<div dir="ltr">Asterisk localip=10.0.0.87, sorry<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div>Ok, so my scheme.<br></div>Kamailio and Asterisk are in Amazon EC2<br></div>Kamailio externip=54.197.230.121 localip=10.145.45.103<br></div>Asterisk localip=10.145.45.103, externip doesn't matter<br><br></div>Call should flow like that:<br></div>webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip<br></div>but now it's webrtc --> kamailio-externip --> kamailio--localip --> asterisk-localip --> kamailio-externip --> peer<br><br></div>I have the voice, but it's wrong scheme, and Asterisk drops call because of retransmissions failure <br><div><div><div><br></div></div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
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...<br>
<br>
Cheers,<br>
Daniel<div><div><br>
<br>
<div>On 24/06/15 15:14, Alexandru Covalschi
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>Heh... <br>
</div>
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. <br>
</div>
<div>Any ideas?<br>
</div>
<div>Asterisk log <a href="http://pastebin.com/MFt9V9qK" target="_blank">http://pastebin.com/MFt9V9qK</a><br>
</div>
<div>Kamailio log <a href="http://pastebin.com/jZceP2Rn" target="_blank">http://pastebin.com/jZceP2Rn</a><br>
</div>
<div>Javascript log <a href="http://pastebin.com/4ZLePyKz" target="_blank">http://pastebin.com/4ZLePyKz</a><br>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-24 1:27 GMT+03:00 Alexandru
Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Well.. Guys, sorry, it was totally my fault.
I just used VPN. </div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-24 0:59 GMT+03:00
Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>I used <a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a>
configuration that 100% works on other then
Amazon EC2 environment and I still get this
error. Maybe it is somehow related to NAT
traversal?<br>
</div>
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra">Kamailio log: <a href="http://pastebin.com/jZceP2Rn" target="_blank">http://pastebin.com/jZceP2Rn</a><br>
</div>
<div class="gmail_extra">javascript log: <a href="http://pastebin.com/9Y4Pv43W" target="_blank">http://pastebin.com/9Y4Pv43W</a><br>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">2015-06-23 20:40
GMT+03:00 Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr">Here is it<br>
<a href="http://pastebin.com/JkkM4M5m" target="_blank">http://pastebin.com/JkkM4M5m</a><br>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-23
18:53 GMT+03:00
Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> 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.<br>
<br>
Looks like websocket
connection is not
available. Can you look at
javascript debug console
in the browser to see what
is printing?<span><font color="#888888"><br>
<br>
Daniel</font></span>
<div>
<div><br>
<br>
<div>On 23/06/15
17:23, Alexandru
Covalschi wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>without
fix_nated_contact
error behaviour
is the same<br>
</div>
maybe I should
upgrade to 4.3 ?<br>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-23
14:08 GMT+03:00
Alexandru
Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr">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<br>
<a href="http://pastebin.com/LvAk2HkP" target="_blank">http://pastebin.com/LvAk2HkP</a><br>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-23
14:03
GMT+03:00
Alexandru
Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr">I
solved the SIP
voice trouble,
but WebRTC
problem still
exists. What
kind of trace
I must do to
make my post
more
informative?<br>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-06-23
10:46
GMT+03:00
Daniel-Constantin
Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hello,<span><br>
<br>
<div>On
23/06/15
04:10,
Alexandru
Covalschi
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>Hello.
I'm trying to
set up this (v
4.2 stable):<br>
</div>
peer
<--> ec2
<--kamailio+rtpengine-->
asterisk<br clear="all">
<div>
<div>
<div>scheme<br>
<br>
</div>
<div>I use
advertised
adress for SIP
and WS
connections.<br>
</div>
<div>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 data to
Kamailio-s
local EC2 IP.</div>
</div>
</div>
</div>
</blockquote>
<br>
</span> you
should grab a
ngrep trace on
server to see
what happens
in the
signaling in
order to be
able to
provide some
hints on
solving it.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<blockquote type="cite"><span>
<div dir="ltr">
<div>
<div>
<div> In case
of WebRTC I
get lot's of
erros:<br>
<br>
Jun 23
01:58:57
kamailio
/usr/sbin/kamailio[18325]:
WARNING:
<core>
[msg_translator.c:2778]:
via_builder():
TCP/TLS
connection
(id: 0) for
WebSocket
could not be
found<br>
Jun 23
01:58:57
kamailio
/usr/sbin/kamailio[18325]:
ERROR:
<core>
[msg_translator.c:1996]:
build_req_buf_from_sip_req():
could not
create Via
header<br>
Jun 23
01:58:57
kamailio
/usr/sbin/kamailio[18325]:
ERROR:
<core>
[forward.c:584]:
forward_request():
building
failed<br>
Jun 23
01:58:57
kamailio
/usr/sbin/kamailio[18325]:
ERROR: sl
[sl_funcs.c:387]:
sl_reply_error():
ERROR:
sl_reply_error
used: I'm
terribly
sorry, server
error occurred
(1/SL)<br>
<br>
</div>
<div>The call
reaches
Asterisk, but
not
vice-versa. No
media is being
transferred.<br>
<br>
</div>
<div>Rtpengine
flags I use:<br>
</div>
<div>For SIP:Â
rtpengine_manage("trust-adress
replace-origin
replace-session-connection
RTP/AVP");<br>
</div>
<div>For WS:Â
rtpengine_manage("trust-address
replace-origin
replace-session-connection
ICE=force
RTP/AVP");<br>
<br>
</div>
<div>Do you
have any ideas
how ti fix
that? I also
make REGFWD's
to Asterisk<br>
</div>
<div>-- <br>
<div>
<div dir="ltr">Alexandru
Covalschi<br>
ABRISS-Solutions
<div>VoIP
engineer and
system
administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</span>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><span><font color="#888888">
</font></span></pre>
<span><font color="#888888">
</font></span></blockquote>
<span><font color="#888888">
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</font></span></div>
<br>
_______________________________________________<br>
SIP Express
Router (SER)
and Kamailio
(OpenSER) -
sr-users
mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru
Covalschi<br>
ABRISS-Solutions
<div>VoIP
engineer and
system
administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru
Covalschi<br>
ABRISS-Solutions
<div>VoIP
engineer and
system
administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru
Covalschi<br>
ABRISS-Solutions
<div>VoIP
engineer and
system
administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</div>
</div>
</div>
<br>
_______________________________________________<br>
SIP Express Router (SER) and
Kamailio (OpenSER) -
sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru
Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and
system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system
administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</div></div></div>
</blockquote></div><br><br clear="all"><br>-- <br><div><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>