Blitzz Help Center

WebRTC - What it is, and how it relates to video call quality.

What is WebRTC

WebRTC (Real-Time Communications) is part of the technology Blitzz uses to connect two or more people on a video call. It acts as a middleman that packages, and compresses data from a video session (e.g., audio created from a conversation, and video captured from the webcam). An encoder compresses this data into Packets. These get sent across your wifi network to whoever you're on talking to. The data stream is condensed so it can be relayed across the network to your participant instantaneously. We refer to everyone as "Endpoints," and each sends their data stream to the other while the connection is live.

How it relates to Calling quality

There are a few reasons why a video session can be choppy, sound bad, or drop altogether. The Data packets being sent to and from each device can get lost in transit. 

 Packet Loss is often caused by

  • High Latency (the time it takes to transfer data packets between endpoints)

  • Congested/inadequate bandwidth (the transfer rate at which data gets sent through your wifi network)

  • Spotty connection (device too far away from wifi source/physical or electrical interference)

Only one participant has to have these problems for both endpoints to be affected. Each endpoint should have a stable connection to wifi, and enough open bandwidth to transfer/receive data simultaneously.  

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