ApiRTC : Add realtime interactions to your website and mobile apps

ApiRTC is a library that allows a developer to easily integrate real-time communications (chat, audio, video) into a web or mobile application.

• Mobile and web services developers need simplified API to integrate real time communication into their services.
• WebRTC Platform as a Service (PaaS) and client side library streamline integration: Hiding interoperability issues from browsers or mobile apps, Managing network traversal and QoS issues and Managing media bridges, IMS interworking etc.

ApiRTC Key Features

Inter-operability:

• Browser interoperability management : Tracking WebRTC API’s evolution since 2012

Media management:

• NAT traversal, QoS, media
• optimization, media prioritization
• STUN, TURN, ICE
• SFU (P2P & SFU routing mode)
• Calls forking, Voice call continuity

Monitoring:

• Statistics (Chat, number of calls, QoS)

Scalability:

• Cloud Architecture
• Media server load balancing
• DB Cluster

Connectors to different ecosystems :

• SIPoWs, SIP…

Security:

• HTTPS, SRTP, authentication

Integration:

• CRMs, SMS invitations etc.

ApiRTC architecture

SMS :

• « Service Management Server »
• Configuration and preferences
• management
• Statistics and logs
• REST API

CCS :

• « Call Control Server » Call signaling, call establishment,
• Users’ presence status management

Media Proxy :

• Media relay
• Implements STUN and TURN protocols

SFU : « Selective Forwarding Unit »

• Manages conferencing streams

Connectors :

• Enable server side interconnection : SFU, SIP Gw, Chat bot

Architecture : deployment

ApiRTC CPass is available on different platform :

• France
• US
• APAC (Singapore)

HDS / HIPAA compliant instance is also available

Deployment can be also done on private or
on-premise