X-Git-Url: http://sjero.net/git/?p=linphone;a=blobdiff_plain;f=NEWS;h=413970e6a9df58885d12ad43a4d936a642b06dd1;hp=ceab2b1794431782a0a6e5523ff2ba5dc118e10f;hb=5f109bfb55224b69dd598d95c2c0dd47fa87b960;hpb=b39938c7fd55f8feaa135bdc3bbbc304f7e72b44 diff --git a/NEWS b/NEWS index ceab2b17..413970e6 100644 --- a/NEWS +++ b/NEWS @@ -1,10 +1,91 @@ -linphone-3.3.0 -- ????????? +linphone-3.xxx -- + * fix bug in zRTP support (upgrade required) + * + +linphone-3.5.2 -- February 22, 2012 + * updated oRTP to 0.20.0 + * updated mediastreamer2 to 2.8.2 + * added ZRTP media encryption + * added SILK audio codec + +linphone-3.5.1 -- February 17, 2012 + * gtk - implement friend search by typing into the friendlist, and friend sorting + +linphone-3.5.0 -- December 22, 2011 + * added VP-8 video codec + * added G722 audio codec + * added SIP/TCP and SIP/TLS + * added SRTP media encryption + * Audio conferencing + * UI: call history tab, menu simplified + * UI: cosmetics for incall views + * UI: integration with libnotify + * UI: show registered SIP accounts + * Fixes for MacOS X, and uses GtkQuartz engine + +linphone-3.4.3 -- March 28, 2011 + * Fully ported to mac os x with gtk-osx (menu integration, bundle generation with "make bundle", sound I/O improved) but still audio only + * Fix stupid warning "no response" that sometimes arrived at end of calls + * limit the size of the log window (to prevent memory drain) + * limit the size of the SDP message by removing unnecessary information (for well known codecs, for H264). + This is to prevent SIP messages from being discarded by routers on the internet when they exceeds in size the internet MTU. + * other sip bugfixes + Requires mediastreamer-2.7.3 + +linphone-3.4.2 -- March 3rd, 2011 + * fix problems with webcams on windows + Requires mediastreamer-2.7.2 + +linphone-3.4.1 -- February 17th, 2011 + * bugfixes + * gtk executable is renamed "linphone" (was linphone-3 before) + Requires mediastreamer-2.7.1 + +linphone-3.4.0 -- February 7th, 2011 + * implement multiple calls feature: + - call hold (with possibility to play a music file) + - call resume + - acceptance of 2nd call while putting the others on hold + - creation of another outgoing call while already in call + - blind call transfer + - attended call transfer + **CAUTION**: LinphoneCoreVTable has changed: pay attention to this when upgrading an old application to a newer liblinphone. + * improve bandwidth management (one b=AS line is used for audio+video) + * improvements in the echo limiter performance + * implement a echo calibration feature (see linphone_core_start_echo_calibration()). + * stun support bugfixes + * possibility to use two video windows, one for local preview, one for remote video (linphonec only) + * optimize by not re-creating streams when SDP is unchanged during a reinvite + * support for sending early media + * doxygen doc and javadoc improvements + * based on mediastreamer-2.7.0, please refer to mediastreamer NEWS for changes. + +linphone-3.3.2 -- July 1st, 2010 + * fix crash when setting firewall address in gtk interface + * fix crash while closing video window on windows + * fix un-sent BYE message in some rare cases. + Requires: + mediastreamer2-2.6.0 + ortp-0.16.3 + +linphone-3.3.1 -- June 3, 2010 + * fix bugs when carrying non ascii displaynames in SIP messages + * fix crash when codecs are incompatible + * fix bug with streams not restarted in case of reinvites + Requires: + mediastreamer2-2.5.0 + ortp-0.16.3 + +linphone-3.3.0 -- May 19, 2010 * liblinphone is ported to iphoneOS and Google Android * Internal refactoring of liblinphone (code factorisation, encapsulation of signaling) * enhancements made to presence support (SIP/SIMPLE) - -linphone-3.2.2 -- ????????? + * new icons + * new tabbed ui + * be nat friendly using OPTIONS request and using received,rport from + responses. + * use stun guessed ports even if symmetric is detected (works with freeboxes) * improve bitrate usage of speex codec * allow speex to run with vbr (variable bit rate) mode * add speex/32000 (ultra wide band speex codec)