Faye
Socket.io
Firebase
NATS
Pusher
deepstream.io
RabbitMQ
Apache Kafka
WebRTC
Zoom
WebEx
ClickMeeting
GoToWebinar
Onstream Media
Zoho Cliq
GoToMeeting
Faye WebRTCBased on our record, WebRTC seems to be a lot more popular than Faye. While we know about 46 links to WebRTC, we've tracked only 2 mentions of Faye. We are tracking product recommendations and mentions on various public social media platforms and blogs. They can help you identify which product is more popular and what people think of it.
Faye WebSocket is a versatile WebSocket implementation derived from the Faye project. Faye is a messaging system that uses the Publish/Subscribe model and the Bayeux protocol. It provides message servers for Node, Ruby, and all major web browsers. - Source: dev.to / 7 months ago
I can find socket.io, faye, deepstream.io , autobahn-js and nchan, centrifugo. Can you also explain why you chose it and if you had troubles with some other solution? Source: about 3 years ago
WebRTC it is: https://webrtc.org/ Yes only the network layer encryption. No file encryption as it will cost client browsers a lot in case of encrypting and then decrypting that at other end. I have written more about it here: https://dikshantraj2001.medium.com/nat-stun-turn-and-ice-466dabbc2fdb Currently, I am using the public STUN servers only. If the IPs are not reachable, it would show an error and won't work... - Source: Hacker News / 4 days ago
You might also consider assessing complementary or alternative technologies; WebSocket and HTTP aren’t the only options when it comes to real-time communication, after all. WebRTC is similar to WebSocket, with the key difference being that it’s used to implement peer-to-peer connections without relying on a server. That can be especially helpful for video calls, allowing participants to communicate directly... - Source: dev.to / 3 months ago
We use WebRTC to gain access to a user’s camera and microphone using the getUserMedia method. Typically, I would gain access to both of these from the same call. However, our experience requires the camera to flip from facing the environment to facing the user and I noticed that the small period of time the flip occurred (and microphone wasn’t available) contributed to a bit of audio lagging in the final recorded... - Source: dev.to / 6 months ago
Low latency streaming (<500ms): The Video SDK's infrastructure is built with WebbRTC, which helps to deliver secure and ultra-low latency video streams to all audiences at different bandwidths. - Source: dev.to / 8 months ago
Web Real-Time Communication (WebRTC) is a technology developed by Google in 2013 for peer-to-peer communication. WebRTC enables web browsers to capture audio, video, exchange data, and teleconferencing without plugins or intermediaries. WebRTC achieve these through APIs and protocols that interact with one another. WebRTC media streaming when used with SocKet.IO will produce an application that streams media and... - Source: dev.to / 8 months ago
Socket.io - Realtime application framework (Node.JS server)
Zoom - Equip your team with tools designed to collaborate, connect, and engage with teammates and customers, no matter where you’re located, all in one platform.
Firebase - Firebase is a cloud service designed to power real-time, collaborative applications for mobile and web.
WebEx - Cisco WebEx Web Conferencing, Online Meetings, Desktop Sharing, Video Conferencing
NATS - NATS.io is an open source messaging system for cloud native applications, IoT messaging, Edge, and microservices architectures.
ClickMeeting - Collaborate with partners and clients using ClickMeeting professional web conferencing software. Try it now, FREE!