[asterisk-bugs] [JIRA] (ASTERISK-27748) Random webrtc calls without audio and ALAW<->OPUS transcoding errors

[Fran Vicente (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27748?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242935#comment-242935 ] 

Fran Vicente edited comment on ASTERISK-27748 at 4/4/18 2:22 AM:
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We are able to reproduce the problem on multiple systems, but on all of them, the Asterisk servers are running virtualized under VMware ESXi. The hosts where the virtual machines run are mostly idle, so it shouldn't be a problem related with high resources utilization.

To discard problems related with packet loss, we have tried with clients running on the same local network, but the problem also occurs.

I found strange that once the error occurs, it continues until the call gets hung up.


was (Author: fvicente):
We are able to reproduce the problem on multiple systems, but on all of them, the Asterisk servers are running virtualized under VMware ESXi. The host where the virtual machines run is mostly idle, so it shouldn't be a problem related with high resources utilization.

To discard problems related with packet loss, we have tried with clients running on the same local network, but the problem also occurs.

I found strange that once the error occurs, it continues until the call gets hung up.

> Random webrtc calls without audio and ALAW<->OPUS transcoding errors
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-27748
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27748
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_opus, Core/CodecInterface
>    Affects Versions: 15.3.0
>         Environment: CentOS release 6.9 (Final)
> Linux fvicastwrtc.local 2.6.32-696.el6.x86_64 #1 SMP Tue Mar 21 19:29:05 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux
> Running on VMWare ESXi 6.0
>            Reporter: Fran Vicente
>            Assignee: Fran Vicente
>              Labels: opus, pjsip, webrtc
>         Attachments: capture2.zip, full, pjsip_extensions_pogp.conf
>
>
> I have an environment which uses an Asterisk 15.3.0 to bridge calls to WebRTC endpoints configured using OPUS codec. The calls are originated on another Asterisk, connected to this using ALAW codec.
> All the calls run the same dialplan (that simply calls the webrtc client) and the same webrtc endpoints, but randomly the following errors shows on the Asterisk console, and I get no audio on the calls. 
> {code}
> [Mar 19 11:53:25] WARNING[8195][C-00000002] translate.c: Out of buffer space
> [Mar 19 11:53:25] DEBUG[8194][C-00000002] translate.c: Sample size different 160 vs 960
> [Mar 19 11:53:25] DEBUG[8194][C-00000002] translate.c: Sample size different 160 vs 960
> [Mar 19 11:53:25] ERROR[8195][C-00000002] codec_opus.c: Opus: Unable to parse packet for number of samples: corrupted stream
> [Mar 19 11:53:25] WARNING[8195][C-00000002] translate.c: no samples for opustolin
> [Mar 19 11:53:25] ERROR[8195][C-00000002] codec_opus.c: Opus: decoding: corrupted stream
> [Mar 19 11:53:25] WARNING[8195][C-00000002] translate.c: Out of buffer space
> [Mar 19 11:53:25] DEBUG[8194][C-00000002] translate.c: Sample size different 160 vs 960
> [Mar 19 11:53:25] ERROR[8195][C-00000002] codec_opus.c: Opus: decoding: corrupted stream
> {code}
> The dialplan simply does:
> {code}
> exten => _X.,1,Dial(PJSIP/${EXTEN})
> {code}
> See the complete log and configuration attached.



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