subject:"\[SR\-Users\] Audio issue when using 2 port ATA"

[SR-Users] Audio issue when using 2 port ATA

2016-01-05 Thread Daniel W. Graham

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
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-06 Thread Daniel-Constantin Mierla

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
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
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

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-06 Thread Daniel W. Graham

Firewall is not doing sip alg, I have compared traces and they are the same.

Daniel W. Graham
CMSInter.net LLC
989.400.4230

On Jan 6, 2016, at 3:05 AM, Daniel-Constantin Mierla 
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
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



--
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

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-06 Thread Daniel-Constantin Mierla

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  LLC
> 989.400.4230
>
> On Jan 6, 2016, at 3:05 AM, Daniel-Constantin Mierla
> 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
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>> -- 
>> 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
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> 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://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
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-06 Thread Daniel W. Graham

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 ; Kamailio (SER) - Users Mailing List 

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 
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://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
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-06 Thread Daniel W. Graham

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-boun...@lists.sip-router.org] On Behalf Of 
Daniel W. Graham
Sent: Wednesday, January 6, 2016 3:06 PM
To: mico...@gmail.com; Kamailio (SER) - Users Mailing List 

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:mico...@gmail.com]
Sent: Wednesday, January 6, 2016 8:09 AM
To: Daniel W. Graham mailto:d...@cmsinter.net>>; Kamailio 
(SER) - Users Mailing List 
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 
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://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
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-07 Thread Daniel-Constantin Mierla

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 ; Kamailio (SER) - Users
> Mailing List 
> *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
> 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

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-07 Thread Daniel W. Graham

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:mico...@gmail.com]
Sent: Thursday, January 7, 2016 4:24 AM
To: Daniel W. Graham ; Kamailio (SER) - Users Mailing List 

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-boun...@lists.sip-router.org] On Behalf Of 
Daniel W. Graham
Sent: Wednesday, January 6, 2016 3:06 PM
To: mico...@gmail.com<mailto:mico...@gmail.com>; Kamailio (SER) - Users Mailing 
List <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:mico...@gmail.com]
Sent: Wednesday, January 6, 2016 8:09 AM
To: Daniel W. Graham mailto:d...@cmsinter.net>>; Kamailio 
(SER) - Users Mailing List 
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 
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://

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-08 Thread Daniel-Constantin Mierla

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:mico...@gmail.com]
> *Sent:* Thursday, January 7, 2016 4:24 AM
> *To:* Daniel W. Graham ; Kamailio (SER) - Users
> Mailing List 
> *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-boun...@lists.sip-router.org] *On
> Behalf Of *Daniel W. Graham
> *Sent:* Wednesday, January 6, 2016 3:06 PM
> *To:* mico...@gmail.com <mailto:mico...@gmail.com>; Kamailio (SER)
> - Users Mailing List 
> <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:mico...@gmail.com]
> *Sent:* Wednesday, January 6, 2016 8:09 AM
> *To:* Daniel W. Graham  <mailto:d...@cmsinter.net>>; Kamailio (SER) - Users Mailing List
> 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
> mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> is the firewall doing SIP ALG?
>
> Can you get a SIP network trace on UA? I

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-08 Thread Daniel W. Graham

I follow now :) tested and working.

Thanks Daniel for the help!

-Dan

From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Friday, January 8, 2016 3:33 AM
To: Daniel W. Graham ; Kamailio (SER) - Users Mailing List 

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:mico...@gmail.com]
Sent: Thursday, January 7, 2016 4:24 AM
To: Daniel W. Graham <mailto:d...@cmsinter.net>; Kamailio 
(SER) - Users Mailing List 
<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-boun...@lists.sip-router.org] On Behalf Of 
Daniel W. Graham
Sent: Wednesday, January 6, 2016 3:06 PM
To: mico...@gmail.com<mailto:mico...@gmail.com>; Kamailio (SER) - Users Mailing 
List <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:mico...@gmail.com]
Sent: Wednesday, January 6, 2016 8:09 AM
To: Daniel W. Graham mailto:d...@cmsinter.net>>; Kamailio 
(SER) - Users Mailing List 
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 
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

Re: [SR-Users] Audio issue when using 2 port ATA

2016-01-08 Thread Daniel-Constantin Mierla

Welcome - glad to hear it was sorted out!

Cheers,
Daniel

On 08/01/16 18:32, Daniel W. Graham wrote:
>
> I follow now :) tested and working.
>
>  
>
> Thanks Daniel for the help!
>
>  
>
> -Dan
>
>  
>
> *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com]
> *Sent:* Friday, January 8, 2016 3:33 AM
> *To:* Daniel W. Graham ; Kamailio (SER) - Users
> Mailing List 
> *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:mico...@gmail.com]
> *Sent:* Thursday, January 7, 2016 4:24 AM
> *To:* Daniel W. Graham 
> <mailto:d...@cmsinter.net>; Kamailio (SER) - Users Mailing List
>  <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-boun...@lists.sip-router.org]
> *On Behalf Of *Daniel W. Graham
> *Sent:* Wednesday, January 6, 2016 3:06 PM
> *To:* mico...@gmail.com <mailto:mico...@gmail.com>; Kamailio
> (SER) - Users Mailing List 
> <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:mico...@gmail.com]
> *Sent:* Wednesday, January

[SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

Re: [SR-Users] Audio issue when using 2 port ATA

11 matches

Site Navigation

Mail list logo

Footer information