2011 April

April 2011

SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of April 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries, using 40 distinct implementations in the interoperability tests.

SIPit, organized by the SIP Forum, is one of the foundations that make SIP work across vendors and implementations. Twice each year, developers from all around the world meet and test, discuss, learn and fix issues both in implementations and standards. During SIPit events, many bugs in the RFCs – or just missing explanations – has been found. Under the leadership of Robert Sparks, SIPit has become the primary event for all SIP developers. Edvina proudly organized SIPit #26 in Stockholm in May 2010.

See the report for SIPit 28 at https://www.sipit.net/SIPit28_summary

I really miss attending SIPit. I do hope I can attend SIPit #29 in the fall. If you are developing SIP software – clients, servers, devices – make sure you attend the next SIPit with your team!  We have a lot to test, from SIP outbound to SRTP, MSRP, IPv6 and TLS.


Sollentuna, Geneva and Prague, April 1st, 2011: The IETF and the IPv6 Forum today launched SIP-six, the new version of SIP that cleans up a lot of misunderstandings and greatly simplifies implementations of SIP. 

  • ”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 our selves that this should be way more simple.”

The SIP-six standard totally removes the dependency of domains and URI syntax. There’s no longer any 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 number as end point 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 single port address with a 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 IP 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.


  • -”We have for a long time pointed out that IPv6 is not only a replacement of IPv4, it’s an enhancement that is the breeding ground for new types of applications.” says Olle E. Johansson, chairman of the SIP-six working group. ”With SIP-six we really use the potential of the IPv6 address space and new technologies, like Mobile IPv6.”

  • -”While the SIP v2 protocol was developed with IPv6 in mind, it never really used the potential of the new IP protocol. With SIP-six we finally solve that issue and move SIP to new areas. By simplifying the protocol and removing DNS and URIs from the design, new developers and vendors will embrace SIP and we’ll see new applications in a short while.” says Bob Plug. ”This will be the final death blow to ISDN, since it’s an easy upgrade to add the IPv6 address prefix and continue to use the same addressing.”

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.