<div dir="ltr">Thank you both for your replies.<div>I realised eventually that I was indeed missing rtpproxy_offer/answer for the RE-INVITEs.</div><div><br></div><div>Callum, I am using rtpproxy_offer/answer but I will keep that in mind for the future.</div><div><br></div><div>Seems to be working now although I think I need to tune the options in this case. Is there a generally recognised method of re-engaging rtpproxy when sessions already exist? I'm thinking the 'l' flag is required?</div><div><br></div><div>Regards</div><div>Mark.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 11 Jun 2019 at 09:16, Callum Guy <<a href="mailto:callum.guy@x-on.co.uk">callum.guy@x-on.co.uk</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hi Mark,<br><div><br></div><div>Can you confirm which engagement functions you are using? Does the SDP look like its being handled for the RE-INVITE transactions?</div><div><br></div><div>I have dealt with some similar scenarios recently and would highly recommend using the separate rtpproxy_offer() and rtpproxy_answer() methods over the dialog bound rtpproxy_engage() - this will give you a more granular control. Generally speaking I would advise that you identity a loose routed INVITE and check for present of SDP and use this to trigger an offer, somewhat like this:</div><div><br></div><div><font face="courier new, monospace">if (loose_route() && is_method("INVITE") && has_body("application/sdp")) {</font></div><div><font face="courier new, monospace"> rtpproxy_offer(OPTS);</font></div><div><font face="courier new, monospace">}</font></div><div><br></div><div>You'll then need to consider the additional scenarios for last SDP and ACK processing to handle both the answer and edge cases however hopefully this will help you move forward in your config.</div><div><br></div><div>Good luck,</div><div><br></div><div>Callum</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 10 Jun 2019 at 19:55, Răzvan Crainea <<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">Hi, Mark!<br>
<br>
Are you engaging RTPProxy for the re-INVITEs too? Can you check if <br>
RTPProxy times out the session due to lack of audio from one side? You <br>
can see that in the logs.<br>
<br>
Best regards,<br>
<br>
Răzvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" rel="noreferrer" target="_blank">http://www.opensips-solutions.com</a><br>
<br>
On 6/10/19 4:23 PM, Mark Farmer wrote:<br>
> Hi all<br>
> <br>
> I'm trying to solve an issue where if the call is placed on hold via a <br>
> re-INVITE, my end stops the audio at that point so when another <br>
> re-INVITE is received to resume the call, the audio does not restart. I <br>
> can still hear the other party but they cannot hear me because I'm not <br>
> sending any RTP any more.<br>
> <br>
> I'm really struggling to pin this one down. Can anyone give me any clues <br>
> as to why this might be happening?<br>
> <br>
> I'm using OpenSIPS 2.4.5 with RTPProxy 2.0<br>
> <br>
> Many thanks<br>
> Mark.<br>
> <br>
> <br>
> -- <br>
> Mark Farmer<br>
> <a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a> <mailto:<a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a>><br>
> <br>
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</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature">Mark Farmer<br><a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a></div>