Literature Review on Voice over Internet Protocol free essay sample

Therefore, standards, specifications and interoperability guidelines were founded in May 1996 to standardize VoIP technology, which was a consortium of major equipment vendors and technology organizations including Cisco, Vocal Tec, 3Com, Microsoft, US Robotics and Net Speak. Nowadays, the VoIP forum is operating under the umbrella of international Multimedia Teleconferencing Consortium (IMTC). However in 2003, there was a public hearing, with the purpose of gathering information on advancements, innovations, and regulatory issues related to VoIP services which was announced by the Federal Communications Commission (FCC) as a VoIP forum. What is VoIP? Voice over Internet Protocol (VoIP) is the assembly of voice into IP data which can be transmitted over an IP network to an addressable (IP address) destination. VoIP calls are packet switched while analog calls are circuit switched. Packet switched data are data that can be routed through different routes on a network to reach a destination while circuit Switched is a connection where a physical path is dedicated between two end points [3]. We will write a custom essay sample on Literature Review on Voice over Internet Protocol or any similar topic specifically for you Do Not WasteYour Time HIRE WRITER Only 13.90 / page [5] Describe VoIP as the transport of voice over IP based data networks like WAN, MAN and LAN. In other words, it is a method for taking analog audio signals and turning them into digital data organized into packets that can be transmitted over the Internet. [9] Defines VoIP as the transmission of voice communications as datagram packets over IP networks. VoIP is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). Voice over IP is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP (internet protocol) network where it is reassembled, decompressed, and converted back into an analog wave form. The transmission of sound over a packet switched network in this manner is an order of magnitude more efficient than the transmission of sound over a circuit switched network [17]. 2. 2Basic Components of VoIP According to [9] there are variety of equipment involved in executing VoIP, in general, though, the term Voice over IP is associated with equipment that provides the ability to dial telephone numbers and communicate with parties on the other end of a connection who have either another VoIP system or a traditional analog telephone. Demand for VoIP services has resulted in a broad array of products. The basic components of VoIP include gateway, IP phone, server, gatekeepers, Call Agents, multipoint control unit e. t. c. A simple diagram of how they are being interconnected is shown below. Figure 2: Basic components of VoIP [3] †¢ Gateway: Interconnects or allows communications among devices that are not accessible within the IP network, e. g. call from or to analog phones. It converts the signals from the traditional telephony interfaces (POTS, T1/E1, and ISDN) to VoIP. Its main function includes voice packetization, compression/decompression, call routing and control signaling. †¢ IP Phone: This can be divided into two, hard phones and soft phones. A hard phone has a terminal that has native VoIP support and can connect directly to an IP network. A soft phone on the other hand runs on software application on computers. They can also be installed on mobile devices that have the same base features as VoIP phones [10]. †¢ Server: Provides management and administrative functions to support the routing of calls across the network. Gatekeepers: Is a centrally controlled entity that performs managemdr5ent functions such as authentication, address mapping and bandwidth management in a VoIP solution for multimedia application such as video conferencing. It provides two distinct and independent services. This is the process in which a name or phone number is being resolved into an IP address and, * CAC (Call Admission Control): It grants permission for a call setup attempt by determining if the network has enough bandwidth for the call. IP network: This can be a private network, an Intranet or the public network such as the Internet. †¢ IP PBX: Internet Protocol private branch exchange (IP PBX) is a telephone switching system situated within the enterprise that switches calls between VoIP users on a local line while enabling users to share some certain number of external phone lines. The gateway serves as a bridge between the H. 323 network and the outside world of (possibly) non-H. 323 devices. This includes SIP networks and traditional PSTN networks [4]. Protocols relating to H. 323 are binary protocol based on ANSI standard 1. It uses SRTP (Secure Real-Time protocol) as a standard protocol for confidential media transport and Multimedia Internet Keying (MIKEY) for exchanging keys. Session Initiation Protocol (SIP) SIP is a protocol standardized by the Internet Engineering Task Force (IETF), and was designed to support the setup of bidirectional communication sessions including, but not limited to, VoIP calls. It is similar in some ways to HTTP (Hyper Text transfer Protocol), in that it is text-based, has a request-response structure, and even uses a mechanism based on the HTTP Digest Authentication for user authentication [13]. It is an IETF specified protocol for initiating a two-way communication session. It is considered by some to be simpler than H. 323 because it is textual based; thereby avoiding the ASN. 1 associated parsing issues that exist with the H. 323 protocol suitem [4]. SIP can operate over a number of transport protocols, including Transport Control Protocol (TCP), User Datagram Protocol (UDP) and Secure Control Transport Protocol (SCTP). UDP is generally the preferred method due to simplicity and performance, although TCP has the advantage of supporting TLS protection of call setup. However, recent work on Datagram TLS (DTLS) may render this irrelevant. SCTP, on the other hand, offers several advantages over both TCP and UDP, including DoS (Denial of Service) resistance, multi-homing and mobility support, and logical connection multiplexing over a single channel. SIP is currently receiving a wide acceptance and will soon be the standard IP signaling mechanism for both multimedia and voice calling service.