8 Jan
2016
8 Jan
'16
6:32 p.m.
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 <dan(a)cmsinter.net>et>; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org>
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 <dan@cmsinter.net><mailto:dan@cmsinter.net>; Kamailio
(SER) - Users Mailing List
<sr-users@lists.sip-router.org><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@gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List
<sr-users@lists.sip-router.org><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 <dan@cmsinter.net<mailto:dan@cmsinter.net>>; Kamailio
(SER) - Users Mailing List
<sr-users@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
<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
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--
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://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
