Default Kamailio 1.5.x install not working?

Ok, I've been beating myself up over this one for over a week now.

I have a default Debian 5.0 Lenny install, running Kamailio 1.5.0~svn20090504-1 package which was downloaded by adding these lines to my /etc/apt/sources.list:

deb http://devel.kamailio.org/debian lenny main
deb-src http://devel.kamailio.org/debian lenny main

Here's what I did:

1. Using the default Kamailio.cfg file, I enabled:
a. MySQL with: sed -i 's///g' kamailio.cfg
b. Authentication with: sed -i 's///g' kamailio.cfg
c. Persistent users with: sed -i 's///g' kamailio.cfg
d. RTPProxy with: sed -i 's///g' kamailio.cfg
e. Enhanced Accounting with: sed -i 's///g' kamailio.cfg

2. I changed the appropriate places in the config for MySQL setup.

3. I have RTPProxy running as
/usr/sbin/rtpproxy -s udp:localhost 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l <my public IP>

4. I added the SQL statements for the enhanced accounting

5. I added 2 users to the database with:
# kamctl add 1000 1234
# kamctl add 1001 1234

Both my SIP clients (1000 & 1001) are behind a NAT (behind the same NAT at the moment).

Now my two SIP clients (1000 & 1001) can connect just fine, they get authenticated, and they can call eachother - however, there is no audio being passed.

I've been thru the forums a dozen times now looking for answers and haven't been able to figure out what the problem is.

Does the default configuration not work out of the box? I see in the log files that nathelper IS detecting RTPProxy, and it IS enabling support for it. I don't know where to go from here..

I just did an apt-get update;apt-get upgrade and it grabbed the latest version "1.5.0~svn20090508-1", but I am still having the same problem of no audio...

Be sure you have 1.5.1 for nat traversal. The route that detected callers behind nat was not enabled in previous version.

miconda :

Be sure you have 1.5.1 for nat traversal. The route that detected callers behind nat was not enabled in previous version.

Ok, and not to sound stupid, but how do I check which version I have?

The debian package was installed directly from:
devel.kamailio.org/debian

Doing an apt-cache show mediaproxy-relay shows:
Package: kamailio
Priority: optional
Section: net
Installed-Size: 4088
Maintainer: Debian VoIP Team <pkg-voip-maintainers@lists.alioth.debian.org>
Architecture: i386
Version: 1.5.0~svn20090508-1

But in the system logs I see...

May 8 15:37:59 sipreg01 /usr/sbin/kamailio[405]: INFO:core:main: version: kamailio 1.6.0-dev0-notls (i386/linux)

So I am guessing that I'm using the latest version?

Bah... I typed mediaproxy-relay above for the apt-cache show and what I meant to type was kamailio, but everything still applies - that's what I get for not copying & pasting...

I've been working on trying to implement mediaproxy instead of rtpproxy because someone else had mentioned that MediaProxy "works better for NAT traffic", what that means I have no idea, but MediaProxy is significantly more complex to implement so I'm not quite there yet...

Never used mediaproxy, but latest version of rtpproxy has quite a lot of features, including ability to record/stream the audio. Check it at rtpproxy.org. It is also developed more close to ser/kamailio - http://sip-router.org project.

reply to a previous message:
- yes, looks to be development version from SVN. That should have the 1.5.1 config

However, have in mind that the development of kamailio is done now in GIT repository hosted at http://sip-router.org

Being the latest config, then run kamailio in debug mode and see if you get some hints there. Use xlog in your config to be sure that the nat related functions are called during the config execution.

miconda :

reply to a previous message: - yes, looks to be development version from SVN. That should have the 1.5.1 config However, have in mind that the development of kamailio is done now in GIT repository hosted at http://sip-router.org Being the latest config, then run kamailio in debug mode and see if you get some hints there. Use xlog in your config to be sure that the nat related functions are called during the config execution.

Ok, what I have found out so far is that if I turn on Symmetric Routing in my Grandstream GXP2000's everything works fine, but I don't have to turn it on for it to work with Asterisk (and the default is off) which leads me to believe that at some point I'm going to have problems with some other devices if I don't figure this out...

And already, my situation is this:

1. I can call Grandstream to Grandstream and everything is fine

2. Calling X-lite to Grandstream everything seems fine.

3. Calling Grandstream to X-lite I get one way audio from the Grandstream.

To eliminate as many variables as possible, I went back to the default Kamailio.cfg and ONLY enabled RTP Proxy. I expected to have it working as well as Asterisk, but obviously I am missing something. I would think my situation would be extremely common...

I just want an OpenSER/SER/Kamailio setup to accept registrations and provide routing regardless of where the client is.