CSipSimple 0.01 has been released !
30 Jan 2011
By r3gis - Android - Permalink
Pleased to announce that the first release of CSipSimple is now out.
Was planned at the end of the last year but one thing after another I delayed its release to ship more features and more enhancement.
There is two very interesting new enhancements :
* The support of SILK codec. It is the codec developed by Skype. It's very interesting cause can run with low bandwidth (and still high quality) and in HD using a higher bandwidth.
I spent time to understand how to integrate a codec to pjsip cause was my first try to do so. But it will benefits for future codecs ... specially codec2 that I started to implement. Code is here and is almost functional but codec2 is not yet officially available for voip and it's under heavy development :). So for now I will delay its release but if anybody wants to test it's easy to activate in a build ;).
* A completely re-worked method to keep alive the network connection. Previously it was delegated to pjsip. The bad side is that pjsip is not able to add CPU wake locks to android system. As consequence I had to implement an "Alarm" that wake up CPU each x seconds and send a packet on the network to keep the connection alive. The code for that is directly take from the Gingerbread stock SIP application :) and then plug to pjsip system which timers are as consequence disabled.
Among other changes : the voice mails notification support, a way to set codecs priorities per network type, and the activation of the speex accoustic echo canceller (which has to be used only with devices that support vfp : armv7 and upper).
There is a lot of exciting future steps. The more exciting is video... I'm waiting for pjsip 2.0 release to be out officially cause it's a little bit annoying for me to re-merge 2.0 with my changes and with current pjsip trunk. There is also multiple calls management that I have already really well started but not released in 0.01. (Code is there but in background and not accessible for users ;) ).
Comments
Just found csipsimple on the market and I had to come tell you what a great job you have done / are doing. This is so much more polished than anything out there right now. Really fantastic stuff, keep up the great work!
JasonGreat project, I'm interested in. I would like to ask you some questions about pjsip and pjsip-jni, may I? Please email me or name an IRC channel. :) Thanks a lot.
greenviragI've used 0.00-16-00 with my asterisk PBX to great success, but 0.01-01
Markseems not to function at all, or perhaps UI thread crashes. ON connect I can hear the other party for a few seconds and the frontend disappears. The notification bar still shows a call in progress, but no access to it. Force close csipsimple and the call ends. down grading back to 0.00-16-00 fixes the problem. The phone is an HTC raphael110, software is xdandroid 2.2.2 built from source. The call quality, even over 3G is great, though I'm sure the telco will figure out how to stop that eventually. I have not yet setup a build environment for csipsimple, but that is on my lit of things to do soon. please let me know if you are aware of this happening on other platforms. Thanks, Mark
Hi Regis
Again, thanks for all your dedication and great work. I'm one of those who've donated to your project, and will again. Just wanted to let you know that I've also reverted back to 0.00-16-00 due to fairly serious issues with all versions since 1-01 including the nightlies. I've switched from an HTC Legend to a Desire Z on froyo and I don't know if that makes matters worse or not. Most common problem is being unable to hangup calls after connected (ie app stays on the dialer page). With 1-01 and onwards this happens on almost every call. I look forward to continue to use and support your sip client. Keep up the good work.
Phil
Phil