Web Phone
Linphone is a Web phone with a Qt interface.
It lets you make two-party calls over IP networks such as the Internet.
It uses the IETF protocols SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol) to make calls, so it should be able to communicate with other SIP-based Web phones. With several codecs available, it can be used with high speed connections as well as 28k modems.
- Devel package for openSUSE:Factory
-
9
derived packages
- Links to openSUSE:Factory / linphone
- Has a link diff
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Checkout Package
osc -A https://api.opensuse.org checkout network:telephony/linphone && cd $_ - Create Badge
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Source Files
| Filename | Size | Changed |
|---|---|---|
| _link | 0000000124 124 Bytes | |
| liblinphone-5.0.36.tar.bz2 | 0020078344 19.1 MB | |
| linphone-build-jsoncpp.patch | 0000001820 1.78 KB | |
| linphone-build-readline.patch | 0000006020 5.88 KB | |
| linphone-fix-pkgconfig.patch | 0000000950 950 Bytes | |
| linphone-link-soci-sqlite3.patch | 0000001265 1.24 KB | |
| linphone-manual.tar.bz2 | 0000025033 24.4 KB | |
| linphone.changes | 0000035568 34.7 KB | |
| linphone.spec | 0000010819 10.6 KB | |
| openldap-bc.tar.bz2 | 0004533823 4.32 MB | |
| reproducible.patch | 0000001485 1.45 KB |
Revision 106 (latest revision is 155)
Giacomo Comes (gcomes.obs)
committed
(revision 106)
- 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
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