Thanks for the info. Looks like I'm dealing with two external problems then. A device that is not doing what it should by specifying the transport, and this other server we are communicating with that regardless of if it get's a UDP request, it will first try TCP on the way out. Ah the joy of working around incompatible devices and servers. I guess this is why we like OpenSIPS so much, it's flexible enough to fix other people's problems ;)
-dg On Sat, Mar 6, 2010 at 12:27 AM, Bogdan-Andrei Iancu <bog...@voice-system.ro > wrote: > Daniel Goepp wrote: > > We actually use record_route_preset not record_route, I would have > > presumed the logic would be the same for both though regarding this. > rr_preset() is for setting your own rr header, overriding the automatic > building of the headers. So you have to red pill - let opensips to do > the job - or the blue pill - you do you yourself, but do it from A to Z > > > 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. > if there is not protocol indication (like SIPS scheme or transport > param), it is UDP - that is what RFC3261 says > > 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. > if you receive the call over TCP and the caller requests TCP for > delivery, you should have the transport=TCP param in the received RURI. > The presence of this param will force opensips to use the indicated > transport for sending the call out. > Se simple fact you receive the call over TCP does not mean anything - > what is important is what protocol is required via RURI. > > Regards, > Bogdan > > 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. > > > > Thanks > > > > -dg > > > > > > On Fri, Mar 5, 2010 at 11:50 AM, Bogdan-Andrei Iancu > > <bog...@voice-system.ro <mailto:bog...@voice-system.ro>> wrote: > > > > 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 <d...@goepp.net > > <mailto:d...@goepp.net> > > > <mailto:d...@goepp.net <mailto:d...@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 > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users@lists.opensips.org <mailto:Users@lists.opensips.org> > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > > Bogdan-Andrei Iancu > > www.voice-system.ro <http://www.voice-system.ro> > > > > > > _______________________________________________ > > Users mailing list > > Users@lists.opensips.org <mailto:Users@lists.opensips.org> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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