Low-latency, high communication quality, security, and reliability

– Excellent quality of audio and video communication
– Fully encrypted web calls
– Extremally reliable platform for web communications

Web Call Server works with Flash RTMFP protocol and WebRTC technology. RTMFP is based on UDP protocol that is widely used in professional VoIP applications for voice and video relay.

RTMFP has a variety of essential advantages in comparison with RTMP(that is based on TCP-protocol) while working with real-time streaming media due to lower communication latency and better toleration of lost data packets in network.

As a result, you get an excellent communication quality in a good network, and acceptable quality of communication while having low-speed Internet connection and a bad link.
Web Call Server uses the newest algorithms for quality control during media relay in conditions of unstable connection(congestion control algorithms).

Besides, Web Call Server uses AES encryption method that makes the channel between browser and server fully encrypted and secured from interception of voice and video traffic.
National Security Agency of the USA resolved that AES cipher is secured enough to use it for cryptographic protection of the information as an issue of state secret and for encryption of documents with privacy mark “TOP SECRET”. Be sure, calls of your users are confidential.