WebRTC

WHIP (RFC 9725) and WHEP (IETF draft) define WebRTC ingest and playback over plain HTTP. How they work, who supports them, and what to build or buy.

A stage-by-stage voice AI latency budget: endpointing, STT, LLM, TTS, transport and telephony, where the extra second hides, and how to claw it back.

Looking for a Vapi or Retell alternative? Signals it’s time to leave, what owning the voice stack means layer by layer, and how to migrate safely.

We scored 8 companies that build custom live virtual classroom software on proof, media depth, and compliance. Trembit ties Fora Soft for the top score.

Build vs buy real-time video, compared: 8 video APIs, 5 open-source media servers, HIPAA, GDPR and KBV rules, and when each path wins. Checked Sep 2026.

Voice AI agent development companies compared for 2026: platforms (Vapi, Retell, LiveKit) vs integrators, telephony, compliance and named proof.

12 questions to ask a WebRTC development vendor — media-server proof, load-test evidence, TURN strategy, code ownership, team continuity.

How to set up a WebRTC data channel, choose a reliability mode, keep it in sync with video, and handle backpressure — spec-grounded, not vendor blogs.

Pipecat vs LiveKit Agents: the real architectural difference — transport-agnostic pipeline vs bundled LiveKit stack — and how to choose.

How to add live captions and real-time transcription to a video platform: streaming vs batch, vendor patterns, accuracy trade-offs, and accessibility.

“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.