[asterisk-dev] Opus and VP8

Lorenzo Miniero lminiero at gmail.com
Sat Jun 1 06:18:44 CDT 2013


2013/5/30 James Mortensen <james.mortensen at voicecurve.com>

>
> On Thu, May 30, 2013 at 5:15 AM, Lorenzo Miniero <lminiero at gmail.com>wrote:
>
>> 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
>>>
>>
>
>
> Hi Lorenzo,
>
> Thanks again for your help.  When you do get back around to this, can you
> specifically test the scenario where you're calling from Chrome to Asterisk
> to the PSTN?  Using the codecs opus to g729 and opus to ulaw?
>
> We really appreciate the work you're doing on this!  Thank you again!
>
>
Hi James,

sure, I'll try and replicate the scenarios as soon as I get back to the lab
on Monday. I've just come back from two days out of town for work and,
although I wanted to start looking into this today, I must admit I
understimated the slumbering power of weekends :-)

Cheers,
Lorenzo



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