I've just posted a new release of the application (both in android market place, archos apps lib and csipsimple website).

In this new version there is :

  • Some improvements in the java application. (Network disconnection, battery usage etc... should be better now).
  • Support of 16KHz. This means that even if the media is negotiate to use 8KHz, resampling will provide to the audio layer 16KHz voice, which reduce by half latency.
  • Integration of gsm, speex and g722 codecs. (I've just tested the speex codec, but I'm pretty confident with others since they are managed by pjsip).
  • Auto update sip stack at each update (if needed).

So what are the next steps :

  • Make it more stable
  • Try to integrate the workaround for android 1.5 audio crash at conversation end
  • Add more options (choose codec, priority....)
  • Enable to choose outgoing account for a call (use options)
  • Integration with tel: and with sip: (a plus could be integration with sms or better with our own proto).
  • Improve UI : in call view & screen orientation management.

This will be done for the 0.00-05 (still a FeteDuSip release). After this step, another alpha will provide SIP SIMPLE support. And maybe if legacy issue can be managed without an in app purchase, the g729 codec support (seems to be important for many people).