A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ...
Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call.
typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn
All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /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 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr:46830-> rtpproxy.720.fr:35000
logs of RTP proxy (option -f to screen the debug) :
# rtp ok :
received command "29638_4 L 3962688592-61598521@10.33.146.4 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1" lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 (RTP) guessing RTCP port for caller to be 50187
# rtp not ok :
received command "29571_2 L 3288148592-61598465@10.33.146.4 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1" lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29571_2 6002 77.246.81.133 "
and hop, no "caller's address filled in" ...
it's exactly the same type a SIP call (with the same ip phone, etc...). 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.
Does someone knows something about that ? thanks !
.Samuel Muller. sml@720.fr
Hello,
does the caller SDP come with a public IP? Otherwise, rtpproxy learns the media IP when first rtp packet is sent.
Cheers, Daniel
On 09.07.2009 23:00 Uhr, Samuel Muller wrote:
A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ...
Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call.
typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn
All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /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 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr:46830->rtpproxy.720.fr:35000 http://rtpproxy.720.fr:35000/
logs of RTP proxy (option -f to screen the debug) :
# rtp ok :
received command "29638_4 L 3962688592-61598521@10.33.146.4 mailto:3962688592-61598521@10.33.146.4 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1" lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 http://77.246.81.136:35000/ sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 http://94.198.149.41:50186/ (RTP) guessing RTCP port for caller to be 50187
# rtp not ok :
received command "29571_2 L 3288148592-61598465@10.33.146.4 mailto:3288148592-61598465@10.33.146.4 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1" lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 http://77.246.81.136:35000/ sending reply "29571_2 6002 77.246.81.133 "
and hop, no "caller's address filled in" ...
it's exactly the same type a SIP call (with the same ip phone, etc...). 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.
Does someone knows something about that ? thanks !
.Samuel Muller. sml@720.fr mailto:sml@720.fr
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
are you 100% the UDP ports 6000-up are open in your firewall?
appears mostly a firewall issue.
__________________________ Omar
On Jul 16, 2009, at 5:06 AM, Daniel-Constantin Mierla wrote:
Hello,
does the caller SDP come with a public IP? Otherwise, rtpproxy learns the media IP when first rtp packet is sent.
Cheers, Daniel
On 09.07.2009 23:00 Uhr, Samuel Muller wrote:
A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ...
Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call.
typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn
All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /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 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr: 46830->rtpproxy.720.fr:35000 http://rtpproxy.720.fr:35000/
logs of RTP proxy (option -f to screen the debug) :
# rtp ok :
received command "29638_4 L 3962688592-61598521@10.33.146.4 <mailto:3962688592-61598521@10.33.146.4
77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1"
lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 <http://77.246.81.136:35000/
sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 <http://94.198.149.41:50186/
(RTP)
guessing RTCP port for caller to be 50187
# rtp not ok :
received command "29571_2 L 3288148592-61598465@10.33.146.4 <mailto:3288148592-61598465@10.33.146.4
77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1"
lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 <http://77.246.81.136:35000/
sending reply "29571_2 6002 77.246.81.133 "
and hop, no "caller's address filled in" ...
it's exactly the same type a SIP call (with the same ip phone, etc...). 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.
Does someone knows something about that ? thanks !
.Samuel Muller. sml@720.fr mailto:sml@720.fr
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com/
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hello guys,
thanks for your help.
the descripton of architecture could help :
it's a sip trunk between an IPBX Mitel through a Juniper SIP ALG on the customer side and Kamailio 1.4.4. We're directly connected through public IPs (only 2 hops via dedicated IP transits, that's cool :).
at the SIP side : 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
at the RTP side : 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
what's new I saw since last debugs :
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. 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). 3. a re-Invite after one minute make up the RTP, and sound is ok after. 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 ?
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.
thanks,
-- *Samuel MULLER* sml@720.fr
On Thu, Jul 16, 2009 at 2:19 PM, Omar Mendoza omar@321communications.comwrote:
are you 100% the UDP ports 6000-up are open in your firewall?
appears mostly a firewall issue.
Omar
On Jul 16, 2009, at 5:06 AM, Daniel-Constantin Mierla wrote:
Hello,
does the caller SDP come with a public IP? Otherwise, rtpproxy learns the media IP when first rtp packet is sent.
Cheers, Daniel
On 09.07.2009 23:00 Uhr, Samuel Muller wrote:
A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ...
Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call.
typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn
All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /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 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr :46830->rtpproxy.720.fr:35000 http://rtpproxy.720.fr:35000/
logs of RTP proxy (option -f to screen the debug) :
# rtp ok :
received command "29638_4 L 3962688592-61598521@10.33.146.4 mailto: 3962688592-61598521@10.33.146.4 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1" lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 < http://77.246.81.136:35000/%3E sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 < http://94.198.149.41:50186/%3E (RTP) guessing RTCP port for caller to be 50187
# rtp not ok :
received command "29571_2 L 3288148592-61598465@10.33.146.4 mailto: 3288148592-61598465@10.33.146.4 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1" lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 < http://77.246.81.136:35000/%3E sending reply "29571_2 6002 77.246.81.133 "
and hop, no "caller's address filled in" ...
it's exactly the same type a SIP call (with the same ip phone, etc...). 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.
Does someone knows something about that ? thanks !
.Samuel Muller. sml@720.fr mailto:sml@720.fr
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com/
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-
Daniel-Constantin Mierla -
Omar Mendoza -
Samuel Muller