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>;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 > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
