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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 describe 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 API description