- Update to 5.3.1015 (no changelog).
- Update to 5.3.95 (no changelog).
- keep openldap patches separated from the source tarball
- Remove the unneeded xsd build dependency
- fix openldap build
* update openldap-bc.tar.bz2 to include openldap upstream
fix for ITS#10011
* add autoreconf command to regenerate a working configure
- Update to 5.3.37 (no changelog).
- Update to 5.3.26 (no changelog).
- Update to 5.3.7:
ADDED:
* Added notion of roles for conferences, so that there can be speakers and listeners.
* Reactions with IM conversations
* New video codec policy, so that hardware-accelerated codecs are prioritized in offer/answer.
* AV-1 codec
* New LinphoneAlert object used to report various QoS alerts during calls.
* SRTP AES GCM mode
CHANGED:
* TLS Client certificate request authentication callback removed (due to mbedtls update).
Application using TLS client certificate must provide it before any TLS connexion needing it.
* Refactoring of LinphoneAddress object implementation, leading to greater internal simplicity and performance.
- Clean up spec file
- Add missing deps on TW.
- Move Obsoletes/Provides to srpm:linphoneqt
(brpm:linphone-desktop) which is what it should have been.
https://forums.opensuse.org/t/tumbleweed-zypper-dup-wants-to-replace-linphone-with-linphone-cli/170249
- Add Obsoletes/Provides to establish an upgrade path for zypper.
https://lists.opensuse.org/archives/list/factory@lists.opensuse.org/thread/DNQSYPMB66XPGRKG6LPXCIH6VPSISG25/
- Fix BuildRequires lime
- Fix error linking on Fedora
- Disable SLP by default for Factory and ALP (bsc#1214884)
- Update to 5.2.98 (no changelog)
- Drop the manual, since it's very outdated and has been replaced with
an online Wiki.
- Update to 5.2.75 (no changelog). This fixes boo#1212956.
- Add set_current_version.patch.
- Update to version 5.2.50:
* Fix bug related to conference created with ICE DTLS SRTP encrypted call
* Rework tone manager:
- Remove reason field from LinphoneToneDescription
- Store tones by ID
- choose tone based on the reason when a call ends
* Do not override lime capability when enabling lime
* Do not allow to add dependency of an account to itself
* Allow to enable or disable tone indications from the core settings
* Fixed Content-Disposition (and others) headers issue in multiparts
* Update chat room flags in DB when calling insertChatRoom
* Fix an issue where on-hold music is played erroneously through a call that has been resumed
* Send invitations to the list of participants to invite and not to all invitees
-Disable QRCode image generation for tumbleweed
* liblinphone is not currently compatible with zxing-cpp 2.0.0
- Update to version 5.2.10:
* Video conferencing feature with "mosaic" and "active speaker + thumbnails" layouts
* Added conferencing scheduling API
* Support for post-quantum robust encryption for ZRTP calls
* Rtp bundle can be enabled per LinphoneAccount
* New APIs on Friend object to be able to set more info
* QRCode image generation
* Licence becomes AGPLv3
- Update to version 5.1.58
https://gitlab.linphone.org/BC/public/liblinphone/-/tags/5.1.58
- Remove RPATH (instead of ingnore) to be able to build on Fedora (no lang)
- Update to version 5.1.48.
* Improved how device ringtone is obtained
* Fixed DNS servers list not updated when network changes
* Set ConferenceId also if conference event package is not enabled
* Make DTLS-SRTP to start immediately after the default candidate pair is verified by ICE.
* increase tests timeout
* Fixed audio echo canceller calibration not being done using speaker on Android
* Fixed audio focus not taken when doing echo canceller calibration if it is disabled for calls
* Fix accepting an sdp with mid but no correponding extmap attribute when it shouldn't
- Use rpmlintrc to ignore RPATH because the environment variable does not work anymore on TW.
- Remove fix that doesn’t seem to be necessary anymore.
- Update to version 5.1.42:
* Fixed incoming call ringtone being played even if device is in do not disturb mode
* Fixed call log duration not available in End state
* Fixed call duration not returning 0 if call didn't connect
* Fixed BufferedReader not closed
* Fixed adding a phone number with an invalid label to a Friend
- Update to version 5.1.24:
* LinphoneRecorder API added to record voice messages
* A recommended video definitions list that filter the supported video definitions list.
* LDAP connectivity into LinphoneMagicSearch API (to lookup contacts by number or name).
* Java wrapper no longer catches app exceptions that happen in listeners.
* linphone_core_enable_mic() is changed to be persistent accross calls and conference.
* To protect against "SIP digest leak", MD5 and digestion without qop=auth can be disabled
by configuration
- build static libldap boo#1198407
- revert license tag change
- do not add bundled libraries as rpm required libs boo#1198407
- do not add bundled libraries as rpm provided libs boo#1198407
- Update to version 5.0.70:
* add tests for register with unresponsive server
* Add telephone event payload from the core if result is empty aka : the first.
* Repair tests failing due to missing root certificate after changing certificate on linphone.or
- Update to version 5.0.67:
* Fix linphone_core_enable_mic() not working for a local conference
* Fix stream direction in remote params
* fix start srtp dtls immediately
* Fix non-working tunnel for incoming calls made without ICE
* Keep ringing audio focus for incoming early media call as soundcard will use RING modei
* Avoid notify ringing of in IncomingEarlyMedia state in order to not send 180 after 183
* Write down call decline reasons into test
* Added config setting to disable audio focus requests
* Fix issue when soundcard may not be used after early media when handling multiple calls
* Do not connect audio to conference during early media
* fix crash get address when participants list is empty
* enable mic by default
- fix license tag
- Update to version 5.0.53:
* Update log file only if enabled to avoid unwanted file creation
* Use belle_sdp_attribute_parse() so that headers are parsed with the correct type
- Update to 5.0.49:
* Fix crash when configuring tones while core is not yet started
- Update to 5.0.47:
* Fix broken reading of video capture device from config file
* Fix crash on PUBLISH when SIP-Etag is missing
* Correctly route sound to speaker after early media and when cancelling call
* Attempt to make local network permission detection more reliable
- Update to 5.0.36:
* Use UTF8 instead of locale in chat message modifiers`
* Fix bad chat room when creating a call
* Crash on ec-calibration : Use tone sent callback only on MS_DTMF_GEN_EVENT
* Added missing conference APIs
* Play ring tone only if tone indications are enabled
* Fix tonemanager on infinite rings and wrong ring type
* fix crash of kickOffConnectivity
* Add option to deactivate potentially weak digest authentication schemes
* Fix issue when receiving an INVITE with ICE and rtcp-mux
* Fix call repair in case of multi account
* Fix regression with ICE not setting candidates correctly for completed check-lists
* Improve reliability of account creation by increasing account creation timeout to 30s
* Change contact address if call in IncomingReceived state is added to conference
* fix crash when session refresh after BYE received
* Repair streaming from file feature of AudioStream/VideoStream
* Fixed error logs showing CoreManager's core being const
* Logging facility optimization
* Fix bug with ChatRoomParams::isGroup() erroneously returning true for some basic chatrooms
* Count unread chat messages in all Chat Rooms with a weak address testing
* avoid to downgrade chat message participant state and add unitest
* Stop audio stream when setting new device
This fix allow changing device on Desktop while ringback
* Audio : Allow setting NULL device (case of no cards available)
* Fixed call to content.isFileEncrypted() on a FileTransferContent
* Fixed mic gain
- Update to 5.0.0:
* Support of Capability Negociation framework - RFC5939
limited to media encryption choice (None, SRTP, DTLS-SRTP, ZRTP)
* New API to manage SIP accounts: LinphoneAccount and LinphoneAccountParams
replacing LinphoneProxyConfig which is now deprecated.
LinphoneProxyConfig remains fully usable for backward compatibility with previous version.
* New implementation of LinphoneAccountCreator relying on http REST API.
* Added LDAP contact provider API integrated with LinphoneMagicSearch
* Added asynchronous API to the LinphoneMagicSearch API (for contact searching).
* Fixed erroneous ICE ufrag and passwd parameters sent in reINVITE while ICE
was refused previously.
* Fixed swift and C# wrappers corner-case usage issues.
- Build and use Belledonne's version of libldap
- delete build-liblinphone4-with-mediastreamer5.patch
- add linphone-build-jsoncpp.patch
- add BuildRequires: jsoncpp-devel
- removed unnecessary baselibs.conf
- Update to 4.5.25:
* Fix an issue with ICE ufrag and passwd sent erroneously during reINVITE
* Fixed crash in message tests suite
* Prevent crash when receiving non-ephemeral message with linphone cpim namespace
* Fixed mic gain
- add build-liblinphone4-with-mediastreamer5.patch
- fix several RPMLINT warning
- Update to 4.5.21:
* Improved metered active network while in background check,
update network reachability when background/foreground
state changes
* Fix setting output audio device that was based on ringer device
on specific call state
* Handle situation where a participant with an invalid address is
added to a chat room
* Fix conference server memory leak
- Update to 4.5.15:
* Don't play the local ringtone when callkit is used
* fix memory leak
* Fix : on incoming call, do not start the ring tone if the
call's state has been modified by listeners
- Update to 4.5.10:
* Fix Callkit audio bug producing only with some old ios devices
* Stop file player to play a file when stopping tone is requested
- Update to 4.5.7:
* Fix bug causing undesired transition from Paused to
StreamsRunning state in case of incoming UPDATE request
* Proper CoreManager destruction
* Abort push notification init on Android if gcm_defaultSenderId
resource isn't found
* Fix lime enabled device receiving an INVITE without SDP crash
* Improved how we detect if firebase is available or not
* Fixed ringing during early media for Android if native ringing
enabled
- Update to 4.5.3:
* CoreService class for Android can make the device vibrate while
incoming call is ringing.
* Audio conference API improved: distribution of participant list
over RFC4575 SUBSCRIBE/NOTIFY.
* enterForeground() and enterBackground() automatically called
(iOS and Android).
* auto acquire and release of audio focus for Android.
* Core.iterate() is being called automatically internally for
Android, it is no longer needed to create a timer in the
application to call it.
* New audio routes API to choose which device use to capture &
play audio (Android & iOS). The application can manage
* audio route changes (bluetooth, speaker, earpiece) throug
liblinphone's API.
* Added API to play user's ringtone instead of default ringtone
for Android.
* Added callback to notify a message is about to be sent.
* iOS: added linphone_core_configure_audio_session() to be called
when used with Callkit
* client-based video conference in active speaker switching mode
(beta feature).
* Warning: some function parameters have been renamed for
consistency, which modified the Swift API (where parameter
names are part of the ABI). As a result, adjustements in
applications are expected when migrating a swift app based on
liblinphone from 4.4 to 4.5.
* Improved Android network manager.
* To make it consistent with other liblinphone's object,
linphone_core_create_subscribe(),
linphone_core_create_subscribe2(),
linphone_core_create_publish() now give ownership of the
returned LinphoneEvent, which means that it is no longer need
to call linphone_event_ref() after calling these functions. As
a consequence, an application not using linphone_event_ref()
should now use linphone_event_unref() when the LinphoneEvent is
no longer used, otherwise it will create a memory leak.
* Real time text related function
linphone_chat_message_get_char() now will always return the
new line character, which wasn't the case before if the
get_char() was done after the composing callback was triggered
for this character.
* linphone_core_interpret_url() will unescape characters first if
possible if only a username is given as input parameter.
* linphone_chat_message_cancel_file_transfer() no longer deletes
the file for outgoing messages.
* magic search result created from filter now applies the
international prefix of the default proxy config if possible.
* To improve performance file transfer progress callback will be
at most notified 100 times.
* Deprecate linphone_core_audio_route_changed() that was
introduced in 4.4, to fix audio issues
* when switching audio to some low sample rate Bluetooth devices
It is now handled internally.
* Internal refactoring of management of locally played tones, in
order to fix race conditions.
* Magic search bar not looking in all friends lists.
* Error IMDN in LIME chat rooms not properly sent.
* Chat message lost during attachment auto download if Core
stopped during the process.
* Windows tests.
* Name of MediaCodec encoder and decoder filters in H264Helper
Java class.
* Both FileContent and FileTransferContent being present in
linphone_chat_message_get_contents() list until upload is
finished.
- Move grammar to liblinphone-data
* boo#1181746
- Force linking to libsoci_sqlite3
* boo#1140595 (comment 15)
* linphone-link-soci-sqlite3.patch
- Update to version 4.4.0:
* See https://gitlab.linphone.org/BC/public/liblinphone/blob/4.4.0/CHANGELOG.md
- Update to version 4.3.1:
* No changelog available.
- Update the licence: it is now GPLv3+.
- Rebase linphone-build-readline.patch,
linphone-fix-pkgconfig.patch.
- Drop linphone-fix-gtkui-build.patch.
- Remove outdated elements from the spec file.
- Add reproducible.patch to sort file list to make build reproducible
(boo#1041090)
- Change the build dependency from udev to libudev-devel.
- Change the RPM group to the more appropriate and precise
Productivity/Telephony/SIP/Clients.
- Add linphone-build-readline.patch: Add the ability to compile
with readline to the build system.
- Build with the readline support.
- Revert back to Python2: building with Python3 has regressions
(boo#1077857).
- Switch to building with Python3.
- Use linphone_gtkui conditionally everywhere
- Use build conditional for C++ bindings also in liblinphone-devel
- Add build condtionally for C++ bindings to fix build in Leap
- Remove build conditionally for ffmpeg
- Remove build conditionally for ldap
- Remove unused BuildRequires for disabled rootca download
- Provide liblinphone for the liblinphone-lang package.
- Fix the licence: there are GPL-3.0 licensed files.
- Update to version 3.12.0:
* C++ and C# bindings.
* Account Creator plugin API.
* Xamarin support.
- Switch from building with autotools to cmake.
- Add linphone-fix-pkgconfig.patch: Install linphone.pc.
- Add linphone-fix-gtkui-build.patch: Fix building of GtkUI.
- Disable the deprecated GTK+ UI for openSUSE Leap 15.x and newer,
but keep for older as the Qt inteface requires Qt 5.9+.
- Split libraries, data files and the CLI.
- Enable building C++ bindings.
- Add baselibs.conf.
- Update to version 3.11.1 (changes since 3.10.2):
* Security vulnerability fix concering TLS. The common name of
certificats delivered by SIP proxies were not compared with
their host name.
* Fix a bug in the P2P presence system. Unchecking
"Allow this contact to see my presence" check box in contact
editing view had no any effect. Then it was not possible to
hide presence informations to a contact once it had been
allowed to see it.
* Performance improvments of the presence list feature.
* IM Encription Engine: Abstraction for messaging encription
protocols.
* A bug fix in Autotools scripts.
- Ensure %icon_theme_cache_* macros are called in post/postun
scripts.
- Update to version 3.10.2 (changes since 3.9.1):
* Add lime_experimental_feature: set to 1 in .linphonerc [GtkUi]
to show the Lime menu in the graphical user interface.
Caution: Experimental.
* Video conference support through a conference server (SDK only).
* Disable dummy STUN packets sending when ICE is activated.
* Signal AVPF support as AVP: Enable rtcp feedback on RTP/AVP by
default.
* Add linphone daemon.
* gtk: Show links to files received in chat by file transfer.
* gtk: Debug window now stores "scroll to end" preference.
* gtk: Add button to take screenshot of video call.
* gtk: Fix busy presence not displayed in red.
* Fix Bad SDP when no audio codec has been enabled.
* Fix DTMF RFC2833 event always goes up in the same
LinphoneCoreListener.
* Fix linphone python version compilation - fixing vcards.
- Use ffmpeg buildconditional also in devel pkg
- Build with ffmpeg unconditionally
- Remove unused gl,glew,vpx,xv
- require a recent enough libbellesip0 package to run (bnc#957599)
- Update to 3.9.1
* Fix crash when recording video calls with the VP8 codec
* Fix H.264 codec support in Mac OS X package
* Fix translation of account assistant
* Bug fixes
- Make build more verbose
- Update to 3.9.0:
* Video recording of calls in MKV format.
* Clickable URLs in chat view.
* Add buttons to change the record and playback volumes during a
call.
* Add button to start chatting without having to create a contact
first.
* Some icon changes.
* Call logs now stored in database.
* Bug fixes.
- Use pkgconfig for ffmpeg to fix build in PMBS
- Update to 3.8.5 (changes since 3.8.0):
+ Application level improvements:
* Auto-answer ability.
* Add support of the StatusNotifierItem standard (modern
GNU/Linux status icons).
* Auto-answering can be set through the preferences panel.
* Fix crash at the end of the audio assistant.
* Fix crash when configuring a remote provisioning.
* Fix regretion in the codec view. Codec which are not usable
because bandwitch limits are to low were not grayed anymore.
* Bug fixes.
* Add translation for 'Arabic' and 'Turkish' items in the
language selection list
+ Liblinphone level improvements:
* Fix audio bug with OPUS codec.
* Fix ICE corner case not properly handled and resulting bad
final ice status.
* Update shared library ABI version to 7 (it should had been
done in 3.8.0).
* Bug fixes.
* Add a built-in XMLRPC client. Linphone does not depend on
libsoup anymore.
- Build with video support outside PMBS for openSUSE Leap 42.1
and newer (where ffmpeg is available).
- Enable ldap support.
- Regenerate configure with autotools to solve build problems in
some cases.
- Remove redefinition of GenericName in desktopfile.
- Update to 3.8.0:
+ Application level improvements:
* The video window has now controls in order to switch
fullscreen mode and terminate call.
* The out of call video preview feature (to test camera) is
moved into the settings and is no longer linked to the in-call
video preview feature.
* Add an assistant to help users to set audio/video parameters.
* Some ergonomics improvments (checkbox to set random port for
UDP and TCP, ...).
* Lots of updated translations. Arabic translation has been
added.
* Experimental feature: play an MKV file by drag-and-dropping
it on the video call window.
+ Liblinphone level improvements:
* Support for RTP/AVPF (RFC4585) for video streams, allowing fast
transmission error recovery with VP8 codec only.
* API enhancements, most objects can be ref-counted.
* Add some getter funtctions to the call information API.
* Add a function in the API to accept early-media calls.
* Add a function to set the SIP transport timeout.
* Add a function to change adaptive rate algorithm at runtime.
* Add support of file transfer.
* Call video recording feature, in mkv format (H264 streams only
for the moment).
* Call playing feature: play an MKV file and send the audio/video
stream through a call.
* Local player API. Play WAV and MKV file and display video on a
specified window display.
* A wrapper for Python has been made.
* Support of multicast IP addresses.
* Support of incoming UPDATEs within dialog (RFC3311).
* Support of SRTP by using packages from distributions.
- Spec cleanup.
- Update to 3.7.0:
Application level improvements:
* It is now possible to configure multiple proxy accounts with different
transports (UDP, TCP, TLS)
* can work with IPv6 and IPv4 simultaneously
* User can choose video rendering method on Linux
* Video HD formats support added, leveraging on multiple cores for encoding
if available
* Keyboard can be used for DTMF input
* Faster and higly responsive UI thanks to fully asynchronous operation of
the liblinphone.
* Addon of opus codec
* Possibility to specify a remote provisionning http URI for configuration
* LDAP search integration for Linux and MacOSX
* is-composing notification in chat area
Liblinphone level improvements thanks to new "belle-sip" SIP stack:
* multiple SIP transports simultaneously now allowed
* IP dual stack: can use IPv6 and IPv4 simultaneously
* fully asynchronous behavior: no more lengthly DNS or connections
* +sip.instance parameter (RFC5626)
* alias parameter (RFC5923)
* better management of network disconnections
* SIP/TLS handled through lightweighted polarssl library (instead of
openssl)
* SIP transaction state machines improved (RFC6026)
* Privacy API (RFC3323, RFC3325)
* Full support of rich presence in (RFC4480)
* Better handling of sips scheme in URIs.
* Messaging: support of is-composing (RFC3994)
* Call transfer fixes in error cases
* Add API for managing SIP SUBSCRIBES/NOTIFY/PUBLISH (linphonecore/event.h)
* bugfixes
- Version pkgconfig(libzrtpcpp) BuildRequires to resolve build for
13.2 and newer.
- Update to 3.6.1:
* UI:
* new friend list and chat messaging UI
* enhanced call history
* call and conference audio recording
* persistent chat history
* DSCP settings for SIP and RTP
* display of call statistics (when clicking on the quality indicator bar)
* Core:
* ICE for efficient RTP exchange
* fix bug in zRTP support (upgrade required)
* call recording
* uPnP
* call statistics
* adaptive bitrate control improvements
* faster call quality indicator feedback
* DSCP settings for SIP and RTP
* detailed call statistics feedback API
- Drop linphone-ffmpeg.patch and linphone-v4l.patch: they were for
shipped mediastreamer but we're using external one.
- Spec cleanup.
- fix build with automake-1.13.1
- add libsrtp and libzrtpcpp as build dependency
- use external libortp and mediastreamer2
- added video support (disabled by default), to enable need to rebuild "with video"
- modify patch linphone-v4l (ffmpeg API changed)
- added linphone-ffmpeg.patch (ffmpeg API changed)
- Use BuildRequires: pkgconfig(gnutls) instead of gnutls-devel for
Factory in keeping with gnutls package naming changes.
- update to version 3.5.2
* updated oRTP to 0.20.0
* updated mediastreamer2 to 2.8.2
* added ZRTP media encryption
* added SILK audio codec
- modify patch linphone-v4l.
- drop applet sub-package. since no useful data.
- Remove redundant/unwanted tags/section (cf. specfile guidelines)
- add libtool as buildrequire to avoid implicit dependency
- update to version 3.4.3, see NEWS file for detailed changelog,
obsoletes patches
- build / packaging improvements:
- remove build date from binaries, use the tarball's timestamp
- verbose build for rpmlint checking
- add AUTHORS, NEWS, COPYING files
- fix rpm group of devel package
- add alsa-devel and libgsm-devel buildrequires
- explicitly enable alsa, ssl, rsvp, ipv6
- fixing 11.4 build
- Add linphone-v4l-2.6.38.patch: use new v4l interface
- Added support for translation-update-upstream.
- Patch up to current git state of branch 3.3.x. Unfortunately,
the git history is not helpful to create a changelog.
But this fixes bnc#664808.
- Run spec file through spec-cleaner.
- Remove build dependencies on libgnomeprintui-devel and
libwnck-devel, they are not needed.
- own parent directory for gnome-panel
- update to linphone-3.3.2
* enhancements made to presence support (SIP/SIMPLE)
* 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)
* answer empty SIP INFO requests
* reverse order of call logs
* optimize video display
* manual or automatic network connectivity management (so that REGISTERs
are only sent when network is up or refreshed when it comes back)
* new in-call layout
* new idle view with two buttons
* ability to dial the number from dialpad
* improve local IP address detection when having multiple networks (ex: VPNs)
* use proxy's received and rport params from via in Contact header when possible
* add an option to ask linphone to place a call, whenever an instance is already running or not:
should be useful for starting calls from a web browser recognizing the 'sip:' uri scheme.
* don't show ffmpeg codecs when encoder are disabled in ffmpeg library.
* update theora default settings to match performance of 1.0 release.
* linphone can now send large video pictures: up to SVGA, configurable via the user interface
* automatic rescaling of the video windows to the video size of incoming stream
* enable lookups of SRV records
* new 'linphonecsh' program to send commands to a linphonec running as daemon using unix sockets or tcp.
- tons of bugfixes, please see NEWS file for details
- update to version 3.0.0
* new graphical interface based on Glade/Gtk
* systray icon
* STUN working for RTP
* accurate bandwidth management (to let video occupy all remaning bandwidth)
* new H264 plugin based on x264 (msx264)
* automatic call hangup upon media transmission faillure
- update patch to fix build
- Translations update
- Translations update.
- Translations update.
- get rid of autoreconf call to make it build again
- Hide online help button, we dont have an XML file to pass to
gnome_help_display so its of no use.
- update to version 2.1.1
* Clean up commands 'nat', 'stun' and 'firewall' to be
more intuitive
* Upgrade to newest gettext
- fix some rpmlint warnings
- get rid of libtool archives
- portate to new libosip version
- fix typo in linphone-1.3.0.patch
- update to version 2.0.1 which
o use new (eX)osip version
o Clean ANSI/C vs. ANSI/C++ differences
o fix a lot of small bugs
- Removed bogus dependency on mDNSResponder.
- update to versin 1.7.1:
* arts sound input/output
* improved sound latency with alsa
* alsa support works with dmix/dsnoop
* fix a lot of bugs
- disable Werror
- use libosip2-devel in buildrequires
- add missing directories
- Do not build unusable static libraries (#238552#c17).
- fix build problems
- Prefix changed to /usr.
- Spec file cleanup.
- update to version 1.3.0 which includes
o a lots of improvements in linphonec (see strk 's Changelogs below)
o telephone event problem with sipomatic.
o updated cz translation.
o fix bugs around addressbook.
o video support with H263-1998/RFC2429 nearly clean.
o added timeout for incoming calls
- Update .desktop file for UI team
- add GenericName to .desktop file
- converted neededforbuild to BuildRequires
- Update to version 1.2.0
- care about return values. Added
* linphone-returnchecks.patch
- update to version 1.1.0 drop
* linphone-manual.patch
* linphone-cs_cz.patch
* linphone-cs_cz-fixes.patch
* linphone-configure.patch
- update cz translation
- fix more prototypes
- fix unitialized vars
- fix prototypes
- fix build with current pkgconfig
- fix bonobo server file to include an icon
- fix desktop file to have both Name and GenericName
- fix return values
- Update to final 1.0.0 version, which is more stable (#74253).
- Moved applet bonobo server description to linphone-applet.
- split the gnome applet into an own package (#74199)
- fix desktop file
- added czech translations by Klara Cihlarova
- add the manual so the manual button works
- mark as NoSource (ffmpeg is there but unused).
- update to 1.0.0pre6
- fixed more compiler warnings
- fix another compile warning
- update to 1.0.0pre4
- split patch in pieces
- Fixed applet broken by bad libdir usage.
- Fixed libexecdir for bi-arch (#47050).
- update to 1.0.0pre1
- update to 0.12.2
- fix some compiler warnings
- use -fno-strict-aliasing
- update to 0.12.1
- make it build
- don't build as root
- update to 0.12.0
- install it all to /opt/gnome
- GNOME prefix change to /opt/gnome.
- Added gnome-panel to neededforbuild.
- fix desktop file path
- libtoolize in osipua subdirectory for proper soname
- created package (version 0.11.0)
- use system speex library
OBS-URL: https://build.opensuse.org/request/show/1246367
OBS-URL: https://build.opensuse.org/package/show/openSUSE:Factory/linphone?expand=0&rev=27
- Update to 5.3.26 (no changelog).
- Update to 5.3.7:
ADDED:
* Added notion of roles for conferences, so that there can be speakers and listeners.
* Reactions with IM conversations
* New video codec policy, so that hardware-accelerated codecs are prioritized in offer/answer.
* AV-1 codec
* New LinphoneAlert object used to report various QoS alerts during calls.
* SRTP AES GCM mode
CHANGED:
* TLS Client certificate request authentication callback removed (due to mbedtls update).
Application using TLS client certificate must provide it before any TLS connexion needing it.
* Refactoring of LinphoneAddress object implementation, leading to greater internal simplicity and performance.
- Clean up spec file
OBS-URL: https://build.opensuse.org/request/show/1156859
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=153
ADDED:
* Added notion of roles for conferences, so that there can be speakers and listeners.
* Reactions with IM conversations
* New video codec policy, so that hardware-accelerated codecs are prioritized in offer/answer.
* AV-1 codec
* New LinphoneAlert object used to report various QoS alerts during calls.
* SRTP AES GCM mode
CHANGED:
* TLS Client certificate request authentication callback removed (due to mbedtls update).
Application using TLS client certificate must provide it before any TLS connexion needing it.
* Refactoring of LinphoneAddress object implementation, leading to greater internal simplicity and performance.
- Clean up spec file
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=151
* Fix bug related to conference created with ICE DTLS SRTP encrypted call
* Rework tone manager:
- Remove reason field from LinphoneToneDescription
- Store tones by ID
- choose tone based on the reason when a call ends
* Do not override lime capability when enabling lime
* Do not allow to add dependency of an account to itself
* Allow to enable or disable tone indications from the core settings
* Fixed Content-Disposition (and others) headers issue in multiparts
* Update chat room flags in DB when calling insertChatRoom
* Fix an issue where on-hold music is played erroneously through a call that has been resumed
* Send invitations to the list of participants to invite and not to all invitees
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=133
* Video conferencing feature with "mosaic" and "active speaker + thumbnails" layouts
* Added conferencing scheduling API
* Support for post-quantum robust encryption for ZRTP calls
* Rtp bundle can be enabled per LinphoneAccount
* New APIs on Friend object to be able to set more info
* QRCode image generation
* Licence becomes AGPLv3
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=129
Fixes boo#1201432
- Update to version 5.1.48.
* Improved how device ringtone is obtained
* Fixed DNS servers list not updated when network changes
* Set ConferenceId also if conference event package is not enabled
* Make DTLS-SRTP to start immediately after the default candidate pair is verified by ICE.
* increase tests timeout
* Fixed audio echo canceller calibration not being done using speaker on Android
* Fixed audio focus not taken when doing echo canceller calibration if it is disabled for calls
* Fix accepting an sdp with mid but no correponding extmap attribute when it shouldn't
- Use rpmlintrc to ignore RPATH because the environment variable does not work anymore on TW.
- Remove fix that doesn’t seem to be necessary anymore.
OBS-URL: https://build.opensuse.org/request/show/989466
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=125
* Fixed incoming call ringtone being played even if device is in do not disturb mode
* Fixed call log duration not available in End state
* Fixed call duration not returning 0 if call didn't connect
* Fixed BufferedReader not closed
* Fixed adding a phone number with an invalid label to a Friend
OBS-URL: https://build.opensuse.org/package/show/network:telephony/linphone?expand=0&rev=123
