27 Feb
2019
27 Feb
'19
11:04 p.m.
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
28 Feb
28 Feb
12:24 p.m.
New subject: kamailio/asterisk no call with sound if use rtpproxy/rtpengine
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.
4:03 p.m.
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(a)pocos.nl)
escribió:
On Wed, Feb 27, 2019 at 04:04:45PM -0400, PICCORO
McKAY Lenz wrote:
what its the relation fo that question? i already said that use socket and
both are in same machine
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
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.
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4:21 p.m.
New subject: kamailio/asterisk no call with sound if use rtpproxy/rtpengine
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.
5:43 p.m.
El jue., 28 de feb. de 2019 a la(s) 09:21, Daniel Tryba (d.tryba(a)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?
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PICCORO McKAY Lenz