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Voip (Voice Over Internet Protocol) Seminar 2004 VoIP-Voice over Internet Protocol Department of Electronics & Communication Govt Engineering College Thrissur VoIP (Voice over Internet Protocol) Submitted On Submitted by 15-10-04 Lakshmi Menon S7 ECE 630 Co-ordinator:Muneera C R Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol ACKNOWLEDGEMENT First,and foremost I thank God Almighty for making this venture a success. I extend my sincere gratitude to Prof. Indiradevi, Head of Electronics and Communication Department, Govt Engineering College, Thrissur for providing me with necessary infrastructure. I would like to convey a deep sense of gratitude to the seminar co-ordinator Mrs. C R Muneera for the timely advices. I also extend my sincere thanks to my friends for their help. Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol ABSTRACT The development of very fast, inexpensive microprocessors and special-purpose switching chips, coupled with highly reliable fibre-optic transmission systems, has made it possible to build economical, ubiquitous, high speed packet-based data networks. Similarly, the development of very fast, inexpensive digital signal processors (DSPs) has made it practical to digitize and compress voice and fax signals into data packets. The natural evolution of these two developments is to combine digitized voice and fax packets with packet data, creating integrated data-voice networks. The voice-over-Internet protocol (VoIP) technology allows voice information to pass over IP data networks. Primarily, the cost savings that accrue from operating a single, shared network have motivated this convergence of telecommunications and data communications. VoIP allows you to make telephone calls using a computer network, over a data network like the Internet. VoIP converts the voice signal from your telephone into a digital signal that travels over the Internet then converts it back at the other end so you can speak to anyone with a regular phone number. Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol INDEX PAGE S.No TOPIC NO. BASIC FLOW OF VoIP NETWORK 1 VOICE GATEWAY 2 A TYPICAL VoIP NETWORK 3 APPLICATIONS 4 IDENTIFICATION OF MAJOR SYSTEM COMPONENTS • Gateways • Gatekeepers 5 • IP Telephones • PC Software Phones VoIP PRODUCTS • Hard Phones 6 • Soft Phones VoIP QoS (Quality of Service) ISSUES • Delay 7 • Lost Packet Compensation • Echo Compensation ADVANTAGES OF USING VoIP 8 TECHNICAL BARRIERS 9 FUTURE OF VoIP TELEPHONY 10 CONCLUSION 11 REFERENCES 12 Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol BASIC FLOW OF VOIP NETWORK The VoIP networks replace the traditional public-switched telephone networks (PSTNs), as these can perform the same functions as the PSTN networks. The functions performed include signaling, databasing, call connect and disconnect, and coding- decoding. Signaling. Signaling in a VoIP network is accomplished by the exchange of datagram messages between the components. The format of these messages is covered by the standard datalink layer protocols. Database services. Database services are a way to locate an endpoint and translate the addressing that two networks use; for example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address and port numbers to identify an endpoint. A call control database contains these mappings and translations. Call connect and disconnect (bearer control). The connection of a call is made by two endpoints opening communication sessions between each other. In the PSTN,the public (or private) switch connects logical channels through the network to complete the calls. In a VoIP implementation, a multimedia stream (audio, video, or both) is transported in real time. The connection path is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and, optionally, network resources are freed. CODEC operations. Voice communication is analogue, while data networking is digital. Analogue waveforms are converted into digital information by using a coder-decoder (CODEC). Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol VOICE GATEWAY The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP network. Gateways are responsible for call origination, call detection, analogue-to-digital conversion of voice, and creation of voice packets (CODEC functions). Voice(analogue and/or digital) compression, echo cancellation, silence suppression, and statistics gathering are their optional features. The gateways must also perform some of the database services, such as phone number translations, host lookup, and signaling. The extent of gateway functionalities is based on the VoIP-enabling products used. Fig. 1 shows the architecture of a typical gateway. The DSP in a gateway is responsible for signal processing functions such as analogue- to-digital conversion of voice signals, voice compression, echo cancellation, and voice- activity detection. The functions like call origination, call detection, signaling, and phone number translations are performed by the microprocessor. Gateways exist in several forms; for example, the gateway could be a dedicated telecommunication equipment chassis, or even a generic PC running VoIP software. FIG 1 ARCHITECTURE OF A TYPICAL GATEWAY Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol A TYPICAL VOIP NETWORK Fig. 2 shows a typical VoIP network. The IP network should ensure smooth delivery of voice and signaling information to the VoIP elements. Since the IP network is to carry both voice and data, it must be able to prioritize the voice traffic. This prioritization is required for real-time VoIP applications to ensure that voice traffic is unaffected by other network traffic. Without prioritization, the voice packets may be bogged down by heavy data traffic like large file transfers using file transfer protocol (FTP).The voice packets are encapsulated with real-time protocol (RTP) and real-time control protocol (RTCP) for real-time transfer. The resource reservation protocol (RSVP) is used at the networking gateways (such as the routers) to reserve a particular amount of bandwidth for real-time applications (VoIP, video multicasting, etc). FIG 2 A TYPICAL FULL SERVICE VOIP NETWORK Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol Unlike the PCM data streams in circuitswitched telephony, VoIP data travels over the networks in packets. In VoIP digitized voice is bundled into IP packets and sent out into the network for delivery. Routers, switches, and other network equipment direct the packets to their destination IP address. This mode is called packetswitched telephony. The transport of voice packets is affected by several factors, such as the amount of bandwidth available in the network connection, the delay that the packet experiences, and any packet loss or corruption that occurs. The ability of the network to deliver the voice packets quickly and consistently is referred to as Quality of Service (QoS). Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol APPLICATIONS A wide variety of applications are enabled by the transmission of VoIP networks. The first application, shown in Figure 1, is a network configuration of an organization with many branch offices (e.g., a bank) that wants to reduce costs and combine traffic to provide voice and data access to the main office. This is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the data. Typically, this network configuration will benefit if the voice traffic is compressed. Voice over packet provides the interworking function (IWF), which is the physical implementation of the hardware and software that allows the transmission of combined voice and data over the packet network. The interfaces the IWF must support in this case are analog interfaces, which directly connect to telephones or key systems. The IWF must emulate the functions of both a private branch exchange (PBX) for the telephony terminals at the branches, as well as the functions of the telephony terminals for the PBX at the home office. FIGURE 1. BRANCH OFFICE APPLICATION Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol A second VoIP application, shown in Figure 2, is a trunking application. In this scenario, an organization wishes to send voice traffic between two locations over the packet network and replace the tie trunks used to connect the PBXs at the locations. This application usually requires the IWF to support a higher-capacity digital channel than the branch application, such as a T1/E1 interface of 1.544 or 2.048 Mbps. The IWF emulates the signaling functions of a PBX, resulting in significant savings to companies' communications costs. FIGURE 2. INTEROFFICE TRUNKING APPLICATION A third application of VoIP software is interworking with cellular networks, as shown in Figure 3. The voice data in a digital cellular network is already compressed and packetized for transmission over the air by the cellular phone. Packet networks can then transmit the compressed cellular voice packet, saving a tremendous amount of bandwidth. The IWF provides the Dept of ECE GEC,TRICHUR Seminar 2004 VoIP-Voice over Internet Protocol transcoding function required to convert the cellular voice data to the
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