They transport audio streams using special media delivery protocols that encode

They transport audio streams using special media

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transmission occurs as IP packets over a packet-switched network. They transport audio streams using special media delivery protocols that encode audio and video with audio codecs, and video codecs. 2.2 Protocols Used Voice over IP has been implemented in various ways using both proprietary protocols and protocols based on open standards. VoIP protocols include: Session Initiation Protocol (SIP) H.323 Media Gateway Control Protocol (MGCP) Gateway Control Protocol (Megaco, H.248) Real-time Transport Protocol (RTP) Real-time Transport Control Protocol (RTCP) Secure Real-time Transport Protocol (SRTP) Session Description Protocol (SDP) Inter-Asterisk eXchange (IAX) Jingle XMPP VoIP extensions Skype protocol
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2.3 Quality of Service Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of Service (QoS) guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attacks than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically. Therefore, VoIP implementations may face problems with latency, packet loss, and jitter. 2.4 Operational Cost VoIP has drastically reduced the cost of communication by sharing network infrastructure between data and voice. A single broad-band connection has the ability to transmit more than one telephone call. Secure calls using standardized protocols, such as Secure Real-time Transport Protocol, as most of the facilities of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream. 2.5 Working First, voice is converted from an analog signal to a digital signal. It is then sent over the Internet in data packets to a location that will be close to the destination. Then it will be converted back to an analog signal for the remaining distance over a traditional circuit switched (PSTN) (unless it is VoIP to VoIP). Your call can be received by traditional telephones worldwide, as well as other VoIP users. VoIP to VoIP calls can travel entirely over the Internet. Since your voice is changed to digital (so that it can travel over the Internet), other great features such as voice messages to email, call forwarding, logs of incoming and outgoing calls, caller ID, etc., can be included in your basic calling plan all for one low price.
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3. SKYPE: WORKING AND ARCHITECTURE 3.1 Skype Network Skype is one of the largest VoIP providers with over 500 million users. Skype was the first peer-to-peer IP telephony network created by the developers of KaZaa.
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