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

[Kevin Harwell (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27748?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243976#comment-243976 ] 

Kevin Harwell commented on ASTERISK-27748:
------------------------------------------

[~fvicente], Is this still a problem?

I went back and tested my scenario again and realized I had a configuration problem for the endpoint I was using (I needed to specify values for the bitrate and max_playback_rate in codecs.conf). Once I did the call/audio in both directions sounded as it should.

That being said I am now unable to replicate your problem. I've gone back too and taken a look at the given pcap and configuration files. I only see the one call negotiated in the pcap and it negotiates L16/8000, which does not correspond to the given log. I'm not seeing any sip trace where a call using opus was attempted.

In order to move forward with this issue we'll require a full Asterisk debug log that includes debug set to at least 5, the sip trace of an offending call, and rtp debug turned on. Also include a packet capture that corresponds to that debug log.

If you are not sure how, please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> 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
>              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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