asterisk users - Apr 2011 - The SIP channel driver - I'm giving up.

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

In article <B4ED7E5E-56D7-4AB3-8F9C-0D120A741700 at edvina.net>,
Olle E. Johansson <oej at edvina.net> wrote:
> 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.
Hmmm, fancy announcing this on 1 April....

-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org