<div dir="ltr"><div><div>Hi Razvan,<br></div> I got it working without branching, after banging head a lot I got to learn unforcing drops the media ports for previous rtpproxy offer/answer and after that directing the new flow though rtpproxy flags,IP media works. I am able to traverse from eternal to internal play media and then on failure do external to external with media flowing between public interfaces. Just wondering if you know this method or certify.<br>
</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi, Salman!<br>
<br>
The sockets used by RTPProxy are created when the session is started (the first offer) and cannot be updated afterwards. Therefore the only solution I can see is to configure a per branch scenario, as you mentioned.<br>
<br>
Best regards,<br>
<br>
Razvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<u></u>com</a><div><div class="h5"><br>
<br>
On 12/30/2013 01:11 PM, Salman Zafar wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
Hi,<br>
I have a scenario of playing media at a private-ip media server and<br>
send BUSY, next in failure route bridge call to a public IP. (SIP to SIP).<br>
<br>
So the scenario is as follows:<br>
<br>
UA(Phone1) -> OpenSIPS/RTpProxy(ei) -> Media-Server (Private IP) -> BUSY<br>
-> OpenSIPS(failure route) -> RTpProxy(ee) -> lookup -> (UA Phone2)<br>
<br>
Now the problem is RtpProxy is being offered (EI flags) in first case<br>
where routing to Media sever at private IP, after failure it is again<br>
used with (EE flags), also in corresponding replies.<br>
<br>
The second time RTpProxy does not effect SDP c= and ports in a way to<br>
build media communication. SDP fix directly does not effect rtp ports.<br>
<br>
Is there any way of using RtpProxy differently in fail-over, or I have<br>
to go for rtpproxy per branch?.<br>
<br>
<br>
Thanks in advance.<br>
<br>
--<br>
Regards<br>
<br>
Salman<br>
<br>
<br>
<br></div></div>
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</blockquote>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br><div dir="ltr"><font face="'times new roman', serif">Regards</font><div><font face="'times new roman', serif"><br></font><div><pre cols="72"><font face="'times new roman', serif">M. Salman Zafar<br>
VoIP Professional<br></font><br></pre></div></div></div>
</div>