|
Phoner
Phoner and PhonerLite are softphone applications for Windows operating systems available as freeware. Phoner is a multiprotocol telephony application supporting telephony via CAPI, TAPI and VoIP, while PhonerLite provides a specialized and optimized user interface for VoIP only. Beside the different user interface focus both programs share the same code base. Both programs use the Session Initiation Protocol for VoIP call signalisation. Calls are supported via server-based infrastructure or direct IP to IP. Media streams are transmitted via the Real-time Transport Protocol which may be encrypted with the Secure Real-time Transport Protocol (SRTP) and the ZRTP security protocols. Phoner provides as well an interface for configuring and using all supplementary ISDN services provided via CAPI and thus needs an ISDN terminal adapter hardware installed in the computer. Both programs support IPv4 and IPv6 connections by using UDP, TCP and TLS. Supported audio formats * G.711 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Phoner
Phoner and PhonerLite are softphone applications for Windows operating systems available as freeware. Phoner is a multiprotocol telephony application supporting telephony via CAPI, TAPI and VoIP, while PhonerLite provides a specialized and optimized user interface for VoIP only. Beside the different user interface focus both programs share the same code base. Both programs use the Session Initiation Protocol for VoIP call signalisation. Calls are supported via server-based infrastructure or direct IP to IP. Media streams are transmitted via the Real-time Transport Protocol which may be encrypted with the Secure Real-time Transport Protocol (SRTP) and the ZRTP security protocols. Phoner provides as well an interface for configuring and using all supplementary ISDN services provided via CAPI and thus needs an ISDN terminal adapter hardware installed in the computer. Both programs support IPv4 and IPv6 connections by using UDP, TCP and TLS. Supported audio formats * G.711 ... [...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] |
Opus (codec)
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, network ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ZRTP
ZRTP (composed of Z and Real-time Transport Protocol) is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over IP (VoIP) phone telephony call based on the Real-time Transport Protocol. It uses Diffie–Hellman key exchange and the Secure Real-time Transport Protocol (SRTP) for encryption. ZRTP was developed by Phil Zimmermann, with help from Bryce Wilcox-O'Hearn, Colin Plumb, Jon Callas and Alan Johnston and was submitted to the Internet Engineering Task Force (IETF) by Zimmermann, Callas and Johnston on March 5, 2006 and published on April 11, 2011 as . Overview ZRTP ("Z" is a reference to its inventor, Zimmermann; "RTP" stands for Real-time Transport Protocol) is described in the Internet Draft as a ''"key agreement protocol which performs Diffie–Hellman key exchange during call setup in-band in the Real-time Transport Protocol (RTP) media stream which has been established using some other signaling protocol such as S ... [...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] |
Pulse-code Modulation
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speex
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on VoIP applications and podcasts. It is based on the CELP speech coding algorithm.Xiph.OrIntroduction to CELP Coding Retrieved 2009-09-01 Speex claims to be free of any patent restrictions and is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/ RTP. It may also be used with the FLV container format. The Speex designers see their project as complementary to the Vorbis general-purpose audio compression project. Speex is a lossy format, ''i.e.'' quality is permanently degraded to reduce file size. The Speex project was created on February 13, 2002. The first development versions of Speex were released under LGPL license, but as of version 1.0 beta 1, Speex is released under Xiph's version of the (revised) BSD license. Speex 1.0 was announced on March ... [...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] |
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] |
Transport Layer Security
Transport Layer Security (TLS) is a cryptographic protocol designed to provide communications security over a computer network. The protocol is widely used in applications such as email, instant messaging, and voice over IP, but its use in securing HTTPS remains the most publicly visible. The TLS protocol aims primarily to provide security, including privacy (confidentiality), integrity, and authenticity through the use of cryptography, such as the use of certificates, between two or more communicating computer applications. It runs in the presentation layer and is itself composed of two layers: the TLS record and the TLS handshake protocols. The closely related Datagram Transport Layer Security (DTLS) is a communications protocol providing security to datagram-based applications. In technical writing you often you will see references to (D)TLS when it applies to both versions. TLS is a proposed Internet Engineering Task Force (IETF) standard, first defined in 1999, and the c ... [...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] |
User Datagram Protocol
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in packets) to other hosts on an Internet Protocol (IP) network. Within an IP network, UDP does not require prior communication to set up communication channels or data paths. UDP uses a simple connectionless communication model with a minimum of protocol mechanisms. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no handshaking dialogues, and thus exposes the user's program to any unreliability of the underlying network; there is no guarantee of delivery, ordering, or duplicate protection. If error-correction facilities are needed at the network interface level, an application may instead use Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) which are designed for this ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
