[asterisk-dev] Opus and VP8

[James Mortensen]

On Wed, May 29, 2013 at 11:06 AM, Lorenzo Miniero <lminiero at gmail.com>wrote:

> 2013/5/29 James Mortensen <james.mortensen at voicecurve.com>
>
>>
>> On Wed, May 29, 2013 at 8:37 AM, James Mortensen <
>> james.mortensen at voicecurve.com> wrote:
>>
>>>
>>>
>>> There appears to be paths for both ulaw as well as g729.
>>>
>>> I did try the patch on Asterisk 11.1.2, but shifted gears when I
>>> realized the crypto headers are buggy as per
>>> https://issues.asterisk.org/jira/browse/ASTERISK-20849.  However, I'll
>>> apply the patch to fix that issue and then try a call on Asterisk 11.1.2.
>>>
>>> I'll follow up with additional details when I know more.  Again, thank
>>> you for your time and your work on this. We're really stoked about getting
>>> opus working in Asterisk! :)
>>>
>>>
>>> --
>>> James Mortensen
>>> Project Manager, VoiceCurve, Inc.
>>> 866-707-4590
>>> james.mortensen at voicecurve.com
>>>
>>
>>
>> Hi Lorenzo,
>>
>> The problem exists on Asterisk 11.1.2 as well.  It sounds like the two
>> parties are talking underwater.  Please let me know if there's any other
>> details that I can get you.
>>
>>
> Ok, thanks for the details and the clarification. Unfortunately I'll be
> out of office for the next two days so I won't be able to look into this
> right away, but I'll check what may be going wrong as soon as I get back to
> work.
>
> Please keep me posted if you find any additional info that may be of use.
>
> Cheers,
> Lorenzo
>
>
>> --
>> James Mortensen
>> Project Manager, VoiceCurve, Inc.
>> 866-707-4590
>> james.mortensen at voicecurve.com
>>
>
>

Hi Lorenzo,

I wanted to let you know that our trunk provider supports both ulaw and
g729. We're using Chrome with JsSIP in the browser.

We've tried calls transcoding to/from ulaw to opus and g729 to opus, and we
still hear the robotic audio in both scenarios, and this is on both
Asterisk 11.1.2 and Asterisk 11.4.0.

What is your setup like, as it sounds like you actually have this working?
 What codecs are you transcoding to/from?  Are you making calls from the
browser at all or are you only testing with that PhonerLite system?

Also, our SIP clients are on Mac OS 10.8.  The Asterisk server is on Ubuntu
12.04.  I installed the opus libraries from this site:
http://www.opus-codec.org/downloads/ using the latest version, 1.0.2

Hope this helps!  Let us know if there's anything else we should try.


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