WebRTC

“AI video enhancement” in a WebRTC product almost always means one of three things: noise suppression (cleaning up what a participant’s microphone picks up), background removal or replacement (blurring or swapping what’s behind them), and super-resolution (upscaling a low-bitrate or low-resolution video track). They get sold as one feature set. They are not one engineering decision. Noise suppression and basic background […]

For a two-way telehealth consultation in 2026, WebRTC is still the answer — and in healthcare the case is stronger than in generic video, not weaker. A clinical conversation needs sub-250ms interactivity, and it needs an auditable story about where patient media is decrypted and who can touch it. WebRTC’s peer-to-peer and SFU models give […]

Per-speaker live captioning is the job of turning the audio in a multi-party WebRTC call into on-screen text that is attributed to the right participant and rendered in step with the video. Architecturally it is four decisions: where you extract the audio, how you attach a speaker label to it, how you time the overlay, […]

WebRTC gets genuinely expensive in one narrow place: large one-to-many broadcast, where every viewer needs its own relayed stream, the SFU — the server that forwards media to each participant — has to be provisioned for peak concurrency, and cascading between regions bills the same bytes more than once. Below roughly 1,000 concurrent viewers per […]

How to architect HIPAA/GDPR-compliant voice on FreeSWITCH: encryption, SIP security, and on-prem “sovereign AI” for regulated voice.

Connect FreeSWITCH to WebRTC the right way: mod_verto vs SIP-over-WebSocket, ICE/STUN/TURN, DTLS-SRTP, and where browsers meet the PSTN.

How to scale FreeSWITCH past a single box: SIP clustering, Kubernetes realities, load balancing with Kamailio, HA failover, and real failure modes.

Low-latency live learning is synchronous, two-way instruction over video where the round-trip delay between teacher and student is short enough that the two can react to each other in real time — demonstrate, imitate, and correct without waiting on the network. It is a fundamentally different engineering problem from a course-video library or a recorded lecture. […]

Build a production voice AI agent on FreeSWITCH: ESL outbound mode, streaming STT→LLM→TTS, barge-in, the real latency budget, and scaling past a demo.

FreeSWITCH vs Asterisk vs Kamailio vs FreePBX: an honest comparison — architecture, scale, and the right base for voice AI.

What FreeSWITCH is, how it compares to Asterisk and Kamailio, and where real-time voice AI fits — from a team that runs production media infrastructure.

Running a voice AI platform in production means paying for five things every call touches: the WebRTC/SFU transport layer that carries the audio, speech-to-text (STT), the LLM that decides what to say, text-to-speech (TTS), and the hosting, bandwidth, and observability underneath all of it. Most public cost breakdowns only price the AI components and quietly […]

LiveKit and Mediasoup are both open-source Selective Forwarding Units (SFUs) for building real-time WebRTC video and voice, but they sit at different levels of the stack. LiveKit is a batteries-included platform: a Go SFU plus official client SDKs, room and participant management, recording/streaming (egress), an AI-agents framework, and an optional managed cloud. Mediasoup is a […]

The OpenAI Realtime API lets a browser or mobile app hold a spoken conversation with a model in near real time — audio in, audio out, over a single low-latency connection. Over WebRTC, that connection is a standard peer connection: your server mints a short-lived ephemeral token, the browser opens an RTCPeerConnection, adds the microphone track, […]

Compare the top WebRTC and video app development companies of 2026 — specialization, compliance, media-server expertise, and proof points for buyers.

Three architectures carry almost every WebRTC video product, and they differ in one thing: where the media flows. Peer-to-peer (P2P) connects participants directly, with no media server in the path. An SFU (Selective Forwarding Unit) receives each participant’s stream once and forwards it to everyone else without re-encoding it. An MCU (Multipoint Control Unit) receives every stream, mixes and transcodes them […]

WebRTC is the browser-native protocol for interactive, two-way real-time communication — calls, telehealth, conferencing — with proven sub-250ms latency. MoQ (Media over QUIC) is an emerging IETF standard built on QUIC for scalable one-to-many streaming, unifying live, time-shifted, and on-demand delivery through a publish-subscribe relay model. They solve different problems. As of 2026 they coexist […]

To migrate from Twilio Video means moving your signaling, media routing, and client SDKs off a managed CPaaS onto either another managed video API or your own WebRTC infrastructure. First, the record straight: Twilio Programmable Video is not being discontinued. Twilio announced an End-of-Life for December 2026 back in March 2024, then reversed that decision in October 2024 — […]

A real-time AI content moderation pipeline is the combination of three things: extracting frames and audio from a live media stream, running AI inference on them, and returning an enforcement action — mute, kick, flag, or blur — back to the session fast enough that harmful content never reaches viewers. The central engineering decision is where […]

A WebRTC rescue is what happens when a live video product is failing — calls drop, connections hang on “connecting,” audio goes one-way — and the team that owns the code can’t find the root cause. Usually the codebase was inherited or built by a team that has moved on, and the engineers left holding it are […]