10 Aug
2017
10 Aug
'17
12:59 a.m.
Hi,
It is better if you want route RTP directly between UA, Dont route the
calls to Asterisk, And do this like below:
UAC1---sip---->kamailio--------->UAC2
In asterisk, there is directmedia options for handle RTP.
Be notice you should use STUN in this regards. becuase of type of nats
in clients, you have some challenge.
On Wed, Aug 9, 2017 at 6:18 PM, wsotest.512 <wsotest.512(a)gmail.com> wrote:
Hi all,
We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp
proxy. Kamailio have public IP, asterisk – no. All calls between clients now
going like that:
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB
All clients of course from Internet and behind Nat. Main problem is amount
of traffic going through Kamailio and Asterisk. We need to pay for every
additional GB behind limit in tariff plan to hosting provider.
So we decided to try route all rtp traffic between users directly.
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> --> --> UserB
Is it possible at all? Maybe someone already did it …
--
BR, Alex
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--Mojtaba Esfandiari.S