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 automatically by OpenSIPS still stands :)<br>
<br>Thanks<br><br>-dg<br>
<br><br><div class="gmail_quote">On Fri, Mar 5, 2010 at 9:18 AM, Daniel Goepp <span dir="ltr"><<a href="mailto:dan@goepp.net">dan@goepp.net</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;">
We are doing some interop work with another switch, and it is having some trouble with TCP vs UDP. Because of the packet size for these specific calls we need to do them TCP. However the record-route in our 200 OK has no transport set, and according to the RFC, no transport for SIP default is UDP. This means that all our signaling is TCP, until we get an ACK back from this box, and it then is UDP, but too big and breaks the call. I have found the add_rr_param, so I could do a add_rr_param(";transport=tcp"), but I only want to do this for calls that are currently using TCP. I looked for a function to test the protocol used, but couldn't find one. Anyone know what it is? Also, it would seem the appropriate thing for OpenSIPS to do would be to automatically put the ;transport=xxx in the RR based on the current protocol of the dialog. Thoughts on that?<br>
<br>Thanks<br><font color="#888888"><br>-dg<br>
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