<div>John,</div><div><br></div>The o= is the Origin IP, which doesn't really matter, the c= is the connection IP, which is the important one. You probably want to use force_rtp_proxy() to rewrite the sdp and make it go through the rtpproxy, and in your asterisk SIP peer, you can either set the peer IP to 4.5.6.7 and have your proxy re-write the RURI to 1.2.3.4 and relay it along, or you could set outboundproxy=4.5.6.7 in sip.conf on your sip peer and just have opensips relay it. You may also need to setup an on-reply route to capture 180/183 messages with SDP and rewrite the IP's in there too.<div>
<br></div><div>--Adam<br><br><div class="gmail_quote">On Fri, Aug 27, 2010 at 10:45 PM, TeleCube - John <span dir="ltr"><<a href="mailto:john@telecube.com.au">john@telecube.com.au</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff" text="#000000">
Hi Max,<br>
<br>
Thanks for your reply.<br>
<br>
I have rtpproxy running and I think it's all configured ok.<br>
<br>
I can use fix_nated_sdp() with flags and see the changes in the sdp
lines in the packets.<br>
<br>
What I am struggling with is just how to go about telling the asterisk
server and the outbound route that all traffic is to run through my
proxy<br>
<br>
I'm not sure what the purpose of the o= and c= lines are and at which
stage in the packet flow I need to re-write which lines.<br>
<br>
If anyone can offer any pointers that will be awesome.<br>
<br>
Kind regards,<br><font color="#888888">
John</font><div><div></div><div class="h5"><br>
<br>
Max Mühlbronner wrote:
<blockquote type="cite">
<pre> Hello,
Yes, the opensips will rewrite the sdp like you said, so it will contain
the ip of the rtpproxy (or mediaproxy). It can be used to proxy the rtp
traffic to a specific destination.
But the rtpproxy does not necessarily have to be the same machine as
your Opensips. The rtpproxy is a different tool which will be enabled by
setting the relevant modparam settings and also calling the rtpproxy in
your routing-script.
<a href="http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html" target="_blank">http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html</a>
maybe there are also other, better tutorials but this link seems to be a
good start for setting up rtpproxy with opensips.
BR
Max M.
Am 27.08.2010 15:36, schrieb TeleCube - John:
</pre>
<blockquote type="cite">
<pre>Hi,
I have a setup as follows:
1.2.3.4 - Outbound Route
4.5.6.7 - Opensips proxy
4.5.6.8 - Asterisk server
The outbound route will only allow any traffic, signaling and media, via
the 4.5.6.7 IP address. It cannot currently accept media from a
different ip than the signaling.
Currently the astx server sends media direct to the outbound route.
Signaling is fine but I need to force the media stream from the asterisk
server through the opensips proxy.
Am I correct in expecting that nathelper/rtpproxy can do this?
Is it simply a matter of re-writing the sdp detail that will tell the
asterisk server to route media through the opensips ip address?
Any examples will be greatly appreciated.
Thanks,
John
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