Project Release Information
This release supports the Opus,
and g.729 audio codecs.
SFLphone accounts now have an "Auto-answer" mode.
Along with a number of bugfixes,
SFLphone's backend has been migrated from PJSIP 1.14.2 to
Support has been added for multi-codec calls (e.g.,
The backend has also migrated threading libraries from Common C++ to pthread.
This release has few changes but some fairly important bugfixes. It fixes history logic and the instant messaging user interface. It improves the SIP core by adding a keep-alive for account registration and updating the Contact header from 200 OK.
This version offers improved performance, better stability, and many bugfixes. It should be the most stable ever released.
This new version fixes several important bugs. Attended transfer is the main new feature. One can now drag and drop two calls on each other and select between transferring the call or creating a conference call.
Refactoring audio RTP session. Updated synchronization between the transport layer and audio layer. A better implementation of SIP early media playback. A configuration serialization engine. Default Evolution addressbook support and addressbook authentication support.