Great News! I'm going to implement SIP-six/SIMPLE over the weekend, shouldn't be a big deal...
Andreas On 04/01/2011 10:54 AM, Olle E. Johansson wrote: > Friends, > > After having spent many years working with the Asterisk SIP channel driver, > Kamailio and the SIPv2 protocol, I have finally realized that this is a dead > end. It's getting nowhere and it's way too complicated to set up, run and > support in working code. > > After realizing this, I started a new standardization project together with > my friends in Canada, Simon and Marc, to develop a working solution based on > the combination of IPv6 and SIP. We have gotten great feedback and now the > IETF, the ITU and the IPv6 forum jointly launches the new standard, SIP-six. > > From the press release: > > "”We realize that 99% of the SIP users use SIP for PSTN phone calls. The > original SIP standards was written with other applications in mind, a vision > that never came true.” said Bob Plug, area director in the IETF. ”That’s why > we sat down and said to ourselves that this should be way more simple.” > > The SIP-six standard totally removes the dependency of domains and URI > syntax. There’s no point in using this, since everyone seems to think that IP > addressing is more than enough. The new standard use part of the vast IPv6 > address space to incorporate the E.164 phone numbers as addresses. This is > the reverse of the reverse phone number usage in the enum standard, which is > no longer needed in SIP-six. > > By using IPv6 mobile IP, phone users register their phones and get access to > their phone number. Users that need security can easily integrate IPsec into > their setup. Media and signalling uses the same addressing scheme and is > mixed so that both SIP-six, RTP and RTCP only uses one port address - but in > this case a port address with 32 bit subaddress identifying the media stream. > This finally solves a lot of the firewall traversal issues that SIP v2.0 had. > With the combination of mobile IP and use of public IPv6 addresses NAT > traversal won’t be an issue. > > The testbed for SIP-six has been running for a year at six choosen large SIP > carriers, with the assistance of Edvina AB in Sweden and ViaGenius in > Montreal, Canada. In an International effort, the testbed is today put in > production and Roboid phones all over the world is automatically connected to > this worldwide network." > > > You will be able to find out more about it here: > http://bit.ly/sipsix > > SIP-six is implemented as a channel driver in Asterisk 2.0, as a replacement > for SIP2.0 in Kamailio 4.0 and a channel module in FreeSwitch - all releases > to be released later today. Softphones for testing will shortly be available > from Blink and Zoiper. > > Looking forward to continue this project with the rest of the > Kamailio/SIP-router community! > Special thanks to Daniel for the reference implementation in Kamailio 4.0! > > Have a nice weekend! > > /Olle > _______________________________________________ > 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
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