Metered blog
In this article we are going to learn about HLS and RTMP and which one on these protocols is best for streaming Here is what you are going to learn in this article 1. Understanding Streaming Protocols 1. What are streaming protocols? 2. How do protocols affect content delivery? 2.
Network performance metrics are important in maintaining network infrastructure Network performance metrics provide information regarding the health and security of a network and highlight areas that require optimization Here are some of the Key Network Performance metrics: 1. Bandwidth 2. Latency 3. Throughput 4. Packet Loss 5. Jitter 6. Error
Python is a versatile and accessible programming language that is known for its clear syntax and readability This makes it a good choice for building webrtc applications We can build a WebRTC server in python by using libraries such as aiortc aortic library * Pure python Implementation: * The aiortc library is
In this article we are going to create a video calling app with Janus WebRTC server and SFU We are also going to learn about Metered.ca SFU and why it is a better option than Janus Server What is Janus? Janus is an Open source, general purpose WebRTC SFU
WHIP (WebRTC-HTTP Ingestion Protocol) and WHEP (WebRTC-HTTP Egress Protocol) are protocols that are designed to streamline signalling in WebRTC with the help of standard HTTP methods * Definition of WHIP: WHIP simplifies how client devices send media streams to the server. * It replaces the complex signalling mechanism that is required with
WebRTC and HLS are both technologies that helps you broadcast live sports to video conferencing Choosing the right streaming protocol is important because it directly impacts quality, latency and reliability of the content that is being streamed to audiences WebRTC (Web Real-Time Communications) WebRTC is an technology that enables real
Simple Peer is a JavaScript Library for implementing peer-to-peer communication This library is built on top of webrtc, at its basic implementation the Simple Peer provides a straightforward API for setting up P2P communication. It has the following features * Signalling * Connection management * Media Streams * Data Channels When and why you
STUN is a protocol that is designed to overcome barriers in communication that are introduced by NAT. STUN protocol enables devices that are behind a NAT to discover their own public IP address and port number. The devices communicate with a STUN server that is on the internet, and the
Transmission Control Protocol TCP and User Datagram Protocol UDP are two core internet protocols. These protocols govern how data is transmitted across the internet TCP: TCP is a connection-oriented protocol that establishes a reliable communication channel between two devices. TCP sends the data packets securely and in correct sequence, it
In this article we are going to learn about transcoding and encoding. Encoding is the process of converting raw media data into digital format that is easy to use, store, process and transmitted and played back. Transcoding is the process of converting already encoded files into a different format. This
Bandwidth and throughput are fundamental concepts in networking, these are often interchangeable used but do have different meanings Bandwidth Bandwidth is the maximum rate at which data can be transmitted through a network connection. This is measured in Mbps or Mega bits per seconds. It can be likened to a
In this article we are going to learn about the WebRTC data channels. Here is what we are going to learn * Understanding WebRTC Data Channels * What is WebRTC? * WebRTC Data Channels Explained * Functionality and features of WebRTC Data Channels * Use-cases of WebRTC Data Channels * Advantages and Benefits of WebRTC Data
In this guide we are going to learn about how to get started with Metered Global TURN servers. Here is what you are going to learn in this guide: 1. Dashboard Overview 2. Selecting TURN Server region 3. Creating your first credential * Adding the Label * Reading the Instructions * Adding the
In this article we are going to learn about end to end encryption in video. The end to end encryption is a process where data that is being transferred from one user to another across a communication channel remains confidential and tamper proof from outside, this also includes the providers
Jitter in networking is the time delay between packets that are being sent over a network connection Whenever data packets are transmitter from a source to a destination, these data packets ideally should arrive in regular intervals of time However in the real world, there are many factors that influence
What is WebRTC SFU (Selective Forwarding Unit)? Selective Forwarding Unit or SFU SFU is an server based architecture that is used in webrtc to conduct multi party video calling, audio calling and data transfer. The SFU acts an intermediary between devices that recives media streams from multiple devices and then
WebRTC or Web Real Time Communication protocol is an open source protocol and technology that enables real time communication directly between web browsers and webRTC enabled applications Using WebRTC you can do video calling, audio calling and data transfer between devices. This capability is implemented using a set of JavaScript
The Secure Real Time Transport Protocol is a cutting edge technology. The SRT protocol is designed to enable the secure and efficient data transmission over the internet The SRT protocol was designed by Haivision, it is an Open source protocol made primarily for the real time video and data delivery
Video compression is the process of reducing the file size of a video without significantly reducing its quality. It reduces the size of the data that is required for video files. The video compression tries to reduce the file size without compromising on reduction in quality of the video Video
Simulcasting, is a short form for the simultaneous broadcasting. This is a process of broadcasting the same media content through multiple distribution channels simultaneously The concept includes streaming the same content across various digital platforms including social media websites, homepage, apps and media sites such as youtube, facebook, linkedIn, Vimeo
In this article we are going to learn about how to implement WebRTC in WebView for IOS, Android and Flutter Implementing WebRTC in IOS WebView Let us create a simple app with WebRTC enables and in a step by step process to better understand how you can implement webrtc in
There are different WebRTC architecture available today, these are SFU, MCU and P2P, selecting one depends on many factors these include Network Conditions Bandwidth availability * High Bandwidth If participants have good quality bandwidth then SFU and peer to peer calling will work for them * Variable Bandwidth If the participants do
Latency in networking is the time it takes for a packet of data to travel from its source to destination. This time is typically measured in milliseconds as the data travels at the speed of light, but variety of factors influence the time required to travel These factors include the
WebRTC and Websockets are both real time technologies, these technologies enable instantaneous exchange of data. Both the technologies are important for applications that require live interactions Common use-cases for these technologies include online gaming, live chats, live streaming and other low latency applications WebRTC (Web Real Time Communications) Webrtc allows
In this article we are going to learn about RTSP RTSP (Real Time Streaming Protocol) is a network control protocol designed to use in communication systems. RTSP protocol provides a framework for real time streaming of audio and video RTSP provides efficient management of audio and video streams with extreamly