Based on the SIP telecommunication protocol
Flexisip easily integrates into your SIP infrastructure to meet various needs.
With the Flexisip server suite, create your own cloud service based on the SIP telecommunications protocol, add instant messaging, video conferencing, and push notifications to your service, or even integrate an embedded server into your connected hardware.
Access all the necessary features to deploy your own unified communication solution, optimized for mobile environments.
To deploy a unified communication service with calling, chat, and video conferencing features.
Flexisip modules provide advanced audio/video calling, instant messaging with file sharing, and video meetings.
These services can be deployed on machines of your choice, on-site or with a hosting provider. With high availability mechanisms and load balancing, Flexisip is particularly well-suited for deploying services targeting a large number of users.
To enhance your existing VoIP platform with unified communication features.
The Flexisip server suite can be deployed alongside an existing SIP service, such as Asterisk, FreeSWITCH, or Mitel.
Thanks to support for recent standards – including numerous SIP protocol extensions – Flexisip can adapt to the constraints of your current systems.
Gradually migrate your users to a fully software-based solution and add new features without losing the benefits of the platforms you have already deployed.
To add internet audio and video calling features to your embedded products.
Thanks to its modular architecture and minimal dependencies, Flexisip runs efficiently on small embedded systems, such as Raspberry Pi.
Flexisip supports IoT-specific features, like Early Media Call Forking, which allows you to see the caller’s video before accepting the call.
Many clients, particularly in intercom and access control systems, use Flexisip for these applications.
To reliably notify your users of incoming calls and messages on smartphones.
On mobile platforms, push notification systems are presented as efficient and highly reliable methods for receiving information when the phone’s screen is off or when the VoIP or messaging application is running in the background.
The Push Gateway mode is a specific operating mode of Flexisip, developed to robustly route incoming SIP calls or text messages to Apple and Google smartphones and tablets. It can be integrated with existing SIP infrastructures that do not support push notifications.
The Flexisip server suite enables account management, conferencing, and telephony integration.
The Flexisip suite consists of a set of server software based on the SIP protocol and written in modern C++ language. It includes a SIP account manager, a proxy, a presence server, a conference server for group calls and multi-party text conversations, as well as a B2BUA module for gatewaying to the PSTN network and connecting to traditional telephony.
The source code is available under two projects: Flexisip (proxy, presence server, conference server, B2BUA) and Flexisip Account Manager (the SIP account manager).
All traffic passes through the proxy module, which serves as the entry and exit point for SIP communications. It is responsible for the following tasks:
Push notifications allow VoIP calls or incoming messages to be delivered to mobile platforms when the app is not active in the foreground or when the phone screen is turned off. Flexisip proxy supports RFC 8599 and the latest requirements for push notification services from Apple (APNs) and Google (Firebase).
For organizations that have already selected a third-party SIP server that does not support push notifications, Flexisip can be deployed in a “Push Gateway” mode. This enables the addition of push notification support to these SIP infrastructures while allowing the existing server to manage the SIP traffic.
Flexisip B2BUA enables the connection of a SIP network with a PSTN gateway or a SIP trunking service provided by a VoIP operator.
Note: We do not provide PSTN lines or virtual numbers.
This software offers several key components for creating a secure user account management system. It is also the recommended tool for deploying the MySQL account database used by the Flexisip proxy.
Account Management API (FlexiAPI)
FlexiAPI is a REST API for managing user accounts:
For example, it allows adding SIP account management features to an existing administration platform.
Remote Configuration Server
This component generates configuration files compatible with Linphone clients. The end user only needs to scan a QR code or click a link received by email for the settings defined in the configuration file to be automatically applied to the application.
You can choose between static configuration (the same settings for all users) or dynamic configuration (a personalized configuration file per user).
Web Platform for SIP Service Administration
This online interface allows you to manage your user base and your service based on the Flexisip server:
Our online SIP service allows Linphone users to communicate with each other for free.
Not convinced by Flexisip yet? Create test accounts and try out the features you can benefit from on your own unified communication service in Linphone.
Deploy only the modules that manage the features you need.
Opt for a cluster deployment with high availability and load balancing.
Flexisip is developed and maintained by our teams in Grenoble. Choose a sovereign solution!
100% of the source code is open source! Rely on transparency and reliability.
Choose between open-source or proprietary licensing.
Linphone is available under a dual licence. Our proprietary licences allow you to modify the source code while keeping your modifications closed-source.
Flexisip can be integrated into your project under the terms of the GNU Affero GPLv3 open-source license.
Our Flexisip experts will be able to answer your questions about the software and help you define the server infrastructure best suited to your needs.
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