Poor call quality has a cause you can measure. A five-step VoIP and WebRTC call quality test, the thresholds to compare against, a symptom-to-cause table, and fixes for the network, QoS, app settings, and SIP-connected AI voice agents.

Noise suppression removes background sound from a microphone signal while keeping the voice. How classic and AI denoisers work, how they differ from noise reduction, echo cancellation and gain control, and how to add noise suppression in the browser, in mobile apps and with VideoSDK.

QoE, or quality of experience, is what the participant took away from a call, while QoS is what the network delivered. The ITU definition, a QoE vs QoS table, the metrics that matter for real-time video, and three ways to measure QoE in production.

Video encoding compresses raw frames into a stream a network can carry, and transcoding converts one encoded stream into another. The main codecs and containers, hardware vs software encoders, and how real-time video calls encode differently, with VideoSDK's encoder presets.

SRTP encrypts every WebRTC call and most secure VoIP calls, but encrypted in transit is not end-to-end. How SRTP protects RTP packets, how DTLS-SRTP, SDES and ZRTP exchange keys, which ciphers and ports it uses, and what a media server can still see.

Learn how to build high-quality real-time video apps with Flutter WebRTC. Get started today and deliver the best user experience with your video apps using Flutter WebRTC.

WebRTC signaling is the process of exchanging information between peers to establish and manage real-time communication sessions, covering details like session descriptions and network configurations.

WebRTC latency is the end-to-end delay from capture to playback in real-time sessions. Learn the causes, measurement methods, and engineering fixes that bring mouth-to-ear delay below 300 milliseconds.

WebRTC video bitrate refers to the amount of data transmitted per second during real-time communication, influencing video quality and bandwidth usage.

This blog believes in a well-defined explanation of the set of theories involved in WebRTC. It is not a tutorial and does not contain many codes. As mentioned above, the series makes full efforts for the readers to make them avail the best knowledge.

WebRTC vs WebSocket, compared on ten dimensions: transport, topology, encryption, head-of-line blocking and cost. Includes a tested echo in each API, when to use each, and how WebSocket signalling starts a WebRTC call.

Build a WebRTC React Native video calling app with step-by-step signaling, ICE, and TURN setup, plus a VideoSDK shortcut for production-ready calls in under a day.