[asterisk-dev] Opus and VP8

[Lorenzo Miniero]

James,

I mostly tested it in conferencing scenarios, using ConfBridge to attach
heterogeneous peers. In those scenarios, I used both Chrome and Firefox
(using the DTLS-SRTP hacks I mentioned in the other thread), and PhonerLite
as well. For other codecs I used a bunch of softphones, our Java applet
(wideband speex) and calls from the PSTN using a VoIP gateway. This way it
seems to work as expected.

At the beginning, through, I mostly used PhonerLite and Linphone calling
each other through Asterisk to test the codec, and I can't recall issues
like those you mentioned, but I didn't test this intensively. As I
anticipated, I'm currently traveling and won't be able to do any testing
until Saturday: I'll keep you posted about this.

Lorenzo
Il giorno 30/mag/2013 00:43, "James Mortensen" <
james.mortensen at voicecurve.com> ha scritto:

>
> 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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