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:mico...@gmail.com] > *Sent:* Wednesday, January 6, 2016 8:09 AM > *To:* Daniel W. Graham <d...@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 > <mico...@gmail.com <mailto:mico...@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 -- 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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