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
- Browser interoperability management : Tracking WebRTC API’s evolution since 2012
- NAT traversal, QoS, media
- optimization, media prioritization
- STUN, TURN, ICE
- SFU (P2P & SFU routing mode)
- Calls forking, Voice call continuity
- Statistics (Chat, number of calls, QoS)
- Cloud Architecture
- Media server load balancing
- DB Cluster
Connectors to different ecosystems :
- SIPoWs, SIP…
- HTTPS, SRTP, authentication
- CRMs, SMS invitations etc.
- « Service Management Server »
- Configuration and preferences
- Statistics and logs
- REST API
- « 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
- Enable server side interconnection : SFU, SIP Gw, Chat bot
Architecture : deployment
ApiRTC CPass is available on different platform :
- APAC (Singapore)
HDS / HIPAA compliant instance is also available
Deployment can be also done on private or
Advantages of ApiRTC
- Without installation
Prices and features of ApiRTC
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