I did more experimenting and seams the issue only exists in two of three configurations. If I can fix the first I think it will fix the second as well. If both ATA ports share the same username and serial forking is used, the issue as described below happens. Looks like the issue is that I never called route(NATMANAGE) in the serial forking failure route.

-Dan *From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Daniel W. Graham *Sent:* Wednesday, January 6, 2016 3:06 PM *To:* miconda(a)gmail.com; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA 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 &lt;dan(a)cmsinter.net <mailto:dan@cmsinter.net>>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org <mailto: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 &lt;miconda(a)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(a)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(a)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

The SDP was updated with RTPProxy IP. Yes, config was written around the default config, here are some snippets of the config that is related. Do I just need to call branch route in the failure route? if ($branch(count) > 0) { t_load_contacts(); t_next_contacts(); t_on_failure("HUNT_FAIL"); } route(RELAY); ------------------ route[RELAY] { if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } branch_route[MANAGE_BRANCH] { xlogl("L_INFO", "$ci : New branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } failure_route["HUNT_FAIL"] { if (!t_next_contacts()) { exit; } t_on_failure("HUNT_FAIL"); t_relay(); } dan-signature *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] *Sent:* Thursday, January 7, 2016 4:24 AM *To:* Daniel W. Graham &lt;dan(a)cmsinter.net&gt;et>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA On 06/01/16 21:28, Daniel W. Graham wrote: I did more experimenting and seams the issue only exists in two of three configurations. If I can fix the first I think it will fix the second as well. If both ATA ports share the same username and serial forking is used, the issue as described below happens. Looks like the issue is that I never called route(NATMANAGE) in the serial forking failure route. If you are having your config based on default kamailio.cfg, then you should engage the branch route before sending out any invite. Cheers, Daniel -Dan *From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Daniel W. Graham *Sent:* Wednesday, January 6, 2016 3:06 PM *To:* miconda(a)gmail.com <mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; <mailto:sr-users@lists.sip-router.org> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA 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 &lt;dan(a)cmsinter.net <mailto:dan@cmsinter.net>>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org <mailto: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 &lt;miconda(a)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(a)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(a)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://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

I follow now :) tested and working. Thanks Daniel for the help! -Dan *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] *Sent:* Friday, January 8, 2016 3:33 AM *To:* Daniel W. Graham &lt;dan(a)cmsinter.net&gt;et>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA You need to engage branch route again in failure route. All those tm route blocks need to be re-engaged for each t_relay(). Cheers, Daniel On 07/01/16 22:09, Daniel W. Graham wrote: The SDP was updated with RTPProxy IP. Yes, config was written around the default config, here are some snippets of the config that is related. Do I just need to call branch route in the failure route? if ($branch(count) > 0) { t_load_contacts(); t_next_contacts(); t_on_failure("HUNT_FAIL"); } route(RELAY); ------------------ route[RELAY] { if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } branch_route[MANAGE_BRANCH] { xlogl("L_INFO", "$ci : New branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } failure_route["HUNT_FAIL"] { if (!t_next_contacts()) { exit; } t_on_failure("HUNT_FAIL"); t_relay(); } dan-signature *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] *Sent:* Thursday, January 7, 2016 4:24 AM *To:* Daniel W. Graham &lt;dan(a)cmsinter.net&gt; <mailto:dan@cmsinter.net>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; <mailto:sr-users@lists.sip-router.org> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA On 06/01/16 21:28, Daniel W. Graham wrote: I did more experimenting and seams the issue only exists in two of three configurations. If I can fix the first I think it will fix the second as well. If both ATA ports share the same username and serial forking is used, the issue as described below happens. Looks like the issue is that I never called route(NATMANAGE) in the serial forking failure route. If you are having your config based on default kamailio.cfg, then you should engage the branch route before sending out any invite. Cheers, Daniel -Dan *From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Daniel W. Graham *Sent:* Wednesday, January 6, 2016 3:06 PM *To:* miconda(a)gmail.com <mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org&gt; <mailto:sr-users@lists.sip-router.org> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA 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 &lt;dan(a)cmsinter.net <mailto:dan@cmsinter.net>>; Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.sip-router.org <mailto: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 &lt;miconda(a)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(a)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(a)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://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://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu