021 at 16:11, Brian West <brian@freeswitch.com (brian@freeswitch.com)> wrote:

[avi at avimarcus.net]

I had to do this to get it to execute on the B leg:<action application="export" data="nolocal:execute_on_answer=playback silence_stream://100"/>

... but it didn't help. Only DTMF worked... either manually dialed or via queue_dtmf
Freeswitch A waited for my DTMF to actually send the silence. 
Version 1.10.6 -release-18-1ff9d0a60e 64bit




 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort noise payload to 13
 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com)] has been answered
 EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com) playback(silence_stream://100)
 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms

 -- 20 seconds later when I pressed a button --
 
 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing file silence_stream://100
 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com)) Callstate Change DOWN -> ACTIVE
 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com)) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com)) Running State Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090)
 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com (JOIN_CLASS_7229999@voip.bestfone.com)) State EXCHANGE_MEDIA
 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA
 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for [5] ts=960 dur=160/160/2000 seq=26795 lw=960







This seemingly shouldn't be an issue. FS1 already has active media from the A leg, so it should initiate to the B leg. The B leg has been instructed to play a file, so it should initiate to the A leg...
But if this is somehow unavoidable, perhaps we need a workaround config, where we have a simple variable in the bridge string to avoid the standoff?
-Avi Marcus










On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian@freeswitch.com (brian@freeswitch.com)> wrote: