Hello guys,<br><br>thanks for your help.<br><br>the descripton of architecture could help :<br><br>it's a sip trunk between an IPBX Mitel through a Juniper SIP ALG on the customer side and Kamailio 1.4.4.<br>We're directly connected through public IPs (only 2 hops via dedicated IP transits, that's cool :).<br>
<br>at the SIP side : <br>sip phone (10.x.y.41) -> (10.x.y.33) ipbx (94.198.148.33) -> sip alg -> (77.246.81.132) kamailio -> (77.246.81.136) audiocodes -> pstn<br><br>at the RTP side :<br>sip phone (10.x.y.41) -> sip alg ( 94.198.148.41, as reflect of the sip phone lan ip) -> (77.246.81.133) rtp proxy -> (77.246.81.136) audiocodes -> pstn<br>
<br>what's new I saw since last debugs :<br><br>1. nat_uac_test (test number "2") does not work well : it detects NAT, but the SIP ALG does its job : it masks the sip uac.<br>2. dns lookups does not work, I had to change the fqdn of the IPBX to its real ip (but it's another problem, rtp weird stuff happens yet).<br>
3. a re-Invite after one minute make up the RTP, and sound is ok after.<br>4. in the logs, the return of $mb doesn't include all the full SIP message each time, only partially as if the buffer was too small (maybe there's nothing here, but I worked too much on the case erfff, I don't know where to search) - is that a reflect of the SIP message Kamailio is taking in charge ?<br>
<br>the RTP ports are full opened between 6000 and 64000 UDP, checked and confirmed - SIP ALG is a Netscreen from Juniper, works well, checked ok.<br><br>thanks,<br><br clear="all">--<br><b>Samuel MULLER</b><br><a href="mailto:sml@720.fr">sml@720.fr</a><br>
<br><br><div class="gmail_quote">On Thu, Jul 16, 2009 at 2:19 PM, Omar Mendoza <span dir="ltr"><<a href="mailto:omar@321communications.com">omar@321communications.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
are you 100% the UDP ports 6000-up are open in your firewall?<br>
<br>
appears mostly a firewall issue.<br>
<br>
__________________________<br><font color="#888888">
Omar</font><div><div></div><div class="h5"><br>
<br>
On Jul 16, 2009, at 5:06 AM, Daniel-Constantin Mierla wrote:<br>
<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br>
<br>
does the caller SDP come with a public IP? Otherwise, rtpproxy learns the media IP when first rtp packet is sent.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<br>
On 09.07.2009 23:00 Uhr, Samuel Muller wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ...<br>
<br>
Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ???<br>
So I don't have any RTP, so no sound during the call.<br>
<br>
typical call schema :<br>
sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn<br>
<br>
All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point.<br>
rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) :<br>
/usr/sbin/rtpproxy -s udp:77.246.81.133 35000 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 77.246.81.133 -m 6000 -M 64000<br>
kamailio is ok too in the config : kamailio IPv4 UDP <a href="http://sip.720.fr" target="_blank">sip.720.fr</a>:46830-><a href="http://rtpproxy.720.fr:35000" target="_blank">rtpproxy.720.fr:35000</a> <<a href="http://rtpproxy.720.fr:35000/" target="_blank">http://rtpproxy.720.fr:35000/</a>><br>
<br>
logs of RTP proxy (option -f to screen the debug) :<br>
<br>
# rtp ok :<br>
<br>
received command "29638_4 L <a href="mailto:3962688592-61598521@10.33.146.4" target="_blank">3962688592-61598521@10.33.146.4</a> <mailto:<a href="mailto:3962688592-61598521@10.33.146.4" target="_blank">3962688592-61598521@10.33.146.4</a>> 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1"<br>
lookup on ports 6008/6010, session timer restarted<br>
pre-filling callee's address with <a href="http://77.246.81.136:35000" target="_blank">77.246.81.136:35000</a> <<a href="http://77.246.81.136:35000/" target="_blank">http://77.246.81.136:35000/</a>><br>
sending reply "29638_4 6010 77.246.81.133<br>
"<br>
caller's address filled in: <a href="http://94.198.149.41:50186" target="_blank">94.198.149.41:50186</a> <<a href="http://94.198.149.41:50186/" target="_blank">http://94.198.149.41:50186/</a>> (RTP)<br>
guessing RTCP port for caller to be 50187<br>
<br>
# rtp not ok :<br>
<br>
received command "29571_2 L <a href="mailto:3288148592-61598465@10.33.146.4" target="_blank">3288148592-61598465@10.33.146.4</a> <mailto:<a href="mailto:3288148592-61598465@10.33.146.4" target="_blank">3288148592-61598465@10.33.146.4</a>> 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1"<br>
lookup on ports 6000/6002, session timer restarted<br>
pre-filling callee's address with <a href="http://77.246.81.136:35000" target="_blank">77.246.81.136:35000</a> <<a href="http://77.246.81.136:35000/" target="_blank">http://77.246.81.136:35000/</a>><br>
sending reply "29571_2 6002 77.246.81.133<br>
"<br>
<br>
and hop, no "caller's address filled in" ...<br>
<br>
it's exactly the same type a SIP call (with the same ip phone, etc...).<br>
I can precise that rtp proxy is running on the same machine that kamailio (v.1.4.4), but listen on a dedicated sub-if.<br>
<br>
Does someone knows something about that ? thanks !<br>
<br>
<br>
.Samuel Muller.<br>
<a href="mailto:sml@720.fr" target="_blank">sml@720.fr</a> <mailto:<a href="mailto:sml@720.fr" target="_blank">sml@720.fr</a>><br>
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</blockquote>
<br>
-- <br>
Daniel-Constantin Mierla<br>
<a href="http://www.asipto.com/" target="_blank">http://www.asipto.com/</a><br>
<br>
<br>
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</blockquote>
<br>
</div></div></blockquote></div><br>