Voip introduction

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Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323. Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks. Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s. VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks. Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal. Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense. The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth. In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal. To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.

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Internet Telephony, often denoted as Voice-over-Internet-Protocol (VoIP), has gained more and more attention world-wide during the last decades. Voice over Internet Protocol (VoIP) technology has become a communication alternative with the continuous increasing of Internet bandwidth and rapid advancement of peer-to-peer (P2P) applications.In this paper three types of VOIP are explained: (i) PC to PC : this is the easiest way to use VOIP, which enables you to talk and communicate Voice over Internet with all people over the world. (ii) PC to Phone: which need a gateway that connects IP Network to phone Network. Its uses a device called an ATA (Analogue Telephone Adaptor).The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VOIP. The ATA is an analogue to digital converter. (iii) Phone to Phone: Where you need more gateways that connect IP network to phone networks, more phone networks that connect Telephone set to gateway, and IP network that connect gateway to gateway. This paper also explains Internet Protocol (IP) that VOIP uses to transmit voice as packets over an IP network as follows: H.323 Protocols that provides the technical requirements for voice communication over LANs, while assuming that Quality of Service isn't provided by LANs. Session Initiation Protocol (SIP) standard which is the standard for establishing VOIP connections. This paper also explores Voice XML which is a markup language derived from XML for writing telephone-based speech applications.In this paper also VoiceXML developed environments(Gateways) are explained, where a list of all VoiceXML developers is included, and the most popular VoiceXML development environments (Gateways) are explained in details.Finally, in this paper ten VoiceXML applications are developed

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VoIP (Voice over Internet Protocol) transmits voice and multimedia content over an internet connection. VoIP allows users to make voice calls from a computer, smartphone, other mobile devices, special VoIP phones and WebRTC-enabled browsers. VoIP is a valuable technology for consumers and businesses, as it typically includes additional features that can't be found on standard phone services. These features include call recording, custom caller ID, and voicemail to e-mail. It is also helpful to organizations as a way to unify communications. The process works similarly to a regular phone, but VoIP uses an internet connection instead of a telephone company's wiring. VoIP is enabled by a group of technologies and methodologies to deliver voice communications over the internet, including enterprise local area networks or wide area networks. A VoIP service will convert a user's voice from audio signals to digital data and then send that data through the internet. If another user calls from a regular phone number, the signal is converted back to a telephone signal before reaching that user. VoIP can also route incoming and outgoing calls through existing telephone networks. However, some VoIP services may only work over a computer or VoIP phone.

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