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
On Wed, Aug 09, 2017 at 04:48:02PM +0300, wsotest.512 wrote:
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> --> --> UserBIs it possible at all? Maybe someone already did it .
It should work, but Asterisk is broken in this respect and may break codecs/dtmf: https://issues.asterisk.org/jira/browse/ASTERISK-25166
The root cause is that Asterisk is initially handling RTP and later tries to reINVITE both legs with the ip of the rtpengine/userb for media. If the ids of codecs/dtmf don't match in the m=audio SDP line RTP will break. There is no way to get Asterisk not to handle initial RTP and no way to not have Asterisk reINVITE if the ids differ.
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 …
--
BR, Alex
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Daniel Tryba -
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wsotest.512