Friends, After having spent many years working with the Asterisk SIP channel driver 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 Asterisk community! Have a nice weekend! /Olle -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
