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MicroSIP
MicroSIP is a portable SIP softphone based on the PJSIP stack available for Microsoft Windows. It facilitates high quality VoIP calls (p2p or on regular telephones) based on the open SIP protocol. MicroSIP falls into the free and open source software category and is being released under the GNU GPL-2.0-or-later. It relies on the PJSIP stack and draws on available features. This software's distinguishing characteristics are: * Profile of a lightweight background application * Small memory footprint (less than 20 MB RAM usage) * Strong adherence to the SIP standard * Support for a number of codecs: Opus, SILK, G.722, G.729, G.723.1, G.711, Speex, iLBC, GSM, AMR, AMR-WB, and video codecs H.264, H.263+, VP8. * STUN and ICE NAT traversal * SIP SIMPLE presence and messaging There are two variants, a full version with video and a "Lite" version for voice and messaging only. See also * Comparison of VoIP software * List of SIP software This list of SIP software doc ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comparison Of VoIP Software
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo. For businesses, VoIP obviates separate voice and data pipelines, channelling both types of traffic through the IP network while giving the telephony user a range of advanced abilities. Softphones are client devices for making and receiving voice and video calls over the IP network with the standard functions of most ''original'' telephones and usually allow integration with VoIP phones and USB phones instead of using a computer's microphone and speakers (or headset). Most softphone clients run on the open Session Initiation Protocol (SIP) supporting various codecs. Skype ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
List Of SIP Software
This list of SIP software documents notable software applications which use Session Initiation Protocol (SIP) as a voice over IP (VoIP) protocol. Servers Free and open-source license A SIP server, also known as a SIP proxy, manages all SIP calls within a network and takes responsibility for receiving requests from user agents for the purpose of placing and terminating calls. * Asterisk * Cipango SipServlets 1.1 application server * ejabberd * FreeSWITCH * FreePBX * GNU SIP Witch * Issabel, fork of Elastix * Kamailio, formerly OpenSER * Mobicents Platform (JSLEE 1.0 compliant and SIP Servlets 1.1 compliant application server) * Mysipswitch * OpenSIPS, fork of OpenSER * SailFin * SIP Express Router (SER) * Enterprise Communications System sipXecs * Yate Proprietary license * 3Com VCX IP telephony module: back-to-back user agent SIP PBX * 3CX Phone System, for Windows, Debian 8 GNU/Linux * Aastra 5000, 800, MX-ONE * Alcatel-Lucent 5060 IP Call server * Aric ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
C (programming Language)
C (''pronounced like the letter c'') is a General-purpose language, general-purpose computer programming language. It was created in the 1970s by Dennis Ritchie, and remains very widely used and influential. By design, C's features cleanly reflect the capabilities of the targeted CPUs. It has found lasting use in operating systems, device drivers, protocol stacks, though decreasingly for application software. C is commonly used on computer architectures that range from the largest supercomputers to the smallest microcontrollers and embedded systems. A successor to the programming language B (programming language), B, C was originally developed at Bell Labs by Ritchie between 1972 and 1973 to construct utilities running on Unix. It was applied to re-implementing the kernel of the Unix operating system. During the 1980s, C gradually gained popularity. It has become one of the measuring programming language popularity, most widely used programming languages, with C compilers avail ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
SILK
Silk is a natural protein fiber, some forms of which can be woven into textiles. The protein fiber of silk is composed mainly of fibroin and is produced by certain insect larvae to form cocoons. The best-known silk is obtained from the cocoons of the larvae of the mulberry silkworm ''Bombyx mori'' reared in captivity (sericulture). The shimmering appearance of silk is due to the triangular prism-like structure of the silk fibre, which allows silk cloth to refract incoming light at different angles, thus producing different colors. Silk is produced by several insects; but, generally, only the silk of moth caterpillars has been used for textile manufacturing. There has been some research into other types of silk, which differ at the molecular level. Silk is mainly produced by the larvae of insects undergoing complete metamorphosis, but some insects, such as webspinners and raspy crickets, produce silk throughout their lives. Silk production also occurs in hymenoptera ( bee ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
SIMPLE (instant Messaging Protocol)
SIMPLE, the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions, is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the Internet Engineering Task Force. Purpose SIMPLE applies SIP to the problems of: * registering for presence information and receiving notifications when such events occur, for example when a user logs in or comes back from lunch; * sending short messages, analogous to SMS or two-way paging; * managing a session of real-time messages between two or more participants. Implementations of the SIMPLE based protocols can be found in SIP Softphones and also in SIP Hardphones. ...[...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] |
Adaptive Multi-Rate Wideband
Adaptive Multi-Rate Wideband (AMR-WB) is a patented Wideband audio, wideband speech coding, speech audio coding standard developed based on Adaptive Multi-Rate audio codec, Adaptive Multi-Rate encoding, using a similar methodology to algebraic code-excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for Plain old telephone service, POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP. AMR-WB is codified as G.722.2, an ITU-T standard speech codec, formally known as ''Wideband coding of speech at around 16 kbit/s using Adaptive Multi-Rate Wideband (AMR-WB)''. G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector. The AMR-WB format has the following paramete ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adaptive Multi-Rate Audio Codec
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s. AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS. It uses link adaptation to select from one of eight different bit rates based on link conditions. AMR is also a file format for storing spoken audio using the AMR codec. Many modern mobile telephone handsets can store short audio recordings in the AMR format, and both free and proprietary programs exist (see Software support) to convert between this and other formats, although AMR is a speech format and is unlikely to give ideal results for other audio. The common filename extension is .amr. There also exists another stor ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
GSM 06
The Global System for Mobile Communications (GSM) is a standard developed by the European Telecommunications Standards Institute (ETSI) to describe the protocols for second-generation ( 2G) digital cellular networks used by mobile devices such as mobile phones and tablets. GSM is also a trade mark owned by the GSM Association. GSM may also refer to the Full Rate voice codec. It was first implemented in Finland in December 1991. By the mid-2010s, it became a global standard for mobile communications achieving over 90% market share, and operating in over 193 countries and territories. 2G networks developed as a replacement for first generation ( 1G) analog cellular networks. The GSM standard originally described a digital, circuit-switched network optimized for full duplex voice telephony. This expanded over time to include data communications, first by circuit-switched transport, then by packet data transport via General Packet Radio Service (GPRS), and Enhanced Data Rates for G ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ILBC
Internet Low Bitrate Codec (iLBC) is a royalty-free narrowband speech audio coding format and an open-source reference implementation (codec), developed by Global IP Solutions (GIPS) formerly Global IP Sound (acquired by Google Inc in 2011). It was formerly freeware with limitations on commercial use, but since 2011 it is available under a free software/open source ( 3-clause BSD license) license as a part of the open source WebRTC project. It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding, with the choice of data frame lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, usually the Real-time Transport Protocol (RTP). iLBC handles lost frames through graceful speech quality degradation. Lost frames often occur in connection with lost or delayed IP packets. Ordinary low-bitrate codecs exploit dependencies between spee ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
