i have setup asterisk/rtpproxy/kamailio following the guide in https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and later NAT traversal with rtpproxy as i understand in simple we must have: phone/caller-->[-->rtpproxy->kamailio->asterisk->kamailio->rtpproxy -]->phone/dest
But when you make a call between two phones, there is no sound...
...to make it work and get sound have to open the ports (range 10000 to 30000 udp) in the asterisk exposing them to the public ip whre its the kamailio and rtpproxy.
We configured kamailio with rtpproxy (also tried with rtpengine), and when I configure Asterisk and Kamailio in "real time" mode, everything goes well, the extensions are registered, the authentication is in the asterisk side (table sipusers requires ignore the version check parameter if you use kamailio 5+) etc etc .. but there's no sound, we have:
kamailio.bindip="10.10.1.1" desc "kamailio.bindip" kamailio.bindport=5060 desc "kamailio.bindport" asterisk.bindip="10.10.1.2" desc "asterisk.bindip" asterisk.bindport=5060 desc "asterisk.bindport" listen=udp:10.10.1.1:5060 advertise 200.1.1.1:5060
at rtpproxy we have:
/usr/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u kamailio -p /var/run/rtpproxy/rtpproxy.pid -l 10.10.1.1 -A 200.1.1.1 -F -m 10000 -M 30000
why we need to make asterisk open the ports directly to the publlic ip?
N]OTE: the public ip are not a real interface in the kamailio/rtppropxy machine, are provided by the service AWS at amazon! a NAT kind i guess!
Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com
On Wed, Feb 27, 2019 at 04:04:45PM -0400, PICCORO McKAY Lenz wrote:
N]OTE: the public ip are not a real interface in the kamailio/rtppropxy machine, are provided by the service AWS at amazon! a NAT kind i guess!
But how are you calling rtp(proxy|engine) from kamailio? I think you need to call rtpengine with the direction option to accomplish what you want in your setup. Also look at the INVITEs (on kamailio) to debug what happens with regards to SDP rewriting.
hih thanks for your respond, but seems you dont paid attention to my problem, with ports opened and redirected to pulbic ip with the AWS firewalling (tech support) call have sound, but and later NAT traversal with rtpproxy as i understand in simple we must have: phone/caller-->[-->rtpproxy->kamailio->asterisk->kamailio->rtpproxy -]->phone/dest then there's no sound...
El jue., 28 de feb. de 2019 a la(s) 05:25, Daniel Tryba (d.tryba@pocos.nl) escribió:
On Wed, Feb 27, 2019 at 04:04:45PM -0400, PICCORO McKAY Lenz wrote:
N]OTE: the public ip are not a real interface in the kamailio/rtppropxy machine, are provided by the service AWS at amazon! a NAT kind i guess!
But how are you calling rtp(proxy|engine) from kamailio? I think you
what its the relation fo that question? i already said that use socket and both are in same machine
need to call rtpengine with the direction option to accomplish what you
so i use unix socket in the rtpproxy one, later with when i changed to rtpengine i use that:
``` cat kamailio.cfg | grep rtp loadmodule "rtpengine.so" modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") rtpengine_manage("co"); rtpengine_manage("cor"); ``` but if you are taking about what ip/interface we used the provided internally: ``` cat kamailio.cfg | grep listen listen=udp:10.10.1.1:5060 advertise 200.1.1.1:5060 ```
the eth0 reports ip 10.10.1.1 in kamailio server, and AWS provide a kind of NAT with ip 200.1.1.1:5060
want in your setup. Also look at the INVITEs (on kamailio) to debug what happens with regards to SDP rewriting.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On Thu, Feb 28, 2019 at 09:03:30AM -0400, PICCORO McKAY Lenz wrote:
hih thanks for your respond, but seems you dont paid attention to my problem, with ports opened and redirected to pulbic ip with the AWS firewalling (tech support) call have sound, but and later NAT traversal with rtpproxy as i understand in simple we must have: phone/caller-->[-->rtpproxy->kamailio->asterisk->kamailio->rtpproxy -]->phone/dest then there's no sound...
You wrote something else in your original message: exposing asterisk RTP 10000-30000 makes it function.
I read that to be an indication that rtp(proxy|engine) isn't rewriting.
El jue., 28 de feb. de 2019 a la(s) 09:21, Daniel Tryba (d.tryba@pocos.nl) escribió:
You wrote something else in your original message: exposing asterisk RTP 10000-30000 makes it function.
that's the problem here! yes!
I read that to be an indication that rtp(proxy|engine) isn't rewriting.
in fact, the first try was with rtpproxy, but due the DEBUG log are very poor we tried with rtpengine, and now i noted that rtpengine only output in log that: ``` Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28721]: DEBUG: timer run time = 0.000005 sec Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28803]: [1551364784.000249] DEBUG: timer run time = 0.000035 sec Feb 28 14:39:45 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000010 sec Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000007 sec Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28732]: DEBUG: timer run time = 0.000008 sec Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28832]: DEBUG: timer run time = 0.000005 sec ```
BUT: rtpproxy seems are working cos if not there's no SIP sesion stablished! but as i siad, no sound if i not exposed the asterisk ports!
NOTE: audio of the call are udp! so that means are not possible? i guess must be possible due the kamailio can manage an route using the rtp[proxy|engine] right?
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Daniel Tryba -
PICCORO McKAY Lenz