Codec opus Exists but not at the desired implementation BBB 2.2.36

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Armin Schindler

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Nov 7, 2021, 11:30:39 AM11/7/21
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Hello all,

we have a BBB 2.2.36 installation with phone dial in possibility running.
The connection between phones and BBB is done via an own Asterisk PBX.
All is working fine and Asterisk is doing the transcoding into Opus for BBB-Freeswitch endpoint.

The problem is that sometimes a phone caller gets disconnected by Freeswitch/BBB with the
following errors in log:

[WARNING] switch_core_media.c:3254 [VBR]: Packet size change detected. Remote PTIME changed from [20] to [3]
[DEBUG] switch_core_media.c:3737 Changing Codec from opus@20ms@48000hz to opus@3ms@48000hz
[DEBUG] mod_opus.c:719 Opus decoder stats: Frames[22948] PLC[0] FEC[0]
[DEBUG] mod_opus.c:734 Opus encoder stats: Frames[0] Bytes encoded[0] Encoded length ms[0] Average encoded bitrate bps[0]
[DEBUG] mod_opus.c:719 Opus decoder stats: Frames[0] PLC[0] FEC[0]
[DEBUG] mod_opus.c:734 Opus encoder stats: Frames[23026] Bytes encoded[2463269] Encoded length ms[460520] Average encoded bitrate bps[42839]
[WARNING] switch_core_codec.c:727 Codec opus Exists but not at the desired implementation. 48000hz 3ms 1ch
[ERR] switch_core_media.c:3774 Can't load codec?
[NOTICE] switch_core_media.c:3775 Hangup sofia/external/xxxxxxxx [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

I think Asterisk wants to change the PTIME for this Opus session into something Freeswitch doesn't like and I tried to
change some Opus settings in our Asterisk, but without success.

Am I right that Asterisk is the problem here? Or can Freeswitch configured to accept this wanted change?
If Asterisk side should be fixed, does someone know which Asterisk-Opus configuration is needed?

Thanks!

Armin

Andrew Wells

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Nov 7, 2021, 12:06:42 PM11/7/21
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What is your sip.conf for asterisk?

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From: Armin Schindler <arm...@gmail.com>
Sent: Sunday, November 7, 2021 11:30 a.m.
To: bigbluebu...@googlegroups.com
Subject: [bigbluebutton-setup] Codec opus Exists but not at the desired implementation BBB 2.2.36

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Armin Schindler

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Nov 7, 2021, 12:37:53 PM11/7/21
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andrew...@kooltel.com schrieb am Sonntag, 7. November 2021 um 18:06:42 UTC+1:

What is your sip.conf for asterisk?

sip? I don't understand what sip has to do with it?
Call establishment works perfectly and in 99% of calls no error occurs.
If the error occurs, it is minutes after call establishment.

Anyway, I am using pjsip and the config is basically default to call BBB.

Armin

 

Andrew Wells

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Nov 7, 2021, 12:42:05 PM11/7/21
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Armin, you said you were using asterisk, now it's pjsip.
Anyways, sounds like you have it figured out anyways.  I won't bother you in trying to help.
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From: Armin Schindler <arm...@gmail.com>
Sent: Sunday, November 7, 2021 12:38 p.m.
To: BigBlueButton-Setup
Subject: Re: [bigbluebutton-setup] Codec opus Exists but not at the desired implementation BBB 2.2.36
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Subject: Re: [bigbluebutton-setup] Codec opus Exists but not at the desired implementation BBB 2.2.36

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Armin Schindler

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Nov 7, 2021, 12:49:02 PM11/7/21
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andrew...@kooltel.com schrieb am Sonntag, 7. November 2021 um 18:42:05 UTC+1:
Armin, you said you were using asterisk, now it's pjsip.

Yes, because it is chan_pjsip module of Asterisk, not the old chan_sip module. It is still Asterisk.
 
Anyways, sounds like you have it figured out anyways.  I won't bother you in trying to help.
 
I don't understand what you mean. Freeswitch hangs up the call after some time with the written
error log. But this happens not every time, just a few times a week.
And I just don't think SIP is the cause/problem here. The log seems clear that the opus rtp
session changes and freeswitch doesn't like that.

Armin


Fred Dixon

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Nov 8, 2021, 8:23:23 PM11/8/21
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Hi Armin,

We don't have an answer for you, but we can see that this was mentioned on the FreeSWITCH mailing list


 We'll dig a bit deeper and see if we can find out more information.

Regards,... Fred

To view this discussion on the web visit https://groups.google.com/d/msgid/bigbluebutton-setup/a193dbff-2b8c-4630-93ea-e122faee72fe%40kooltel.com.


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Fred Dixon

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Nov 9, 2021, 7:47:50 AM11/9/21
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Hi Armin,

Confirmed that FreeSWITCH does not support this ptime. 
  

Can you post your Astrisk configuration that is trying to constrain the ptime.  There might be others here that could help out.

Regards,.. Fred

P.S.  You probably need to engage the Astrisk community on this one as well.

Armin Schindler

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Nov 10, 2021, 3:50:27 AM11/10/21
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Hi Fred,

thank you for your investigation.
So it seems Opus of Asterisk changes parameters while in session to something it shouldn't.

The codec config for Opus in Asterisk doesn't have a ptime setting. The other settings like
max_bandwidth, complexity, bitrate, cbr (constant bitrate), max_playback_rate, etc. do change
SDP values, but in session the ptime change still happen.
I tried different settings with the paramaters described here: https://wiki.asterisk.org/wiki/display/AST/Codec+Opus
without success. 

I will ask Asterisk community as well.

I though changing codec to G.722 between Asterisk and BBB will help, but in Freeswitch log
I can see the ptime is changed from 20ms to 30ms at the beginning of the call and the sound is choppy.
As far as I know G.722 is always 20ms. Since my Asterisk installation runs G.722 with a lot of phones
without problem, it looks like an issue with freeswitch after all.

regards
Armin
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