8 Jan
2016
8 Jan
'16
7:32 p.m.
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(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
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 <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
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(a)cmsinter.net
<mailto:dan@cmsinter.net>>; Kamailio (SER) - Users Mailing List
<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
<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
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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(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://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
