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Hello,<br>
<br>
from ngrep, if 00900223 is same as 900223, then it is the incoming
call that has caller==callee. This is before the call gets to SIP
server, so it has nothing to do with SIP server itself, no matter it
does rtpproxy or not.<br>
<br>
Here is the head part of the incoming INVITE:<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:00900223@sip.malagasy.com:5060;user=phone">sip:00900223@sip.malagasy.com:5060;user=phone</a> SIP/2.0.<br>
Via: SIP/2.0/UDP
192.168.18.74:5060;branch=z9hG4bK8031929703792643643-157087377.<br>
From:
"900223"<a class="moz-txt-link-rfc2396E" href="sip:900223@sip.malagasy.com:5060;user=phone"><sip:900223@sip.malagasy.com:5060;user=phone></a>;tag=c0a80101-95cf690.<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:00900223@sip.malagasy.com:5060;user=phone"><sip:00900223@sip.malagasy.com:5060;user=phone></a>.<br>
<br>
RTPProxy itself can be used without any nat involved in a call,
simply for proxying the media stream.<br>
<br>
If you use latest 3.2.x, to enable nat-traversal/rtpproxy, just add<br>
<br>
#!define WITH_NAT<br>
<br>
after the first line in config. You don't need to remove the lines
with '#!ifdef WITH_NAT'.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
On 4/10/12 12:58 PM, Rabary wrote:
<blockquote cite="mid:4F84126A.1030703@gulfsat.mg" type="cite">Hi
mailing,
<br>
<br>
I installed kamailio 3.2, rtpproxy 1.2.1, callcontrol 2.0.15 on a
vz, and cdrtool 8.2.5 and freeradius 2.1.10 on another vz.
<br>
<br>
The 2 vz container have public ip address, and the UAC have
private ip address.
<br>
<br>
I want to use rtpproxy, and the following are what ps and netstat
command returns about rtpproxy:
<br>
teddy@kamailio:~$ ps aux | grep rtpproxy
<br>
teddy 22866 0.0 0.0 3312 800 pts/0 S+ 11:08 0:00
grep rtpproxy
<br>
teddy 31326 0.0 0.0 11360 804 ? Ssl Apr06 0:02
/usr/sbin/rtpproxy -F -l our_public_ip -s udp:localhost 22222
<br>
<br>
teddy@kamailio:~$ sudo netstat -pln | grep rtp
<br>
udp 0 0 127.0.0.1:22222
0.0.0.0:* 31326/rtpproxy
<br>
<br>
And my problem is: when I make calls, for example the UAC1 calls
UAC2, with ngrep I see the UAC1 calls UAC1 and not UAC2, and I
don't know why.
<br>
<br>
In kamailio config file, there are directive WITH_NAT for
everything related to rtpproxy (loadmodule, routing logic, etc) as
you can see in the attached conf file.
<br>
<br>
I try disable the use of this directive WITH_NAT so I disable the
use of rtpproxy in kamailio and it works: when UAC1 calls UAC2,
the UAC2 is ringing.
<br>
<br>
Please tell me what am I doing wrong, or to use rtpproxy without
NAT is not possible ?
<br>
<br>
In attached file the kamailio config file, the diff between
rtpproxy enable and rtpproxy disable, and the result of ngrep and
what is in syslog when I made calls.
<br>
<br>
Thanks in advance.
<br>
<br>
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<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
<a class="moz-txt-link-freetext" href="http://www.asipto.com/index.php/kamailio-advanced-training/">http://www.asipto.com/index.php/kamailio-advanced-training/</a></pre>
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