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WebSockets vs Server-Sent-Events vs Long-Polling vs WebRTC vs WebTransport | RxDB - JavaScript Database
For modern real-time web applications, the ability to send events from the server to the client is indispensable. This necessity has led to the development of several methods over the years, each with its own set of advantages and drawbacks. Initially, long-polling was the only option available. It was then succeeded by WebSockets, which offered a more robust solution for bidirectional communicati
Build real-time video and audio apps on the world’s most interconnected network
Build real-time video and audio apps on the world’s most interconnected network09/27/2022 This post is also available in 简体中文, 日本語 and Español. In the last two years, there has been a rapid rise in real-time apps that help groups of people get together virtually with near-zero latency. User expectations have also increased: your users expect real-time video and audio features to work flawlessly. W
GitHub - pojntfx/weron: Overlay networks based on WebRTC.
weron provides lean, fast & secure overlay networks based on WebRTC. It enables you too ... Access nodes behind NAT: Because weron uses WebRTC to establish connections between nodes, it can easily traverse corporate firewalls and NATs using STUN, or even use a TURN server to tunnel traffic. This can be very useful to for example SSH into your homelab without forwarding any ports on your router. Se
GitHub - firefart/stunner: Stunner is a tool to test and exploit STUN, TURN and TURN over TCP servers.
Stunner is a tool to test and exploit STUN, TURN and TURN over TCP servers. TURN is a protocol mostly used in videoconferencing and audio chats (WebRTC). If you find a misconfigured server you can use this tool to open a local socks proxy that relays all traffic via the TURN protocol into the internal network behind the server. I developed this tool during a test of Cisco Expressway which resulted
WebRTC-HTTP ingestion protocol (WHIP)
Workgroup: wish Internet-Draft: draft-ietf-wish-whip-01 Published: 20 October 2021 Intended Status: Standards Track Expires: 23 April 2022 Authors: WebRTC-HTTP ingestion protocol (WHIP) Abstract While WebRTC has been very sucessful in a wide range of scenarios, its adoption in the broadcasting/streaming industry is lagging behind. Currently there is no standard protocol (like SIP or RTSP) designed
GitHub - m1k1o/neko: A self hosted virtual browser that runs in docker and uses WebRTC.
Welcome to Neko, a self-hosted virtual browser that runs in Docker and uses WebRTC technology. Neko is a powerful tool that allows you to run a fully-functional browser in a virtual environment, giving you the ability to access the internet securely and privately from anywhere. With Neko, you can browse the web, run applications, and perform other tasks just as you would on a regular browser, all
Real-Time Communications at Scale
For every successful technology, there is a moment where its time comes. Something happens, usually external, to catalyze it — shifting it from being a good idea with promise, to a reality that we can’t imagine living without. Perhaps the best recent example was what happened to the cloud as a result of the introduction of the iPhone in 2007. Smartphones created a huge addressable market for small
WebRTC is now a W3C and IETF standard | Articles | web.dev
WebRTC is now a W3C and IETF standard Stay organized with collections Save and categorize content based on your preferences. A brief overview of the history, architecture, use cases, and future of WebRTC. The process of defining a web standard is a lengthy process that ensures usefulness, consistency and compatibility across browsers. Today the W3C and IETF mark the completion of perhaps one of th
On the Road to WebRTC 1.0, Including VP8
Safari 11 was the first Safari version to support WebRTC. Since then, we have worked to continue improving WebKit’s implementation and compliance with the spec. I am excited to announce major improvements to WebRTC in Safari 12.1 on iOS 12.2 and macOS 10.14.4 betas, including VP8 video codec support, video simulcast support and Unified Plan SDP (Session Description Protocol) experimental support.
WebRTC Load Testing in 5 Minutes - Ant Media Server
WebRTC Load Testing in 5 MinutesPublished by Burak Kekeç on March 1, 2019March 1, 2019 In this post, we will tell how you can do WebRTC load testing in 5 minutes. We will run tests on AWS instances. There will be two instances: one for Ant Media Server and other for Ant Media Test Server. We have a post about how to start Ant Media Server on AWS. So we will mostly focus on Test Server Side. Create
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GitHub - GRVYDEV/S.A.T.U.R.D.A.Y: A toolbox for working with WebRTC, Audio and AI
LiveKit Docs
GitHub - livekit/livekit: End-to-end stack for WebRTC. SFU media server and SDKs.
Snapdrop
GitHub - jsmouret/grpc-over-webrtc: gRPC over WebRTC
GitHub - ant-media/Ant-Media-Server: Ant Media Server is a live streaming engine software that provides adaptive, ultra low latency streaming by using WebRTC technology with ~0.5 seconds latency. Ant Media Server is auto-scalable and it can run on-premise
GitHub - gavv/webrtc-cli: WebRTC command-line peer.
GitHub - jlaine/aiortc: WebRTC and ORTC implementation for Python using asyncio
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