<div dir="ltr"><div class="gmail_extra">Hello Salman,</div><div class="gmail_extra"><br></div><div class="gmail_extra">Can you please elaborate on how you got this working? I have the same problem and can't get it to work.</div><div class="gmail_extra"><br></div><div class="gmail_extra">In failure route, I do:</div><div class="gmail_extra">unforce_rtp_proxy()<br></div><div class="gmail_extra">Then when I have a new destination, I do:</div><div class="gmail_extra">rtpproxy_offer("rocie");<br></div><div class="gmail_extra"><br></div><div class="gmail_extra">However, I end up with messed up SDP, in my second invite. It doesn't remove the old IP addresses and only adds the IP addresses again:</div><div class="gmail_extra"><div class="gmail_extra">o=Sonus_UAC 9216 20203 IN IP4 10.160.11.16210.160.11.162a Capabilities </div><div class="gmail_extra">c=IN IP4 10.160.11.16210.160.11.162udio 2311822970AVP 0 8 100 </div></div><div class="gmail_extra"><br></div><div class="gmail_extra"><br></div><div class="gmail_extra">Please let me know how I can fix this.</div><div class="gmail_extra"><br></div><div class="gmail_extra">Thanks.</div><div class="gmail_extra"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jan 6, 2014 at 10:26 AM, Salman Zafar <span dir="ltr"><<a href="mailto:msalman212@gmail.com" target="_blank">msalman212@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><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"><div><div class="h5"><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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style: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>
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<br>
On 12/30/2013 01:11 PM, Salman Zafar wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div>
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></div><br><br clear="all"><br></div></div><span class=""><font color="#888888">-- <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>
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