How does Fusion Client SDK improve audio & video quality over the Internet?

Fusion Client SDK includes an impressive amount of functionality to ensure the best video & audio quality.  When incorporating video and audio into an app, communications are likely going over Wi-fi, the Internet or 4G data in network conditions that are beyond control.  Fusion Client SDK incorporates many technologies to ensure every call is of utmost quality: voice only, one-way video or two-way video.

WebRTC – state of art call quality

With WebRTC at the core, Fusion Client SDK benefits from 20+ years of VoIP technology built into the standard and embedded technologies. By leveraging WebRTC, Fusion Client SDK delivers superior audio and video quality from mobile apps and browsers into the enterprise.

Advanced codec support

WebRTC includes some of the most advanced video and audio compression technologies with minimal bandwidth footprint, packet loss concealment and variable bitrates built-in. For example, the Opus audio codec provides both narrowband and HD quality voice. Opus builds upon elements of Skype’s SILK codec to ensure unmatched performance over the Internet.

Dynamic jitter buffers

WebRTC includes a dynamic jitter buffer as well as an advanced error concealment algorithm for hiding the effects of impaired network connections exhibiting network jitter and packet loss. By buffering the minimum amount of media, this feature keeps latency low while maintaining the highest voice and video quality.  

Acoustic echo canceler (AEC)

WebRTC’s Acoustic Echo Canceler is an advanced software-based signal-processing component that removes echo resulting from the voice being played out from the active microphone.

Noise reduction

Another software-based signal processing element removes background noise associated with Voice over IP such as hiss, fan noise and more. 







Comments are disabled on these articles if you require help contact

Have more questions? Submit a request