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

[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:miconda at gmail.com]
Sent: Wednesday, January 6, 2016 8:09 AM
To: Daniel W. Graham <dan at cmsinter.net>; Kamailio (SER) - Users Mailing List <sr-users at 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 at gmail.com<mailto:miconda at 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
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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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