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

Avi Marcus avi at avimarcus.net
Thu Oct 7 06:35:07 UTC 2021


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
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