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

Thu Oct 7 10:40:33 UTC 2021

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
>
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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