[SR-Users] 2 phones behind same NAT, Kamailio on Pub IP One Way Audio

[Daniel-Constantin Mierla]

Hello,

if both phones are behind same nat and no nat processing enabled in
kamailio.cfg, the audio should work fine. Be sure there is no ALG in
your nat router or firewall. The best is to look at singaling, on the
server you can use:

ngrep -d any -qt -W byline "sip" port 5060

If the phones are behind nat, you need to enable nat traversal in
kamailio config by defining WITH_NAT, install it and configure to listen
for control commands on the socket specified to rtpproxy kamailio module
parameters (see the comments at the top of default kamailio.cfg).


Cheers,
Daniel

On 29/04/15 05:03, Todd R. wrote:
> This is my first go round' with Kamailio but I have been messing with
> it off and on for a few weeks.
>
> Running latest version on latest CentOS 7.x.
>
> Kamailio installed on VM with public IP in one location, 2 SIP phones
> behind same NAT, both registered to Kamailio fine.
>
> I can call one phone to the other and it rings and I can answer it but
> get one way audio most of the time, sometimes no audio at all.
>
> I just did a standard install with MySQL and no other modules.
>
> What am I missing, do I need RTPPROXY? Will Kamailio allow extension
> to extension calls from/to phones behind NATS without any additional
> modules?
>
> I tried installing it with yum, it installed but it won't start. If I
> need it, I will remove the YUM version and install from source or GIT
> or some other method.
>
> I do NOT want any media passing through this server, that's one big
> reason I am learning something OTHER than Asterisk which I am very
> familiar with.
>
> I see all these RTP modules, music on hold etc but I specifically
> don't want to install anything that causes media to pass through the box.
>
> Finding the learning curve much steeper than Asterisk back in the day
> and a REALLY tough time find step by step examples to get started,
> especially on CentOS.
>
> I bought the draft of the book and it will be a great resource but at
> the moment, it's not helping me get started or get my first instance
> up and running.
>
> I need to add SIP trunks to originate and terminate calls but I can't
> even think about that since I can't even get audio on extension to
> extension calls yet.
>
> I also see that Kamailio fails to startup at boot because it can't
> connect to MySQL, when I restart Kamailio it starts fine. I think the
> issue is that Kamailio is trying to start before MySQL is up and
> running, I guess I can do a delayed start of Kamailio to fix that later.
>
> Any help would be appreciate by this NEWB.
>
> Thanks in advance for any assistance.
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com

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