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ApiRTC is a library which allows a developer to easily integrate real-time communications (chat, audio, video) into a web or mobile application.
Behind the scenes, the WebRTC technology is used : this means that the media flows are encrypted and exchanged in peer-to-peer when possible.
Otherwise a media relay (TURN server) may be required.
ApiRTC is the entry point of our platform that will manage all connection establishments for you including user connection, user presence and media establishment.
ApiRTC library is available in different format depending of the integration you need :
- Native Application : Swift for iOS, Java for Android
ApiRTC enables you to exchange different media content type :
- Datachannel (files ...)
- JSON data
And manage important features such as :
- Media optimization
- Firewall traversal
- Network and QoS testing
- SIP interconnection
- Call recording and replay
ApiRTC call anatomy
Let's say that we have a mobile user that wants to call a desktop user (Client 1 calling Client 2). The signaling offer of the Client 1 goes through the signaling server (CCS), and is relayed to the desktop client who can choose to accept or refuse the call
Now both clients are aware of the capabilities of each other (audio/video codecs supported for instance). In a typical scenario they will at this point also be able to “talk” to each other: the signaling server is essentially useless now. They are now able to exchange their media (audio/video) flux between each other.
As we see with this simple example, the signalisation phase is absolutely crucial to establish calls. There are also many factors not presented in this simple example (differences between browsers, media constraints, fallback if there are firewalls…), but this simple scenario is helpful to understand the following tutorials.
These tutorials describes some of the possibilities of ApiRTC, the goal of our tutorials is to help you in the ApiRTC integration.
For more details, you can also check our complete
UserAgent is the starting point of apiRTC and enables you to manage several important aspects such as :
- User registration
- Media devices management
- Offline features ...
More details on our UserAgent description and UserAgent API
UserAgent-1 : Registration : external user management
Manage your users in your database
UserAgent-2 : Registration with integrated users management
Users are managed on Apizee Cloud
Conversation will enable you to manage multi-party communication : conference, whiteboard, groupChat ...
These tutorials describes some of the possibilities related to Conversation API
Create a conversation to establish multi-party conference : audio, video
Conversation-2 : Conference with media devices selection
Create a conversation to establish multi-party conference : audio, video and select your media devices
Conversation-3 : Conference with QoS Monitoring and active speaker detection
Learn how to add QoS Monitoring and active speaker detection on your conference
Conversation-4 : Conference with media muting and screensharing
Learn how to mute your local stream and share your screen in a conference
Conversation-5 : Group Chat
Exchange chat messages between all users in a Conversation
Conversation-6 : Whiteboard
Learn how to start a multi-party Whiteboard on a conversation
Conversation-7 : Whiteboard with invitation
Learn how to start a Whiteboard and invite users
Conversation-8 : Advanced streams Pub/Sub
Learn how to select streams and media type on publish and subscribe
Conversation-9 : Join / Leave
Learn how to join / leave a conference
Conversation-10 : Moderation
Learn how to accept / refuse guest users
Conversation-11 : Recording
Create a conversation recording
Conversation-12 : Conference & Bluetooth (BLE) device
Learn how to use Bluetooth (BLE) device in conference context
Conversation-13 : Conference, chat, file transfer
Learn how to run video conference with chat and file transfer
Conversation-14 : Conference & video file publishing
Learn how to publish a video file as the conversation stream
Conversation-15 : Background substraction
Activate background substraction on conference : blur, transparent or image
Conversation-16 : Media control
Adds the usage of our Stream helpers to control zoom, torch, takePhoto, etc...
Conversation-17 : External RTMP Streaming
Learn how to publish a stream to an external RTMP server
Learn how to establish One-to-One communication on ApiRTC.
One-to-One-1 : Calls
Establish audio, video or screenSharing calls between two users
One-to-One-2 : Chat
Exchange instant messaging between two users
One-to-One-3 : send File
Exchange any type of file between two users
One-to-One-4 : Accept - Refuse calls
Establish audio, video or screenSharing calls between two users with accept/refuse possibility
One-to-One-5 : Call & Send file
Presents how to send file during P2P call
One-to-One-6 : Call, Snapshot & DataChannel
Presents how to exchange snapshot data using DataChannel
Stream Management tutorials
Stream is used to manage MediaStream.
These tutorials describes some of the possibilities related to Stream API
Stream-1 : Select Media
This demo teaches you how to select a media device to start a stream
Stream-2 : Screensharing
Learn how to use screen display for communication
Stream-3 : Recording
This demo teaches you how to locally record a stream
Stream-4 : Snapshot
This demo teaches you how to create a snapshot image from a video stream
Stream-5 : Video resolution
Learn how to start a stream with a selected video resolution and then use it for a call
Hybrid mobile application tutorials
Learn how to integrate ApiRTC in your Hybrid mobile application.
Mobile - Cordova
Create an hybrid application using Cordova framework on Android and iOS
Mobile - Ionic framework
Check our github repository for an integration sample with Ionic
Discover APIs to improve your apiRTC experience.
Tools - Precall Test
Allow your visitors to test their Bandwidth, Devices, and Calltest.