Hello, is the firewall doing SIP ALG?
Can you get a SIP network trace on UA? If yes, compare it with the one captured on server. Cheers, Daniel On 06/01/16 01:50, Daniel W. Graham wrote: > > Setup is - > > > > 2 port ATA <> FIREWALL <> KAMAILIO / RTPPROXY <> ASTERISK > > > > If I have a single port in use behind the firewall, all NAT functions > work properly and media is relayed through rtpproxy. > > > > If I have both ports in use behind the firewall, when outbound calls > from UA are placed there is two way audio on both calls. However if > inbound calls are placed to UA, the first call works, second call only > has outbound audio. > > > > Different SIP URI is used for each port. > > > > If the firewall is eliminated everything works fine. > > > > Anyone have an idea how to troubleshoot or what could be missing? I > have done packet captures on both the UA side and Kamailio side, and I > see two RTP flows (rtp ports match on both sides as well) despite lack > of inbound audio on the second call. > > > > If I can post anything config wise that would help let me know. > > > > Thanks! > > > > -Dan > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
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