<div dir="ltr">It should be acceptable to repeat rtpproxy_offer in the failure route - perhaps there is an argument that you'd need to disengage the first session before starting a new one but I don't recall doing that myself. Just to clarify you are using the direction options to manage the bridge interfaces used for each scenario? i.e. rtpproxy_offer("rfie"), rtpproxy_offer("rfei"), rtpproxy_offer("rfee") etc?</div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 19 Nov 2019 at 13:54, Mark Farmer <<a href="mailto:farmorg@gmail.com">farmorg@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>I have been using bridged mode all along.</div><div dir="ltr">I've flipped over to RTPproxy for now, this is my interface configuration, 10.147 being the NATed subnet & 10.150 being the internal subnet:<div><br></div><div>-l <a href="http://10.150.50.51/10.147.52.52" target="_blank">10.150.50.51/10.147.52.52</a> -A <a href="http://10.150.50.51/XXX.XXX.XXX.XXX" target="_blank">10.150.50.51/XXX.XXX.XXX.XXX</a> <-- PUBLIC IP<br></div><div><br></div><div>I think I understand offer/answer params but calling rtpproxy_offer() in both the initial routing and again in the failure route breaks the SDP which I believe is expected.</div><div>If I don't call rtpproxy_offer for the initial INVITE then the SDP is broken for that leg.</div><div><br></div><div>Clearly I'm missing something somewhere...</div><div><br></div><div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 19 Nov 2019 at 13:28, Callum Guy <<a href="mailto:callum.guy@x-on.co.uk" target="_blank">callum.guy@x-on.co.uk</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="auto">You might want to read up on bridge mode, it allows you to meet the finely control which interface is presented during the SDP rewrites.<div dir="auto"><br></div><div dir="auto">All of the information on the various use cases is available in the module docs, I've used both successfully including some pretty complex request routing. </div><div dir="auto"><br></div><div dir="auto">The move to offer/answer with interface specifications works great, you'll just have to fire off the offer with different params when in the failure route so it will override your initial public/public selection from the initial invite processing</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 19 Nov 2019, 12:27 Mark Farmer, <<a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div dir="ltr"><div dir="ltr">Hi Răzvan</div><div dir="ltr"><br></div><div>My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & internal - for Asterisk.</div><div>The issue is that if a call from one registered user to another is rejected & goes to failure_route() then I send the call to an Asterisk box for voicemail which is connected via the internal interface.</div><div><br></div><div>When the call is routed to Asterisk, I need the RTP to flow between RTPproxy & Asterisk on the internal interfaces so I need to have the SDP correct before it hits Asterisk. RTP to & from the phone needs to use the public interface.</div><div><br></div><div>Initial media flow:</div><div>phone<-->OpenSIPS/RTPproxy<-->phone</div><div><br></div><div>Voicemail media flow:</div><div><span style="color:rgb(0,0,0)"><font face="arial, sans-serif">phone<-->OpenSIPS/RTPproxy<-->Asterisk</font></span><br></div><div><span style="color:rgb(0,0,0)"><font face="arial, sans-serif"><br></font></span></div><div>What is the best way to achieve this?</div><div><br></div><div>Many thanks!</div><div>Mark.</div><div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea <<a href="mailto:razvan@opensips.org" rel="noreferrer" target="_blank">razvan@opensips.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Yes, the problem is definitely the fact that you are calling <br>
`rtpproxy_offer()` for the initial invite. Hence, when you run <br>
`fix_nated_sdp()`, you're trying to change the same IP once again - this <br>
is not possile in OpenSIPS.<br>
But I wonder why you need the `fix_nated_sdp()` if you are using <br>
RTPProxy. Can't you just use the `ip_address`[1] field to advertise the <br>
proper IP int he c= line.<br>
<br>
[1] <br>
<a href="https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer" rel="noreferrer noreferrer" target="_blank">https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer</a><br>
<br>
Best regards,<br>
Răzvan<br>
<br>
On 11/13/19 1:51 PM, Mark Farmer wrote:<br>
> Hi everyone<br>
> <br>
> In my failure_route I'm routing to an Asterisk box for voicemail & I <br>
> need to change the SDP c/o parameters to use the correct internal IP <br>
> address but using fix_nated_sdp() is not taking effect.<br>
> <br>
> if (t_check_status("486|408|603")) {<br>
> xlog("CUSTOM_LOG: User replied $T_reply_code - Routing <br>
> to Asterisk Voicemail service.");<br>
> prefix("VMR_");<br>
> rewritehostport("<a href="http://10.150.50.53:2404" rel="noreferrer noreferrer" target="_blank">10.150.50.53:2404</a> <br>
> <<a href="http://10.150.50.53:2404" rel="noreferrer noreferrer" target="_blank">http://10.150.50.53:2404</a>>");<br>
> force_send_socket(udp:10.150.50.51);<br>
> fix_nated_sdp(10,"10.150.50.51");<br>
> <br>
> if (!t_relay()) {<br>
> send_reply(500,"Internal Error");<br>
> }<br>
> exit;<br>
> }<br>
> <br>
> I get the CUSTOM_LOG entry so I know that the route is executing.<br>
> <br>
> Maybe I'm doing something wrong with the flags, I've tried:<br>
> fix_nated_sdp(2,"10.150.50.51");<br>
> fix_nated_sdp(8,"10.150.50.51");<br>
> fix_nated_sdp(10,"10.150.50.51");<br>
> <br>
> But when I examine the SDP in the resulting invite, the c/o parameters <br>
> are never changed.<br>
> I'm using rtpengine_offer/answer in the initial routing, could it be <br>
> related to that?<br>
> <br>
> I'm using OpenSIPS 3.0.1<br>
> <br>
> Best regards<br>
> Mark.<br>
> <br>
> <br>
> <br>
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> <br>
<br>
-- <br>
Răzvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" rel="noreferrer noreferrer" target="_blank">http://www.opensips-solutions.com</a><br>
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</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr">Mark Farmer<br><a href="mailto:farmorg@gmail.com" rel="noreferrer" target="_blank">farmorg@gmail.com</a></div>
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