[Freeswitch-users] opus and freeswitch

Kamil Nigmatullin kamil.nigmatullin at gmail.com
Sat Jul 12 21:34:11 MSD 2014


Opus is meant to be dynamic. It has everything inside the library. But you
should feed the parameteres of how the server sees client and vise versa.
So I think it is a Skype's behaviour and that's why it is works good in
internet enviroment. However, Kristian, the man who respond me first time,
thinks of it a bit different way.


2014-07-11 23:15 GMT+06:00 Yosi Taguri <yosi at taguri.com>:

> We're moving away to freeswitch these days.
> What algorithm did you use to make opus dynamic?
>> Sent from Mailbox <https://www.dropbox.com/mailbox>
>
>
> On Fri, Jul 11, 2014 at 7:47 PM, Kamil Nigmatullin <
> kamil.nigmatullin at gmail.com> wrote:
>
>> Hello,
>> We have our own sip stack written on Android's Java and rtp side written
>> in C. With Server side we actually do not have ready solution now. Mod opus
>> works, but as I said it has statically defined parameters that actually cut
>> off all great features of opus codec. We may be implementing it inside
>> mod_opus somehow or maybe by modifying rtpproxy. Defenetly it won't be
>> asterisk.
>>
>>
>> 2014-07-11 20:11 GMT+06:00 Yosi Taguri <yosi at taguri.com>:
>>
>>> Hi Kamil,
>>> Noticed the thread on freeswitch and you saying that you figured it out
>>> on the client side.
>>> What sip stack are you using?
>>> What did you end up doing on the server side?
>>>
>>> We struck the same problem with asterisk.
>>>
>>> --
>>>
>>> Yosi Taguri | Founder & CTO | Yalo | http://yaloapp.com
>>> <http://ahhhpah.com> | Mobile: +972-54-4816020 | Google Voice US
>>> Mobile: +1 (415) 935-1126 | Email: yosi at taguri.com
>>>
>>>
>>>
>>
>>
>> --
>> Kamil Nigmatullin
>> Manager
>> Tel: 77272323748
>> mob: 7 777 2517003
>> ICQ: 31008382
>> Skype: Kamil Nigmatullin
>>
>
>


-- 
Kamil Nigmatullin
Manager
Tel: 77272323748
mob: 7 777 2517003
ICQ: 31008382
Skype: Kamil Nigmatullin
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