[OpenSIPS-Users] Transport identification
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Fri Mar 5 20:50:03 CET 2010
Hi Daniel,
But record_route() will automatically do double routing if a proto / ip
/ port change is detected between the inbound and outbound interface.
You need to have the "double_rr" enabled (which is by default).
Could you check how the RR added by OpenSIPS looks like ?
Regards,
Bogdan
Daniel Goepp wrote:
> Oh man...then TWO seconds after sending this, I find:
>
> if(proto==TCP)
> {
> log("the SIP message was received over TCP\n");
> };
>
> However my other comment of perhaps this should be handled
> automatically by OpenSIPS still stands :)
>
> Thanks
>
> -dg
>
>
> On Fri, Mar 5, 2010 at 9:18 AM, Daniel Goepp <dan at goepp.net
> <mailto:dan at goepp.net>> wrote:
>
> 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?
>
> Thanks
>
> -dg
>
>
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--
Bogdan-Andrei Iancu
www.voice-system.ro
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