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CSipSimple
CSipSimple is a Voice over Internet Protocol (VoIP) application for Google Android operating system using the Session Initiation Protocol (SIP). It is open source and free software released under the GPL-3.0-or-later license. The project was abandoned in October 2017. As of 26 May 2019, CSIP no longer has an active website and is no longer available on the Play Store. Users with CSip already installed did not have the app removed from their device. Details It relies on the PJSIP SIP stack and get features provided by this SIP stack. The key features of this software are: * Multi-codec support: Speex (narrow-band/wide-band), G.711 ( u-law/a-law), GSM, iLBC, G.729 (support dropped with r2180, need to buy a licensed g729 plug-in), G.722, AMR (narrow-band), iSAC, SILK (narrow-band/wide-band/ultra wide-band) (support dropped in 2014) * A plug-in adds support for Codec2, G.726, G.722.1 and Opus * A plug-in adds video calling with VP8, H264 and H263-1998 codecs * Multi-accoun ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Packet Loss Concealment
Packet loss concealment (PLC) is a technique to mask the effects of packet loss in voice over IP (VoIP) communications. When the voice signal is sent as VoIP packets on an IP network, the packets may (and likely will) travel different routes. A packet therefore might arrive very late, might be corrupted, or simply might not arrive at all. One example case of the last situation could be, when a packet is rejected by a server which has a full buffer and cannot accept any more data. Other cases include network congestion resulting in significant delay. In a VoIP connection, error-control techniques such as automatic repeat request (ARQ) are not feasible and the receiver should be able to cope with packet loss. Packet loss concealment is the inclusion in a design of methodologies for accounting for and compensating for the loss of voice packets. PLC techniques * Zero insertion: the lost speech frames are replaced with silence. * Waveform substitution: the missing gap is reconstructed ... [...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] |
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] |
Internet Speech Audio Codec
internet Speech Audio Codec (iSAC) is a wideband Speech communication, speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, e.g. Real-time Transport Protocol, RTP. It is one of the codecs used by AOL Instant Messenger, AIM Triton, the Gizmo5, Tencent QQ, QQ, and Google Talk. It was formerly a proprietary software, proprietary codec licensed by Global IP Solutions. As of June 2011, it is part of open-source software, open source WebRTC project, which includes a royalty-free license for iSAC when using the WebRTC codebase. Parameters and features * Sampling frequency of 16 kHz (wideban ...[...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] |
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] |
Secure Real-time Transport Protocol
The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast applications. It was developed by a small team of Internet Protocol and cryptographic experts from Cisco and Ericsson. It was first published by the IETF in March 2004 as . Since RTP is accompanied by the RTP Control Protocol (RTCP) which is used to control an RTP session, SRTP has a sister protocol, called Secure RTCP (SRTCP); it securely provides the same functions to SRTP as the ones provided by RTCP to RTP. Utilization of SRTP or SRTCP is optional in RTP or RTCP applications; but even if SRTP or SRTCP are used, all provided features (such as encryption and authentication) are optional and can be separately enabled or disabled. The only exception is the message authentication feature which is indispensable and required when using SRTC ... [...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] |
Traversal Using Relays Around NAT
Traversal Using Relays around NAT (TURN) is a protocol that assists in traversal of network address translators (NAT) or firewalls for multimedia applications. It may be used with the Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). It is most useful for clients on networks masqueraded by symmetric NAT devices. TURN does not aid in running servers on well known ports in the private network through a NAT; it supports the connection of a user behind a NAT to only a single peer, as in telephony, for example. TURN is specified by . The TURN URI scheme is documented in {{IETF RFC, 7065. Introduction NATs, while providing benefits, also come with drawbacks. The most troublesome of those drawbacks is the fact that they break many existing IP applications, and make it difficult to deploy new ones. Guidelines have been developed that describe how to build "NAT friendly" protocols, but many protocols simply cannot be constructed according to those guidelines. Exampl ... [...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] |
H263-1998
H.263 is a video compression standard originally designed as a low-bit-rate compressed format for videotelephony. It was standardized by the ITU-T Video Coding Experts Group (VCEG) in a project ending in 1995/1996. It is a member of the H.26x family of video coding standards in the domain of the ITU-T. Like the previous H.26x standards, H.263 is a block-based hybrid video coding scheme using 16×16 macroblocks of YCbCr color sample arrays, motion-compensated prediction, an 8×8 discrete cosine transform for prediction differences, zig-zag scanning of transform coefficients, scalar quantization, run-length transform coefficient symbols, and variable-length coding (basically like Huffman coding but with structured coding tables). The first (1995) version of H.263 included some optional features including overlapped block motion compensation and variable block-size motion compensation, and the spec was later extended to add various additional enhanced features in 1998 and 2000. S ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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