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@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 --> UserBAll 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--> --> --> UserBIs it possible at all? Maybe someone already did it …
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BR, Alex
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