Kamailio will be back on VoIP Users Conference, the edition scheduled for the 15th of November, 2013, starting at 17:00GMT. It is the time to give an update about what we have new in the upcoming major release, v4.1.0, which is around the corner, scheduled to be out slightly after mid of November.
There are loads of new features, including 10 new modules, among highlights:
- prepaid system – credit control module
- sip-t/i handling
- ephemeral authentication mechanism suitable for WebRTC sessions
- embedded Java interpreter
- SRTP encryption/decryption via rtpproxy-ng to connect WebRTC endpoints with classic SIP phones
- consistent work on IMS extensions
- gzip compress/decompress SIP message body (keep below UDP MTU/save bandwidth)
- Dnssec support and sctp transport as modules
- SLA/BLA (call pickup, blinking lamps) enhancements
We will try to engage many Kamailio developers and community members, therefore be sure you join the conference. At this moment, confirmed participants are:
- Alex Balashov
- Andrew Mortensen
- Carlos Ruiz Díaz
- Charles Chance
- Daniel-Constantin Mierla
- Jason Penton
- Hugh Waite
- Peter Dunkley
- Richard Good
- Victor Seva
There plenty options to dial in via SIP or PSTN, or simply listen/watch audio/video streams on the web – see more details at:
Stay tuned for updates regarding the Kamailio project participants and content of the session!
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