Incidentally, the "3pcc draft" is an RFC so make sure you are looking at the latest.
JAIN-SLEE may or may not be a good choice depending upon what you want to do. Its pretty heavy weight machinery to do simple things. If you are interested, check mobicents.dev.java.net. Its usable and stable. ( If bugs exist blame me for it :-) ) The NIST-SIP distribution is the Reference Implementation of the JAIN-SIP standard ( see http://jain-sip.dev.java.net sorry for the shameless advertising) has some call flows for third party call control. Its really very simple. Two of the four flows in the RFC are implemented. See examples/tpcc and please continue the discussion specific to NIST-SIP on the NIST mailing list for the project. Thanks On Thu, 2006-10-05 at 14:33 +0100, Walton, Ashley B (Ashley) wrote: > Hi Warren, > > Thanks very much for the advice. It is much appreciated (I have had a > look at the 3pcc draft) and I will explore JAIN SLEE. > > Regards, > Ashley > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Warren > Crossing > Sent: 05 October 2006 07:39 > To: [email protected] > Subject: Re: [Sip-implementors] SIP Call Control > > Also have a look draft-rosenberg-sip-3pcc-01 talks about the concept, > make sure your useragent supports reinvite (fairly unusual if it > doesn't) > > also consider using something like JAIN SLEE which has a NIST SIP RA and > Third party call control libraries available - i.e. don't write your own > unless you need to! > > Gardell, Steven wrote: > > Many third party call control implementations wind up using a B2BUA > > (back to back user agent) as described in RFC3725 and other documents. > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Walton, > > > Ashley B (Ashley) > > Sent: Thursday, September 28, 2006 8:43 AM > > To: [email protected] > > Subject: [Sip-implementors] SIP Call Control > > > > Gentlemen, > > > > I have started what I suspect is going to be a long journey into SIP > > and despite having done a fair amount of research thus far I get the > > impression I have just begun to scrape the surface. > > > > One thing that I have not been able to gain any clear resolution in my > > > mind about is third party call control interaction for UA's (i.e. by > > CTI > > applications) is defined, I have come across the mention of SIP REFER > > in passing but this does not seem to handle such things in entirety. > > In my mind I would imagine a stateful proxy may be able to influence > > call interaction for a UA/SIP end point (comments?). > > > > Does anybody have any insight in this regard? > > > > Thanks, > > Ashley > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
