lol! On Fri, Apr 1, 2011 at 12:21 PM, Meftah Tayeb <tayeb.mef...@gmail.com>wrote:
> i disagree > i liked the idea > i hop they will by true. > my FUCKING ISP did block sip in leyer7 mode then ipv6 is my friend;) > > On 01/04/2011 11:51, Klaus Darilion wrote: > >> Didn't you wanted to call it "SIP-sexy" >> >> >> >> On 01.04.2011 10:54, 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 >>> >> >> _______________________________________________ >> 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 >> > > > -- > Meftah Tayeb > inum: +883510001288000 > phone: +13477595883 > > > > _______________________________________________ > 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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