Do you have proper routing rules between the local ips of kamailio and
asterisk? Why aren't you use only external IPs if they are on different
servers? Asterisk has also the option to set external ip. It can reduce
the complexity of doing bridging of signaling and rtp. Once you get that
working
I got bridging working well on internal interfaces in case of simple SIP
calls on a bit other configuration. But editing this config to support
WebRTC causes same problems. I need internal interfaces on asterisk to
completely close external ones (Security etc.).
Can you specify exactly which side received what IP and what you would
expect there? It is not easy to digests lots of logs and also guess what
would you expect to happen...
Cheers,
Daniel
On 24/06/15 15:14, Alexandru Covalschi wrote:
Heh...
Well, I still have troubles with my configuration.
Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user 300 is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87))
Asterisk localip=10.0.0.87, sorry
2015-06-24 16:24 GMT+03:00 Alexandru Covalschi 568...@gmail.com:
Ok, so my scheme.
Kamailio and Asterisk are in Amazon EC2
Kamailio externip=54.197.230.121 localip=10.145.45.103
Asterisk localip=10.145.45.103, externip doesn't matter
Call should flow like
Ok, so my scheme.
Kamailio and Asterisk are in Amazon EC2
Kamailio externip=54.197.230.121 localip=10.145.45.103
Asterisk localip=10.145.45.103, externip doesn't matter
Call should flow like that:
webrtc -- kamailio-externip -- kamailio-localip -- asterisk-localip
but now it's webrtc --
Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log
without fix_nated_contact error behaviour is the same
maybe I should upgrade to 4.3 ?
2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:
Here's the trace on port which I use for ws server. Don't look at
fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
There are no major changes in 4.3 comparing with 4.2 in regards to
websocket -- the implementation is quite mature for a long time.
Looks like websocket connection is not available. Can you look at
javascript debug console in the browser to see what is printing?
Daniel
On 23/06/15 17:23,
Here is it
http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
There are no major changes in 4.3 comparing with 4.2 in regards to
websocket -- the implementation is quite mature for a long time.
Looks like websocket connection is not
I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
works on other then Amazon EC2 environment and I still get this error.
Maybe it is somehow related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn
javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable):
peer -- ec2 --kamailio+rtpengine-- asterisk
scheme
I use advertised adress for SIP and WS connections.
The problem is that on SIP I get one way audio - I can receive audio
from asterisk,
I solved the SIP voice trouble, but WebRTC problem still exists. What kind
of trace I must do to make my post more informative?
2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v
Here's the trace on port which I use for ws server. Don't look at
fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
establish a ws connection properly. Client is SIPML5 demo phone
http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:
I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
works on other then Amazon EC2 environment and I still get this error.
Maybe it is somehow
Hello. I'm trying to set up this (v 4.2 stable):
peer -- ec2 --kamailio+rtpengine-- asterisk
scheme
I use advertised adress for SIP and WS connections.
The problem is that on SIP I get one way audio - I can receive audio from
asterisk, but I can't transmit audio there - my SIP UA tries to send
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