sip-js http://code.google.com/p/sip-js/
unfornately I can't run its demo

sip on the web
http://sip-on-the-web.aliax.net/
(using oversip http://www.oversip.net/
and jssip https://github.com/versatica/JsSIP)
please see the jssip demo:
https://github.com/versatica/jssip-demos

current problem is even chrome canary build
's getUserMedia doesn't work,
according to 
http://www.webrtc.org/running-the-demos,
it should work?

#suggestions:
 I think this is regression of chrome, so should wait a bit longer
to let chrome become normal.
mainly this jsPhone seems using sip over websocket, so we need a sip
proxy server which supports websocket endpoint protocol, that's why I
using oversip
installation should be very simple(http://www.oversip.net/documentation/1.3.x/installation/)


tried freeswitch own flash(flex) client
=purpose=
  using freeswitch mod_rtmp endpoint to communicate with each other
=tried works=
  can connect/register to freeswitch, needs xijing to take a look at it to make it work because we modified freeswitch
=conclusion=
  using freeswitch own flash client
=reference=
  http://wiki.freeswitch.org/wiki/Mod_rtmp