We actually use record_route_preset not record_route, I would have presumed the logic would be the same for both though regarding this. I do not explicitly set double_rr, so it should be the default of on as you point out. My work around of testing and setting manually does appear to be working now. I also wonder about what the default protocol / priority would be to use on a call. For example a SIP URI call where you don't know if the called party is UDP or TCP, it appears to default to UDP. However if the endpoint I'm calling from is using TCP, I would prefer to have the outbound attempt first try TCP, then go to UDP if it fails. The system we are sending calls to actually just prefers TCP all the time, and then fails over to UDP if it can't complete the call. We are still experimenting with this, but due to the large packet sizes we are seeing with video, TCP is working better for us in most situations.<br>
<br>Thanks<br><br>-dg<br><br><br><div class="gmail_quote">On Fri, Mar 5, 2010 at 11:50 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Daniel,<br>
<br>
But record_route() will automatically do double routing if a proto / ip<br>
/ port change is detected between the inbound and outbound interface.<br>
You need to have the "double_rr" enabled (which is by default).<br>
<br>
Could you check how the RR added by OpenSIPS looks like ?<br>
<br>
Regards,<br>
<font color="#888888">Bogdan<br>
</font><div class="im"><br>
Daniel Goepp wrote:<br>
> Oh man...then TWO seconds after sending this, I find:<br>
><br>
> if(proto==TCP)<br>
> {<br>
> log("the SIP message was received over TCP\n");<br>
> };<br>
><br>
> However my other comment of perhaps this should be handled<br>
> automatically by OpenSIPS still stands :)<br>
><br>
> Thanks<br>
><br>
> -dg<br>
><br>
><br>
> On Fri, Mar 5, 2010 at 9:18 AM, Daniel Goepp <<a href="mailto:dan@goepp.net">dan@goepp.net</a><br>
</div><div class="im">> <mailto:<a href="mailto:dan@goepp.net">dan@goepp.net</a>>> wrote:<br>
><br>
> We are doing some interop work with another switch, and it is<br>
> having some trouble with TCP vs UDP. Because of the packet size<br>
> for these specific calls we need to do them TCP. However the<br>
> record-route in our 200 OK has no transport set, and according to<br>
> the RFC, no transport for SIP default is UDP. This means that all<br>
> our signaling is TCP, until we get an ACK back from this box, and<br>
> it then is UDP, but too big and breaks the call. I have found the<br>
> add_rr_param, so I could do a add_rr_param(";transport=tcp"), but<br>
> I only want to do this for calls that are currently using TCP. I<br>
> looked for a function to test the protocol used, but couldn't find<br>
> one. Anyone know what it is? Also, it would seem the appropriate<br>
> thing for OpenSIPS to do would be to automatically put the<br>
> ;transport=xxx in the RR based on the current protocol of the<br>
> dialog. Thoughts on that?<br>
><br>
> Thanks<br>
><br>
> -dg<br>
><br>
><br>
</div>> ------------------------------------------------------------------------<br>
<div><div></div><div class="h5">><br>
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<br>
<br>
--<br>
Bogdan-Andrei Iancu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
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