Add one-to-one and group text communications capabilities to your web application.
Set up browser-based one to one audio calls or web conferences.
Set up browser-based, one to one video calls, video broadcast (1:n) or video conferences (n:n).
Manage user presence in real-time. Add presence indicators to your app.
Use WebRTC to transfer files over the data channel.
Share the actions you’re performing on the web-browser in real-time. Co-browse while having a text, audio or video conversation.
Share your computer screen with your visitor to show them your special offers, new products…
Draw and share annotations in real-time with one or multiple parties.
Trigger snapshots on local and remote party sides and use a secure secured peer-to-peer datachannel to transfer files.
Add real-time interaction at the right moment of the customer journey.
Get detailed statistical reports regarding API usage.
ApiRTC Architecture :
ApiRTC platform :
Main components of ApiRTC platform are :
- CCS : Call Control Server
- STUN/TURN servers
- Media servers : SFU
- VoIP GW
- Connectors to different ecosystems : SIPoWs, SIP, IMS …
ApiRTC features :
- Interoperatibility: ApiRTC manages browser interoperability
- Presence: presence group registration and subscription
- Media management: NAT traversal (STUN/TURN/ICE), QoS, media optimization
- Monitoring and statistics
- Scalability: SFU Cluster
- Connections : Connectors to different ecosystems
- Secure: HTTPS, SRTP, authentication, private cloud or on-premise deployment options