[SR-Users] The SIP protocol v2 - we're giving up.

[Meftah Tayeb]

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
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>
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-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883