<div><a href="http://www.vocal.com/sip-1/call-transferring/">http://www.vocal.com/sip-1/call-transferring/</a><br></div><div><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Nov 22, 2012 at 10:58 AM, Dmytro Bogovych <span dir="ltr"><<a href="mailto:dmytro.bogovych@gmail.com" target="_blank">dmytro.bogovych@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">REFER<br>
<a href="http://tools.ietf.org/html/rfc3515" target="_blank">http://tools.ietf.org/html/rfc3515</a><br>
<div><div class="h5"><br>
<br>
On Thu, Nov 22, 2012 at 3:55 PM, Grant Bagdasarian <<a href="mailto:GB@cm.nl">GB@cm.nl</a>> wrote:<br>
> Hello,<br>
><br>
><br>
><br>
> I’ve been searching the internet to find an explanation on how SIP transfer<br>
> works using Re-INVITE and/or UPDATE, but I can’t seem to find a good source.<br>
><br>
><br>
><br>
> From what I understand(and this is the way we do it), the following happens:<br>
><br>
><br>
><br>
> Bob=Caller<br>
><br>
> Alice=Called<br>
><br>
> John=Transfer party<br>
><br>
><br>
><br>
> 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK.<br>
><br>
> 2) Alice transfers the call to John using Re-INVITE.<br>
><br>
> a. Alice calls John. The usual INVITE,Trying,200 OK, ACK.<br>
><br>
> b. Alice Re-INVITEs Bob using INVITE with adjusted SDP.<br>
><br>
> 3) Bob is connected to John through Alice in some magical way. I’m<br>
> guessing because the SDP has been changed and for some reason the RTP stream<br>
> flows between Bob and John through Alice?<br>
><br>
><br>
><br>
> Is this correct? If not, perhaps someone could explain it to me from<br>
> scratch.<br>
><br>
><br>
><br>
> Maybe useful to know that we are using Cisco equipment for call handling<br>
> (VXML and TCL scripts).<br>
><br>
><br>
><br>
> Thanks,<br>
><br>
><br>
><br>
> Grant<br>
><br>
><br>
</div></div>> _______________________________________________<br>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
><br>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</blockquote></div><br><br clear="all"><div><br></div>-- <br><div>-----BEGIN GEEK CODE BLOCK-----</div><div>Version: 3.12</div><div>GCM GCC GIT d? s:+ a? C+++ UB++ UL+++ P++ L++ E-- W++ N++ o-- K+ w(++)</div><div>O M(+) V- PS+ PE Y+ PGP t-- 5-- X R- tv+ b+(++) DI+(++) D++</div>
<div>G e++ h---- r+++ y++++</div><div>-----END GEEK CODE BLOCK------</div><br>
</div>