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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.
Peer-to-peer communication is a decentralized network model where each participant, or 'peer', has equal privileges and can initiate or complete transactions without the need for a central server. This architecture enhances scalability and resilience, but can also pose challenges in terms of security and data consistency.
JavaScript APIs are interfaces that allow developers to interact with and manipulate web browser features or services, enabling dynamic and complex web functionalities. They provide a standardized way for scripts to interact with a variety of web technologies, ranging from the Document Object Model (DOM) to more advanced features like WebGL and service workers.
Real-Time Protocols are essential for enabling the transmission of data with minimal delay, which is crucial for applications such as video conferencing, online gaming, and live streaming. These protocols prioritize timely delivery over reliability, often using techniques like packet prioritization and network congestion management to maintain low latency communication.
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.
Interactive Connectivity Establishment (ICE) is a framework used to allow web browsers and other clients to connect with peers, such as in WebRTC, by finding the best path through NATs and firewalls. It enhances real-time communication by dynamically selecting the most efficient network route using a combination of STUN and TURN servers.
Network Address Translation (NAT) traversal is a technique used to establish and maintain Internet protocol connections across gateways that implement NAT, which often blocks incoming traffic by default. It is essential for enabling peer-to-peer communication and direct connectivity in applications such as VoIP, online gaming, and peer-to-peer file sharing.
Secure Real-time Transport Protocol (SRTP) is an extension of the Real-time Transport Protocol (RTP) that provides encryption, message authentication, and integrity, ensuring secure transmission of audio and video data over the network. It is widely used in applications like VoIP and video conferencing to protect against eavesdropping and tampering during data transmission.
Data channels are pathways that facilitate the transfer of information between systems or components, ensuring efficient communication and data integrity. They are crucial in network architecture, enabling seamless data exchange and integration across diverse platforms and technologies.
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.
Streaming protocols are essential for the real-time delivery of multimedia content over the internet, ensuring smooth playback by managing data transmission between client and server. They handle challenges like latency, buffering, and varying network conditions to provide a seamless user experience.
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