[Shtoom] recent changes ; planning for 0.3

[Anthony Baxter]

So I've recently completed a fairly major set of reworkings
of the RTP/SDP handling and the audio device handling. In
addition, I've put in a simple minded playout buffer that
should hopefully stop the creeping delay problem that many
people have complained about - it will add a 30ms delay to
incoming calls. In addition, speex support is checked in,
although it's not been tested against any other clients
yet, but I believe I've got it right. I will be checking in support
for a bunch of the G.72x codecs in the next week, after I
test them against cisco's implementation.

So, what's left before 0.3 release? Here's the entries from
TODO:

Definite before 0.3:
     - make sure Doug is up-to-date with recent refactorings
     - all interfaces must be current and correct
     - Asterisk is rejecting our ACK - e.g. to 613 at fwd.pulver.com
     - doug: implement leg bridging (inbound and outbound)
     - doug: use dropCall()'s deferred to generate correct events
     - doug: s.d.conferencing's entrance soundburst bug
     - ui(qt): hide multiple call support for now
     - Some sort of "debug dump" command that dumps relevant info (ui,
       audio, &c) for bugreports.
     - package for Mac, Windows
     - phone: when on a call, decline a second incoming call?
     - documentation, documentation, documentation
     - document Doug API
     - update website
     - create shtoom wiki
     - set svn:executable on the scripts
     - set up a decent roundup instance, load all entries from TODO

Maybe before 0.3:
     - G.72x codec
     - doug: placeCall to a broken address doesn't give generate a
       correct event
     - let doug handle wav and au files natively.
     - MediaLayer: a more sophisticated playout buffer
     - cocoa native UI into it's own thread to workaround broken
       cfreactor
     - stun: timeout the STUN call, generate a nice error
     - sip: retry registration failures (e.g. for testcall)
     - sip: move more stuff into the Dialog class
     - ui(tk): new dialogs that don't block, and have a timeout.
     - sip/call: the state should become an object, encapsulating the
       current state, and the state machine. This makes it _much_
       easier to handle retransmits and the like.
     - RTCP + NAT == RFC 3605

If there's other things you'd like to see, now's the time to speak up.

Anthony


-- 
Anthony Baxter     <anthony at interlink.com.au>
It's never too late to have a happy childhood.

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