SIPit is the main interoperability event for all things SIP. It’s organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet’s IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us to build a network with other developers in the business, a network which helps when we have bugs that involve interoperability with these devices or servers. SIPit has proven important for the success of Asterisk, and thus it is also important for everyone in the Asterisk community.
Now, when we are working on the next long-term release (1.8) we really need to test again and make sure that we interoperate properly. New stuff, like Terry’s SRTP branch, my RTCP work and the call completion and caller ID update work needs serious testing. We need feedback to be able to fix the issues with the TCP and TLS support. (more…)
© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.
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