<div dir="ltr"><div dir="ltr">Hi Răzvan,</div><div dir="ltr"><br></div><div>Very kind of you to review this problem. I will go ahead and raise an issue as requested. </div><div><br></div><div>I can confirm that I have had some success using the offer/answer model but it certainly adds a level of complexity which would be best dealt with behind the scenes by the module especially where I am operating in bridge mode - the module e/i flags get a bit more challenging when you are implementing in the reply routes for example.</div><div><br></div><div>Many thanks,</div><div><br></div><div>Callum</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 20 May 2019 at 13:28, Răzvan Crainea <<a href="mailto:razvan@opensips.org">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">Hi, Callum!<br>
<br>
I've just checked this and this looks like a bug in the <br>
rtpproxy_engage() function, not handling properly scenarios where late <br>
negociation is happening in the middle of the call. Please open a bug <br>
report for this on our tracking system[1].<br>
In the meantime, the only solution is to work rtpproxy manually, using <br>
the offer/answer model.<br>
<br>
[1] <a href="https://github.com/OpenSIPS/opensips/issues" rel="noreferrer" target="_blank">https://github.com/OpenSIPS/opensips/issues</a><br>
<br>
Best regards,<br>
Răzvan<br>
<br>
On 5/15/19 7:36 PM, Callum Guy wrote:<br>
> Hi All,<br>
> <br>
> I am working on a problem where for a few destinations my OpenSIPs is <br>
> receiving RE-INVITE messages with late SDP. This is causing a breakdown <br>
> in the rtpproxy engagement and causing the audio to fail mid call.<br>
> <br>
> The OpenSIPs deployment is acting as a SIP proxy which traverses NAT and <br>
> rtpproxy is used in bridging mode. I am using rtpproxy_engage to tie the <br>
> integration to the dialog session and this is for all other purposes <br>
> working as expected.<br>
> <br>
> My failure scenario is when the remote system sends a RE-INVITE message <br>
> which includes no SDP. This passes through to my FreeSWITCH server which <br>
> responds with a 200 including SDP. This message is processed fine and <br>
> interacts with rtpproxy as expected and provides the remote with the <br>
> correct public IP and port for RTP (the same as returned during call <br>
> setup). In response the remote system returns an ACK with SDP which <br>
> triggers an OpenSIPs error message (below) which results in the remotes <br>
> public IP being passed through in SDP which causes the FreeSWITCH to <br>
> start sending RTP direct resulting in one way audio as the media server <br>
> is not publicly accessible.<br>
> <br>
> *ERROR:rtpproxy:engage_force_rtpproxy: not a late negotiation - ACK <br>
> cannot have SDP body*<br>
> <br>
> As I understand it the FreeSWITCH behaviour is OK, although I am not <br>
> clear why it feels the need to resend the SDP. All I want to happen in <br>
> this scenario is for rtpproxy module to re-write the SDP in the way it <br>
> has for all previous messages. I am very interested to hear if there is <br>
> any reason for rtpproxy to disallow late negotiation in this scenario, <br>
> if anyone can point to a relevant RFC that would be interesting!<br>
> <br>
> Is there any way around this other than some sort of manual SDP re-write <br>
> (not helpful to me as I am using a pool of rtpproxy instances)? Might I <br>
> have more luck with offer/answer or indeed rtpengine?<br>
> <br>
> I've illustrated the scenario better on the following link (sngrep paste):<br>
> <br>
> <a href="https://gist.githubusercontent.com/spacetourist/ef0478c0bf4e2d736f9b5663042087dd/raw/6f0a984a1a2838e7e2c4539f059fd68935a3b0b1/gistfile1.txt" rel="noreferrer" target="_blank">https://gist.githubusercontent.com/spacetourist/ef0478c0bf4e2d736f9b5663042087dd/raw/6f0a984a1a2838e7e2c4539f059fd68935a3b0b1/gistfile1.txt</a><br>
> <br>
> Thanks, looking forward to any advice!<br>
> <br>
> Best regards,<br>
> <br>
> Callum<br>
> <br>
> <br>
> <br>
> <br>
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-- <br>
Răzvan Crainea<br>
OpenSIPS Core Developer<br>
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