On 04/01/2011 11:54 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.
I call April Fools. :)
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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