[asterisk-dev] SIPit 31

[Matthew Jordan]

Hey everyone -

Myself, Joshua Colp, and Olle Johansson will all be at SIPit 31 this year.
For myself and Josh, we're going to be focussing heavily on testing out the
PJSIP stack in Asterisk 13 - I'm sure Olle will be up to all sorts of
torturous tests with both Kamailio and Asterisk!

If you're going to be there as well, be sure to let us know - we'd love to
coordinate testing with as many folks as possible. If you aren't going to
be there but you have some ideas for things to test, feel free to reply to
this e-mail as well. Right now, Josh and I have on our "high-level,
non-scenario specific" list:

   - Video/voice calls
   - Hold/unhold
   - Transfers (attended and blind using REFER and INVITE w/ Replaces)
   - UDP, TCP, TLS, and Websocket transports (IPv4 and IPv6)
   - STUN, TURN, and ICE
   - SRTP (SDES and DTLS)
   - DNS SRV
   - UPDATE, re-INVITE
   - PRACK
   - Presence (dialog-info+xml, pidf+xml, xpidf+xml, cpim-pidf+xml)
   - MWI (simple-message-summary)
   - T.38 (UDPTL)
   - Outbound registration
   - Inbound registration (multiple contacts per AOR)
   - COLP
   - DTMF (RFC2833, INFO, inband)
   - Diversion support (inbound and outbound)
   - Caller-ID (From, Remote-Party-ID, P-Asserted-Identity)

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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