[asterisk-dev] Opus and VP8

James Mortensen james.mortensen at voicecurve.com
Tue May 28 13:10:10 CDT 2013


Hi Lorenzo,

Not trying to hijack the thread, but Andrea appears to be away for the next
few days, and I'm experiencing the same exact problem with the audio issues
and want to help move us both forward.  So I can answer the questions below
to get you more data. Please see below:


[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.
>
>
I assume the ptime is this maxptime value we see in the SDP from the Chrome
to Asterisk call leg:

a=rtpmap:111 opus/48000/2
a=maxptime:60
a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0

If that's not the ptime, please let me know where I might find that and
I'll get it for you.

And here you can see the call leg from the Chrome to Asterisk portion of
the call is opus, with the other leg using ulaw.  I've tried with g729 on
the Bandwidth to Asterisk leg and still hear the same underwater-like call
audio that Andrea describes.

ip-10-188-135-200*CLI> sip show channels
Peer             User/ANR         Call ID          Format           Hold
  Last Message    Expiry     Peer
54.X.X.X    1000            55srb9arjvh7rnv  (opus)           No       Rx:
ACK                    ws-Opensip
67.231.X.X     +15036XXXXXX     48c000dd54ba525  (ulaw)           No
Tx: ACK                    BANDWIDTH
2 active SIP dialogs
ip-10-188-135-200*CLI>


Also, here is the output from opus set debug huge:

[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 18 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 20 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 77 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 19 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 67 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 17 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 71 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 21 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 21 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 23 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 69 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 19 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 72 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 21 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 74 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 23 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 74 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 21 bytes
[Opus] [Decoder #12 (8000)] 960 samples, 75 bytes
[Opus] [Decoder #12 (8000)]   >> Got 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)] 160 samples, 320 bytes
[Opus] [Encoder #9 (8000)]   >> Got 960 samples, 25 bytes


Unlike Andrea, who applied the patch to 11.3.0, I successfully applied the
patch to Asterisk 11.4.0, but the issues we're both experiencing appear to
be the same. If that's not the case, I can start a new thread.  Hope this
helps! :)

-- 
James Mortensen
Project Manager, VoiceCurve, Inc.
866-707-4590
james.mortensen at voicecurve.com
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