The Linphone application is most likely the very first open source software in the world to provide post quantum encryption to secure voice and video communications. For that purpose, the Linphone team developed a modified version of ZRTP, the well-known encryption protocol. This module is an extension to the bctoolbox library providing Post Quantum Cryptography using the Crystals-KYBER algorithm.
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Products
Linphone
is an open source instant messaging and voice/video over IP (VoIP) phone that makes it possible to communicate freely with people over the internet via voice, video and text messaging.
Since it follows open standards from the telecommunications industry (SIP, RTP), Linphone is interoperable with most PBXs and SIP servers, including our own Flexisip server, and can be used with any SIP VoIP operator.
Discover complementary products in our complete solution for instant messaging and IP-to-IP calls, such as Linphone applications, Liblinphone cross-platform VoIP SDK and Flexisip server.
Many new features are still under development on a medium- or long-term basis. Please feel free to contact us if you want to know more about our development roadmap, or if you are interested in financing the development of a particular feature.
products list
Linphone for smartphones and tablets
Linphone is one of the most famous open source softphones in the world. It is fully SIP-based, for all calling, presence and IM features.
Use Linphone for free, and enjoy its intuitive interface and advanced features with our free SIP service or with an existing SIP account.
Easy-to-use graphical interface :
- account creation assistant
- smart contact list, with address book synchronisation for smartphones
- call history
- "Invite your friends" option
- remote provisioning from a QR Code or a URL
Advanced calling features :
- HD video calls with video preview (H.264, H.265 and VP8 codecs supported)
- audio/video conference calls and scheduled meetings
- call transfer and multi-call management (pause and resume)
- call recording and replay (audio only)
Increased audio and video quality :
- state-of-the-art adaptive jitter buffer algorithm
- innovative adaptive bitrate control
- HD voice (Opus support) and echo cancellation
- call quality indicator and display of advanced call statistics
Enhanced Instant Messaging experience :
- one-to-one and multi-participant conversations
- intuitive message delivery status (IMDN)
- file sharing (photo, video, document, voice/video message)
- multi-device support
- advanced IM notifications, with "reply to" and "mark as read" options
Secure communications :
- secure user authentication
- end-to-end encryption of audio/video calls, messages and shared files
- post quantum encryption
- ephemeral messaging
Available for:
Android
iOS
Linphone for desktop platforms
Linphone for desktop platforms is perfectly suited for both standard users with no specific knowledge of VoIP/SIP and for advanced audio/video calling use cases in a professional environment.
It is fully SIP-based, for all calling, presence and IM features.
Use Linphone Desktop for free, and enjoy its modern look and feel and its many advanced features with our free SIP service or with an existing SIP account.
Complete, intuitive User Interface :
- simplified user experience with swift access to major features
- intelligent search bar for ease of communicating
- contact list with presence status
- unified call and chat history
- quick access to your recent events
- multi-account and multi-device support
Enhanced Instant Messaging Experience :
- instant messaging with file sharing and groups
- access to chat views during calls
- intuitive message delivery status (IMDN)
Advanced calling features :
- HD video calls in full screen mode (H.264, H.265 and VP8 codecs supported)
- optimised audio/video quality, thanks to our new jitter buffer and bitrate adaptation algorithms
- HD voice (Opus support) and echo cancellation
- audio/video conference calls with up to eight participants, with three display modes (active speaker, mosaic, audio only)
- multiple call management (pause and resume)
- call transfer, auto-answer, DTMF support
- call quality indicator and display of advanced call statistics
- secure communications with SIP/TLS, SRTP and ZRTP and post quantum encryption
Available for:
GNU / Linux
macOS
Windows Desktop
Linphone web integration
Linphone Desktop is now able to interact with web browsers and can be integrated with your Web Services.
Since we are experiencing increasing demand for advanced solutions for professionals, we have made Linphone for Desktop capable of receiving information from a web browser to start a call, open the app, invite participants to a conference call etc.
Available for:
Chrome
Edge
Firefox
Safari
Liblinphone SDK
Liblinphone is a high-level open source library that integrates all the SIP-based voice/video calling, conferencing and instant messaging features into a single easy-to-use API.
This is the VoIP SDK engine on which Linphone applications are based!
Liblinphone combines our media processing and streaming toolkit (Mediastreamer2) with our user-agent library for SIP signaling (belle-sip).
Signaling features :
- user authentication via digest (SHA256 or md5) and TLS client certificates
- multiple call management: hold, resume, transfer and more
- multiple SIP proxy support: registrar, proxies, outbound proxies
- presence using list subscription to presence server
- DTMF (telephone tones)
- interoperability with IMS networks
- suitable for building a VoLTE-compatible client
Media features :
- HD audio calls with support for Opus codec
- HD video calls with support for H.264, H.265 and VP8 codecs
- state-of-the-art adaptive jitter buffer algorithm
- innovative adaptive bitrate control
- ICE support for optimised NAT traversal
- end-to-end media encryption using SRTP, SRTP-DTLS and ZRTP protocols
- post quantum encryption using Crystals-Kyber algorithm
- audio and video call recording
Instant Messaging features :
- one-to-one and group instant messaging
- ephemeral messaging
- voice/video messaging
- delivery and "is typing" notifications
- end-to-end encryption for one-to-one and group conversations
- file sharing, with configurable settings to automatically / manually download attachments
Consult our technical corner to see the list of supported standards, download pre-compiled binaries or the SDK source code, and find more information about available tutorials and documentation.
Flexisip Server
Flexisip is a complete and scalable SIP server suite that includes several modules: a proxy, a presence server, a conference server, a back-to-back user agent, and Flexisip account manager for the creation and administration of SIP accounts.
Flexisip is an easy-to-install SIP server solution, offering all the features required to deploy your own SIP service tuned for mobile or desktop applications, « out of the box ».
The free sip.linphone.org SIP service has run on Flexisip since 2011, and enables Linphone users to create their own SIP addresses to connect with each other.
Proxy :
- NAT-aware with built-in media relay module and ICE/STUN support
- interconnected with Apple's and Google's push notification systems
- high availability and cluster mode for large deployments
- high-level event logging for activity monitoring
- digest and TLS client certificate authentication
Presence :
Flexisip manages SIP SUBSCRIBE and PUBLISH from client applications, in order to :
- identify users of the service within the address book
- provide users with real-time presence status
Conference (for group chat and group calls):
- entirely SIP-based solution
- efficient management of groups and participants
- multi-device support
- notification of group events (users joining or leaving, new admins etc.)
B2BUA:
- media encryption transcoding
- media codec transcoding
- SIP bridge, for interconnection with PSTN gateways
Flexisip account manager:
- SIP account creation and management, by end-users or admins
- remote provisioning of Linphone-based apps (including app's settings, user account, contact list...)
- web admin platform
Flexisip can be integrated into your SIP infrastructure for various purposes, for example IN THE CLOUD in order to run your own VoIP service, EMBEDDED in small hardware systems or in its PUSH GATEWAY mode of operation.
Other components
Tunnel: VoIP anti-blocking solution
Our tunnelling solution has been developed for VoIP operators who wish to bypass VoIP blocking in some countries or illegitimate firewall restrictions on 3G networks.
All SIP and RTP traffic is tunnelled through a single secure https connection to a detunneliser server.
Linphone Instant Messaging Encryption
Linphone Instant Messaging Encryption library (LIME) provides end-to-end encryption for instant messaging, allowing users to exchange messages privately and asynchronously. It supports multiple devices per user and multiple users per device.
LIME offers two major security benefits to Instant Messaging users:
- End-to-end encryption, which means that only you and your contact (and not even the server) can decrypt the content that you shared
- Perfect forward secrecy, which ensures that encrypted messages cannot by decrypted by a third party, if a key is compromised in the future
Mediastreamer2
Mediastreamer2 is a powerful lightweight streaming engine for voice/video telephony applications.
This open source library is responsible for receiving and sending all multimedia streams in Linphone, including voice/video capture, encoding and decoding, and rendering.
Bcg729
Bcg729 is an open source implementation of both encoder and decoder of the ITU G729 Annex A/B speech codec.
The library is fully portable and can be executed on many platforms including both ARM and x86 processors. libbcg729 supports concurrent channels encoding/decoding for multi call application such as conferencing.
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