SIP/2.0 200 OK Via:SIP/2.0/tcp 192.168.60.130:2378;branch=z9hG4bKPjbaf223aecad0451ab6f5b80e26e5143f;rport From:<sip:22222239 at 192.168.104.2>;tag=1e68f3259c4242d6ba74c69ace4f5f44 To:<sip:2230 at 192.168.104.2>;tag=1482606729-1205334993760 Call-ID:5dbde9f0111241c0b3e109cce578f6a7 CSeq:28477 INVITE Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported:timer Accept:multipart/mixed,application/media_control+xml,application/sdp Contact:<sip:192.168.104.2:5060> Take a look at the contact header - there is no transport parameter, thus the UAS announces UDP as transport for indialog requests. Either a bug or config error in the UAS. Or if there is a proxy in the middle which does protocol change then it should do record routing. regards klaus Sasa Coh schrieb: > Hello! > > See attachment with the pjsua log output. Btw. it works on openser but > not on broadworks. > > Thanks, > Sasa > > > On Wed, Mar 12, 2008 at 9:29 PM, Klaus Darilion > <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote: > > > > Sasa Coh wrote: > > Hello Benny, > > > > I'm testing pjSIP's behavior using pjsua (latest version) and > found out, > > that (by my opinion) pjsua has an error when using TCP transport. > > I was following instructions from documentation (as below): > > > ------------------------------------------------------------------------ > > Using TCP Transport > > > > By default, TCP transport will be created and initialized. > However, TCP > > will not be used automatically unless the destination URL has > > ";transport=tcp" parameter in it. (Note: this behavior may change > once > > we support resolving NAPTR records). > > > > TCP can be specified when registering to server and when sending > > outgoing requests. To use TCP when registering, add > ";transport=tcp" in > > the registrar's URL, for example with "--registrar > sip:example.com <http://example.com> > > <http://example.com>;transport=tcp" option. > > > > Similarly ";transport=tcp" parameter needs to be added in the > > destination URL when making outgoing calls, subscribing presence, or > > sending outgoing MESSAGE request. > > > ------------------------------------------------------------------------ > > > > And now, back to the problem: > > - First, register via TCP - that went OK. > > - Then, in case of an outgoing call, everything is OK too. > > - The error occurs at the time when I put this call on hold. The > > re-INVITE (sent by pjSIP) that goes out (signaling HOLD condition) is > > sent via UDP and not TCP as I would expect! > > > > Am I doing something wrong or is this an error in pjSIP? How can I > > overcome this? > > Show the SIP packets (e.g. using "ngrep -W byline port 5060"). I guess > there is the transport parameter missing in the contact header or in the > record-route header. > > regards > klaus > > > > > Kind regards, > > Sasa > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
