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

[Daniel-Constantin Mierla]

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 at lists.sip-router.org] *On
> Behalf Of *Daniel W. Graham
> *Sent:* Wednesday, January 6, 2016 3:06 PM
> *To:* miconda at gmail.com; Kamailio (SER) - Users Mailing List
> <sr-users at 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 at gmail.com]
> *Sent:* Wednesday, January 6, 2016 8:09 AM
> *To:* Daniel W. Graham <dan at cmsinter.net <mailto:dan at cmsinter.net>>;
> Kamailio (SER) - Users Mailing List <sr-users at lists.sip-router.org
> <mailto: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
>
>              
>
>
>
>             _______________________________________________
>
>             SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
>             sr-users at lists.sip-router.org
>             <mailto:sr-users at 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 at lists.sip-router.org
>         <mailto:sr-users at 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

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