What is acceptable network latency?

Typically, anything at 100ms is acceptable for gaming. However, the 20ms to 40ms range is considered optimal. So simply put, low latency is good for online gamers while high latency can present obstacles.

How does WebRTC measure performance?

Two methods to measure WebRTC performance WebRTC performance can be measured in two ways: on live user sessions or on predictable synthetic traffic. Measuring live user sessions enables you to understand the perceived quality your users are currently experiencing.

What is the latency of WebRTC?

With sub–500 milliseconds of real-time latency, WebRTC is the fastest protocol on the market. WebRTC was built with bidirectional, real-time communication in mind.

How much bandwidth does WebRTC use?

Current WebRTC implementations use Opus and VP8 codecs: The Opus codec is used for audio and supports constant and variable bitrate encoding and requires 6–510 Kbit/s of bandwidth.

What is considered slow latency?

Low latency is not the same as low speed. Latency is measured in milliseconds, and indicates the quality of your connection within your network. Anything at 100ms or less is considered acceptable for gaming. However, 20-40ms is optimal.

How do I find my WebRTC stats?

You can collect statistics at various levels throughout the WebRTC hierarchy of objects. Most broadly, you can call getStats() on an RTCPeerConnection to get statistics for the connection overall.

How do I check my WebRTC bandwidth?

You can see the current traffic generated by webrtc by going to chrome://webrtc-internals and inspecting all the charts.

Does WebRTC scale?

Despite a popular myth, WebRTC does scale. Since WebRTC creates a peer-to-peer connection, some wrongly assume it is unwieldy to scale even into the hundreds. Breaking through this conventional thinking, Red5 Pro reimagined the entire architecture.

Why is WebRTC so complicated?

WebRTC is an over-engineered Rube Goldberg machine for many reasons, including the fact that it tries to include so many capabilities in a single standard. It’s also complex because STUN, TURN, and ICE are overly complex. WebRTC should have used a simpler underlying design. You really, really don’t need all that.

Is 0 jitter possible?

Jitter is the irregular time delay in the sending of data packets over a network. Acceptable jitter means what we are willing to accept as the irregular fluctuations in data transfers. Jitter should be below 30 ms. Packet loss shouldn’t be more than 1\%.