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Hello,<br>
<br>
<div class="moz-cite-prefix">On 04/09/14 09:20, Abhishek Saini
wrote:<br>
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<blockquote
cite="mid:CAPeZvWhFSm+0vNMnNs8v8gJYD6yV_2MyNWwYsKkd=V2yBmcR2Q@mail.gmail.com"
type="cite">
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<div>Hi Daniel,<br>
<br>
</div>
Thanks for reply.<br>
<br>
</div>
I did install patched rtpproxy and did configure it the
way you have described (advertising address - found that
after posting the comment). But it still does not seem to
work.<br>
<br>
</div>
I don't quite know how can i debug, if rtpproxy is actually
being used.<br>
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</blockquote>
use ngrep to look at sip traffic, like:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in
the SDP. Also, the media IP in SDP should change from incoming
INVITE to what is sent out in the IP of rtpproxy.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<blockquote
cite="mid:CAPeZvWhFSm+0vNMnNs8v8gJYD6yV_2MyNWwYsKkd=V2yBmcR2Q@mail.gmail.com"
type="cite">
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Regards,<br>
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Abhishek<br>
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<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Thu, Sep 4, 2014 at 12:34 PM,
Daniel-Constantin Mierla <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:miconda@gmail.com"
target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
no time to look at config, but if you run the sip server on
a private IP behind a port forwarding address, you have to
use also rtpproxy with advertising address -- see the second
parameter of rtpproxy_manage() or search on the web for a
patch to rtpproxy to add advertising address via command
line parameter.<br>
<br>
Cheers,<br>
Daniel
<div class="HOEnZb">
<div class="h5"><br>
<br>
On 03/09/14 12:23, Abhishek Saini wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
<br>
I have setup kamailio 4.1.0 on an EC2 xlarge instance.
The voice and video calls seem to work well when both
the devices are connected to the same network,
however, when one device connects to a different
network (the two devices now are on different
networks), they are able to register on SIP server,
and even call can be triggered and accepted between
the two devices but there is no video/audio
transmission.<br>
<br>
I have setup rtpproxy but i don't know whether it's
working or not.<br>
<br>
Any help on this would be highly appreciated.<br>
<br>
<br>
Following is my kamailio.cfg file:<br>
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<span class="HOEnZb"><font color="#888888">
-- <br>
Daniel-Constantin Mierla<br>
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href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a>
- <a moz-do-not-send="true"
href="http://www.linkedin.com/in/miconda"
target="_blank">http://www.linkedin.com/in/miconda</a><br>
Next Kamailio Advanced Trainings 2014 - <a
moz-do-not-send="true" href="http://www.asipto.com"
target="_blank">http://www.asipto.com</a><br>
Sep 22-25, Berlin, Germany<br>
<br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list<br>
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target="_blank">sr-users@lists.sip-router.org</a><br>
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</font></span></blockquote>
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<br>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
Next Kamailio Advanced Trainings 2014 - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
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