<div dir="ltr"><div class="gmail_extra">Hello Răzvan & Salman,<br></div><div class="gmail_extra"><br></div><div class="gmail_extra">Thank you for your responses.</div><div class="gmail_extra"><br></div><div class="gmail_extra">I was able to fix it by moving rtpproxy_offer to branch route instead of having it in the main route. In failure route, I only needed to do unforce.</div><div class="gmail_extra"><br></div><div class="gmail_extra">Regards,</div><div class="gmail_extra">Ali Pey</div><div class="gmail_extra"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Oct 9, 2014 at 3:31 AM, 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">
<div bgcolor="#FFFFFF" text="#000000">
<div><tt>Hi, Ali!<br>
<br>
For the initial branch (in request route) are you using
engage_rtpproxy()? If so, try to use rtpproxy_offer().<br>
<br>
Best regards,<br>
</tt>
<pre cols="72">Răzvan Crainea
OpenSIPS Solutions
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On 10/09/2014 12:06 AM, Ali Pey wrote:<br>
</div></div></div><div><div class="h5">
<blockquote type="cite">
<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><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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<div>
<div><br>
<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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<span><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
VoIP Professional
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