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Session Description Protocol (SDP) is a format for describing the parameters of multimedia communication sessions, enabling the announcement and negotiation of media streams over networks. It is primarily used in conjunction with protocols like SIP and RTSP to establish, modify, and terminate multimedia sessions such as voice calls and video conferences.
Session Initiation Protocol (SIP) is a signaling protocol used to initiate, maintain, modify, and terminate real-time communication sessions involving video, voice, messaging, and other communications applications. It operates predominantly in the application layer of the Internet Protocol Suite and is essential for facilitating the setup and management of multimedia communication sessions over IP networks.
The Real-Time Streaming Protocol (RTSP) is a network control protocol designed to control streaming media servers and establish and manage media sessions between endpoints. It is primarily used for establishing and controlling media sessions for applications like video conferencing, live streaming, and surveillance systems, enabling real-time data delivery over IP networks.
Session management is a crucial aspect of web security and user experience, ensuring that user interactions are tracked and managed securely across multiple requests. It involves maintaining the state of a user's session, typically through the use of session IDs, to authenticate and authorize user actions while preventing unauthorized access.
Network protocols are standardized rules that govern how data is transmitted and received across networks, ensuring reliable and secure communication between different devices and systems. They are essential for interoperability, enabling diverse devices and applications to communicate seamlessly within and across networks.
Media encoding is the process of converting media files from one format to another, allowing for efficient storage, transmission, and playback across different platforms and devices. It involves compressing data to reduce file size while maintaining quality, using codecs that balance efficiency and compatibility.
Transport protocols are essential for managing the transmission of data across networks, ensuring that data is sent, received, and processed efficiently and accurately. They provide mechanisms for error detection, flow control, and congestion management to maintain the integrity and performance of data communication.
Web Real-Time Communication (WebRTC) is an open-source project that enables peer-to-peer audio, video, and data sharing directly between web browsers without needing plugins. It facilitates Real-Time Communication by using JavaScript APIs, making it essential for applications like video conferencing, file sharing, and live streaming.
The Real-Time Protocol (RTP) is a network protocol designed for delivering audio and video over IP networks, ensuring real-time data transmission with minimal latency. It is widely used in streaming media systems, video conferencing, and telephony applications, working in conjunction with the Real-Time Control Protocol (RTCP) to monitor data delivery and provide quality feedback.
Signaling servers facilitate the initial connection setup between peers in a peer-to-peer communication network by exchanging control information needed to establish a direct peer connection. They do not handle the actual media or data transfer, which occurs directly between peers after the connection is established.
Concept
WebRTC (Web Real-Time Communication) is an open-source project that enables web applications and websites to capture, and potentially broadcast, audio and/or video media, as well as to exchange any type of data between browsers without requiring an intermediary. It facilitates peer-to-peer communication, making it a powerful tool for applications like video conferencing, file sharing, and live streaming, all directly within the web browser.
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