Is the SDP updated with the IP of RTPProxy?
Cheers, Daniel
On 06/01/16 21:06, Daniel W. Graham wrote:
I do control, this particular setup is in my lab. I just took another look at the captures and see both RTP streams (viewing in front of firewall). First call rtp is sourced from Kamailio(rtpproxy) second call rtp is sourced from one of the backend asterisk servers (which is where the issue is, should also be from rtpproxy).
-Dan
*From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] *Sent:* Wednesday, January 6, 2016 8:09 AM *To:* Daniel W. Graham dan@cmsinter.net; Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA
Is the firewall a system that you control and can do traces on it? Can you see rtp coming to it? Is it forwarded?
Cheers, Daniel
On 06/01/16 13:40, Daniel W. Graham wrote:
Firewall is not doing sip alg, I have compared traces and they are the same. Daniel W. Graham CMSInter.net <http://cmsinter.net> LLC 989.400.4230 On Jan 6, 2016, at 3:05 AM, Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>> wrote: 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 <mailto: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://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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://twitter.com/#%21/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu