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WebRTC Gateway
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins. Usage scenario To enable browsers using different application providers to communicate with each other (e.g. a user logged into application providers X wants to call someone that is logged into application provider Y) a so-called WebRTC trapezoid can be used. In this case the two providers use a widely used VoIP signalling protocol such as SIP to federate between them. However, each of their respective browser-based clients signals to its server using proprietary application protocols built on top of HTTP and WebSocket. This component that mediates between WebRTC and SIP is referred to as a WebRTC Gateway. Beside connecting different WebRTC a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
WebRTC
WebRTC (Web Real-Time Communication) is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. Supported by Apple, Google, Microsoft, Mozilla, and Opera, WebRTC specifications have been published by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). According to the webrtc.org website, the purpose of the project is to "enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols". History In May 2010, Google bought Global IP Solutions or GIPS, a VoIP and videoconferencing software company that had developed many components required for RTC, such as codec ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the Transport Layer of the TCP/IP suite. SSL/TLS often runs on top of TCP. TCP is connection-oriented, and a connection between client and server is established before data can be sent. The server must be listening (passive open) for connection requests from clients before a connection is established. Three-way handshake (active open), retransmission, and error detection adds to reliability but lengthens latency. Applica ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Kamailio
Kamailio, formerly OpenSER (and sharing some common history with SIP Express Router (SER)), is a SIP server licensed under the GPL-2.0-or-later license. It can be configured to act as a SIP registrar, proxy or redirect server, and features presence support, RADIUS/syslog accounting and authorization, XML-RPC and JSON-RPC-based remote control, SQL and NoSQL backends, IMS/VoLTE extensions and others. Kamailio is a Hawaiian word. Kama'ilio means talk, to converse. "It was chosen for its special flavour." Features Kamailio is written in pure C with architecture-specific optimizations; it can be configured for many scenarios including small-office use, enterprise PBX replacements and carrier services—it is SIP signaling server—a proxy—aiming to be used for large real-time communication services. Features include: * SIP telephony system * SIP load balancer * SIP security firewall * Least cost routing engine * IMS/VoLTE platform * Instant messaging and presence services * S ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Session Border Controller
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks. Early deployments of SBCs were focused on the borders between two service provider networks in a peering environment. This role has now expanded to include significant deployments between a service provider's access network and a backbone network to provide service to residential and/or enterprise customers. The term "session" refers to a communication between two or more parties – in the context of telephony, this would be a call. Each call consists of one or more call signaling message exchanges that control the call, and one or more call media streams which carry the call's audio, video, or other data along with information of call statistics and quality. Together, these streams make up a session. It is the job of a session border controller to exert influence over the data flows of sessions. The term "border" refers to a point of demarcation b ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
WebRTC2SIPt
WebRTC (Web Real-Time Communication) is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. Supported by Apple, Google, Microsoft, Mozilla, and Opera, WebRTC specifications have been published by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). According to the webrtc.org website, the purpose of the project is to "enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols". History In May 2010, Google bought Global IP Solutions or GIPS, a VoIP and videoconferencing software company that had developed many components required for RTC, such as cod ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Interactive Connectivity Establishment
Interactive Connectivity Establishment (ICE) is a technique used in computer networking to find ways for two computers to talk to each other as directly as possible in peer-to-peer networking. This is most commonly used for interactive media such as Voice over Internet Protocol (VoIP), peer-to-peer communications, video, and instant messaging. In such applications, communicating through a central server would be slow and expensive, but direct communication between client applications on the Internet is very tricky due to network address translators (NATs), firewalls, and other network barriers. ICE is developed by the Internet Engineering Task Force MMUSIC working group and is published as RFC 8445, as of August 2018, and has obsolesced both RFC 5245 and RFC 4091.RFC 4091, ''The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework'', G. Camarillo, J. Rosenberg (June 2005) Overview Network address translation (NAT) becam ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
STUN
STUN (Session Traversal Utilities for NAT; originally Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators) is a standardized set of methods, including a network protocol, for traversal of network address translator (NAT) gateways in applications of real-time voice, video, messaging, and other interactive communications. STUN is a tool used by other protocols, such as Interactive Connectivity Establishment (ICE), the Session Initiation Protocol (SIP), and WebRTC. It provides a tool for hosts to discover the presence of a network address translator, and to discover the mapped, usually public, Internet Protocol (IP) address and port number that the NAT has allocated for the application's User Datagram Protocol (UDP) flows to remote hosts. The protocol requires assistance from a third-party network server (STUN server) located on the opposing (public) side of the NAT, usually the public Internet. STUN was first announced in RFC 3489; the title was ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Codec
A codec is a device or computer program that encodes or decodes a data stream or signal. ''Codec'' is a portmanteau of coder/decoder. In electronic communications, an endec is a device that acts as both an encoder and a decoder on a signal or data stream, and hence is a type of codec. ''Endec'' is a portmanteau of encoder/decoder. A coder or encoder encodes a data stream or a signal for transmission or storage, possibly in encrypted form, and the decoder function reverses the encoding for playback or editing. Codecs are used in videoconferencing, streaming media, and video editing applications. History In the mid-20th century, a codec was a device that coded analog signals into digital form using pulse-code modulation (PCM). Later, the name was also applied to software for converting between digital signal formats, including companding functions. Examples An audio codec converts analog audio signals into digital signals for transmission or encodes them for storage. A receiv ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Codec
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains bo ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Opus (audio Format)
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC. Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, networke ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
