Since you are exchanging media in real-time in a video application, you depend on the network to a greater extent than you might in other kinds of apps. In Figure 1 below, you can see an official table of bandwidth requirements from our Twilio-powered group conversations for audio and video.
| Latency (RTT) | < 200ms |
|---|---|
| Jitter | < 30ms |
| Packet Loss | < 3% |
| Bandwidth (Uplink/Downlink) per participant | 32-100kbps (audio only) |
150kbps (video only) |
Figure 1. Table of bandwidth requirements from Twilio
The required bandwidth is always related to the available video quality of the bubble participants. You should always consider a higher video resolution.
| Bubble type | Minimal required bandwidth |
|---|---|
| 9 participants video room with HD VP8 Video (1280x720) + HD Audio | ~6 Mbps |
| 9 participants video room with SD VP8 Video (640x480) + HD Audio | ~4 Mbps |
| 9 participants video room with Lo-res VP8 Video (240x180) + HD Audio | ~2 Mbps |
Figure 2. Bandwidth video requirements estimate assuming a user is receiving all the participants’ audio and video (except their own) at 30fps. The bandwidth requirement will vary depending on the codec, resolution, and frame rate.
Thanks to WebRTC, the video call will adapt to network conditions, so you just need to make sure you reach the lo-res minimum video quality. If the user is unable to meet the minimum requirements for video, they will need to switch to audio-only mode. In case the video quality drops we show a network quality indicator in the top right corner of a participant's video.
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