http://www.vocal.com/sip-1/call-transferring/
On Thu, Nov 22, 2012 at 10:58 AM, Dmytro Bogovych <dmytro.bogov...@gmail.com > wrote: > REFER > http://tools.ietf.org/html/rfc3515 > > > On Thu, Nov 22, 2012 at 3:55 PM, Grant Bagdasarian <g...@cm.nl> wrote: > > Hello, > > > > > > > > I’ve been searching the internet to find an explanation on how SIP > transfer > > works using Re-INVITE and/or UPDATE, but I can’t seem to find a good > source. > > > > > > > > From what I understand(and this is the way we do it), the following > happens: > > > > > > > > Bob=Caller > > > > Alice=Called > > > > John=Transfer party > > > > > > > > 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK. > > > > 2) Alice transfers the call to John using Re-INVITE. > > > > a. Alice calls John. The usual INVITE,Trying,200 OK, ACK. > > > > b. Alice Re-INVITEs Bob using INVITE with adjusted SDP. > > > > 3) Bob is connected to John through Alice in some magical way. I’m > > guessing because the SDP has been changed and for some reason the RTP > stream > > flows between Bob and John through Alice? > > > > > > > > Is this correct? If not, perhaps someone could explain it to me from > > scratch. > > > > > > > > Maybe useful to know that we are using Cisco equipment for call handling > > (VXML and TCL scripts). > > > > > > > > Thanks, > > > > > > > > Grant > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GCM GCC GIT d? s:+ a? C+++ UB++ UL+++ P++ L++ E-- W++ N++ o-- K+ w(++) O M(+) V- PS+ PE Y+ PGP t-- 5-- X R- tv+ b+(++) DI+(++) D++ G e++ h---- r+++ y++++ -----END GEEK CODE BLOCK------
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