Re: [Freeswitch-users] opus and freeswitch
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@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@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@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@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
I don't. I think what you describe should be implemented using the means available within the codec, not RTCP. On Saturday, July 12, 2014, Kamil Nigmatullin <kamil.nigmatullin@gmail.com> wrote:
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@taguri.com <javascript:_e(%7B%7D,'cvml','yosi@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@gmail.com <javascript:_e(%7B%7D,'cvml','kamil.nigmatullin@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@taguri.com <javascript:_e(%7B%7D,'cvml','yosi@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@taguri.com <javascript:_e(%7B%7D,'cvml','yosi@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
-- Sent from mobile device
participants (2)
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Kamil Nigmatullin
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Kristian Kielhofner