[asterisk-dev] Opus and VP8

[Andrea Suisani]

On 05/28/2013 03:58 PM, Lorenzo Miniero wrote:
> Hi Andrea,
>
>
> 2013/5/28 Andrea Suisani <sickpig at opinioni.net <mailto:sickpig at opinioni.net>>
>
>     Hi Lorenzo,
>
>     Firstly let me thank you for the work you're doing!
>
>
> Glad you're finding it useful!

:)

[cut]

>
>
> Are you sure the choppy sound was not caused by the mobile connectivity of the mobile phone?

yes I'm. in the same configuration calling the same mob number I'm able to place
a good call on both side using ulaw code.

> even though I guess that was not the case, considering the caller side (the browser) was receiving good audio instead, origating from the mobile phone itself. Besides, have you verified Opus was actually
> used? I think that, as the patch is right now, Opus is not very high in the codec priority list when negotiating.

I've checked twice just to be sure :) And I can confirm that tthe caller leg was using opus as codec.

[May 28 11:17:40] VERBOSE[16613][C-00000000] app_dial.c:     -- Called SIP/beffect-sip/32XXXXXXXXXXX
[May 28 11:17:40] DEBUG[16613][C-00000000] channel.c: Driver for channel 'SIP/webrtc_form_cloud2_1369732652-00000000' does not support indication 3, emulating it
[May 28 11:17:40] VERBOSE[16613][C-00000000] codec_opus.c: [Opus] Created encoder #3 (8000->opus)
[May 28 11:17:40] DEBUG[16613][C-00000000] channel.c: Set channel SIP/webrtc_form_cloud2_1369732652-00000000 to write format slin
[May 28 11:17:40] DEBUG[16613][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer tic
>
> That said, I'm not sure what the cause of the issue could be. What ptime are the two peers using?
> What is the transcoding path
> as displayed on the Asterisk console? Does the same happen in other scenarios as well, e.g., the browser interacting with another
> browser, or a softphone using a different codec (e.g., speex) at different rates (e.g., 16kHz vs 8kHz)?

now I'm out of office I will be able to provide this info
only on Thursaday.


> The 'opus set debug' command can be enabled when an encoder/decoder is created,
> to check whether decoding and encoding give any error, and the size of packets/samples that are extracted out of those processes.
> Try enabling a "normal" debug to see if any of error appears there (e.g., decoding errors
> that may result in corrupt outgoing audio) .

Ok I will


thanks again for your work

Andrea