[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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