Release Notes: This major milestone provides many bugfixes, new features, and enhancements, including presence subscription, improved NAT support, improved audio quality, and ringtone support.
Release Notes: This is primarily a bugfix release. Some highlights include improvements to the Evolution address book plugin, a major rewrite of the GNOME client, and improved STUN support. In video: Symmetric RTP/RTCP (RFC 4961) was implemented.
Release Notes: This release supports the Opus, iLBC, 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 2.0.1. Support has been added for multi-codec calls (e.g., PCMA outgoing, PCMU incoming). The backend has also migrated threading libraries from Common C++ to pthread.
Release Notes: This release is the first to have been ported to ARM. As a result of this effort, users can now choose to disable many features/dependencies. In addition, video support has been greatly improved, with features such as call transfer, hold, and H.264 profile and level negotiation implemented for video calls. In the Gnome client, instant messaging was reimplemented and as a result, it no longer depends on Webkit. PJSIP has been updated to 1.14.2.
Release Notes: This release adds support for CCRTP 2.0.2. In addition the Gnome client now builds with either GTK+2.0 or GTK+3.0. Compatibility with several SIP providers has been fixed, such as ekiga.org and sip2sip.info. Efforts have been made to improve test coverage and quality assurance, striving to ensure that regressions are caught early. The KDE client is now fully up-to-date with the latest version of the sflphone daemon.
Release Notes: 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.
Release Notes: This version offers improved performance, better stability, and many bugfixes. It should be the most stable ever released.
Release Notes: 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.
Release Notes: 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.
Release Notes: There are some small bug fixes, and advanced tests have been added on several VoIP products such as Cisco, Patton, Freeswitch and Asterisk.