[asterisk-dev] Status of OPUS codec work

Olle E. Johansson oej at edvina.net
Fri Aug 31 08:52:25 CDT 2012


31 aug 2012 kl. 15:29 skrev Paul Belanger <paul.belanger at polybeacon.com>:

> On 12-08-31 02:24 AM, Olle E. Johansson wrote:
>> 
>> 30 aug 2012 kl. 21:39 skrev Andrew Latham <lathama at gmail.com>:
>> 
>>> On Thu, Aug 30, 2012 at 3:32 PM, Olle E. Johansson <oej at edvina.net> wrote:
>>>> Friends,
>>>> I remember someone doing work with integrating the Opus codec before. Did it go anywhere?
>>>> 
>>>> If not, was there any technichal reasons for not completing it?
>>>> 
>>>> Just curious.
>>>> /O
>>>> --
>>> 
>>> Check the archives for August 1st 2011.  Looks like the standard was
>>> not done and there were limited uses.
>> 
>> Well, it's different now. And the Internet community needs more implementations of Opus,
>> more voice minutes. It's a cool codec and if anyone that understands the codec interfaces
>> in Asterisk has some spare bandwidth I really would like to see an Asterisk implementation.
>> 
>> Opus is likely to be mandatory to implement for WebRTC.
>> 
> Lets raise some funds on kickstarter and get dvossel to do the work.
Good idea!

/O
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 2307 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120831/52afa427/attachment.bin>


More information about the asterisk-dev mailing list