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
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