Voice Over Internet Protocl

Voiceover Directory

Voice Over Internet Protocl


Introduction The term VoIP refers to the transfer of Voice over the Internet Protocol (IP) of the TCP/IP protocol suite. Using “VoIP” technology we can make traditional telephone calls from either computer or phone to other computer or phone using both public switched telephone network (PSTN) and internet (which is packet switched network). All you need is an Internet connection for VoIP. This technology really changes everything because it allows people to receive phone calls from anywhere that an internet connection exists, just in the same way you can receive your emails anywhere that you can connect to the internet.

The term “VoIP technology” covers a range of technologies, including voice-over-IP (VoIP) and fax-over-IP services, which are carried over both the Internet and private IP-based networks. VoIP is part of packet voice, which includes voice-over-asynchronous-transmission-mode (ATM) and frame-relay networks, which run faster than IP but are less common. VoIP connects across combinations of PCs, Web-based telephones, and phones connected via public telephone lines to remote voice gateways. Because information travels in discrete packets, it doesn’t need to rely on a continuously available switched circuit.

Using VoIP we can enhance the traditional PBX by combining voice and data services onto a single network. The end user devices (also called client device) are normally referred to as VoIP phone are used in VoIP. Development of the ‘VoIPphone’ will require the development of a ‘ system on a chip’ which combines digital signal processing (DSP) functions, micro-controller (MCU) functions, analog interface, telephone user interface and associated glue logic.

Uses of VoIP VoIP service is deployed in enterprise and service provider network for various reasons. Most of these can be categorized into following.

Ø Better bandwidth utilization by:

§ Using compression

§ Exploiting silence periods during conversations

§ Sharing of equipment for voice and data traffic (unified processing)

Ø Introduction of new services:

§ Conferences, distance learning, etc.

Working of VoIPThe basic steps involved in originating an VoIP call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is then reversed at the receiving end — switching the digital format back to analog so the telephone call goes through as normal.VoIP calls originate on any broadband line: coaxial cable, DSL (Digital Subscriber Line), wireless or even satellite. The call is routed to the VoIP Company, where a computer converts the sound into data packets – similar to the packets used to transfer internet data such as email. Sending data by packets is far more efficient as it enables the same line to handle more information simultaneously. These data packets are sent through any of the Internet’s multiple networks to a recipient of the call. The caller can receive the call via a wireless provider, a broadband provider, or a local phone carrier. In order to understand VoIP it is essential to have a complete understanding of what the difference between circuit switching and packet switching. A normal telephone uses circuit switching for phone calls, which involves routing of your call through the switch at your local carrier to the person you is calling. The connection of two points in both directions is known as circuit. Packet switching on the other hand is more efficient in transmitting data since small amount of data, which is called a packet, is sent from one system to another. In a VoIP system, once the called party answers, voice must be transmitted by converting the voice into digitized form, then segmenting the voice signal into a stream of packets. The first step in this process is converting analog voice signals to digital, using an analog-digital converter. Since digitized voice requires a large number of bits, a compression algorithm can be used to reduce the volume of data to be transmitted. Next, voice samples are inserted into data packets to be carried on the Internet. The protocol for the voice packets is typically the Real-time Transport Protocol (RTP). RTP packets have special header fields that hold data needed to correctly re-assemble the packets into a voice signal on the other end. However, voice packets will be carried as payload by UDP (User datagram protocol) protocols that are also used for ordinary data transmission.

ADVANTAGES The following are some of the advantages of VoIP:

ü VoIP is cost effective – using VoIP products long distance phone calls and international calls can be made within the price of a local call. The caller simply connects to the Internet (with the price of a call to his local Internet provider) and using the appropriate software, calls other computers running similar VoIP applications, or even other telephones anywhere in the world. (Performing a PC-2-Phone conversation requires a VoIP Gateway to be present at the remote location).

A growing amount of communication operators throughout the world utilize VoIP as the modern communication method for long distance calls, enabling a full Phone-2-Phone conversation, which is carried over an IP network. (Fully transparent to the caller) Using the IP networks for voice transportation allows for a greater deal of phone calls to be made simultaneously, thus reducing the operators’ costs. Furthermore, large companies can use their intranet as their internal enterprise phone network (iPBX). This enables lower maintenance fees, and cheap communication to remote sites and branches of the organization.

ü Convergence – By using VoIP, a company can have one comprehensive solution handling both data and telephone communication, all on the same platform and supported by a single vendor. This allows companies to use a single system for all their communication needs and prevents the overhead caused when dealing with several software packages and platforms.

ü Maintenance (Upgrades to existing services or introducing new services) can be easily done, as most applications are actually software based and do not require any hardware replacements and configuration.

ü Smart Net – Being software based, VoIP products and services enable various smart solutions. Achieving the same management capabilities on standard PSTN requires substantial hardware changes whilst most of the VoIP solutions can easily managed by a click of a mouse. An example could be the routing of a phone call to a subscriber in a predefined way: a schedule is set, and all phone calls are diverted to different locations (e.g. Home, Cellular, Business, and Voicemail) according to this schedule. Performing this capability in VoIP networks is trivial (Software solution), while achieving the same functionality on standard PSTN requires a great deal of effort.

ü New age multimedia – Because we treat voice as data and due to the fact that we can use voice services and telephony services from our PC, We can use voice applications as another application on our computer. In that way we can use the same hardware to browse the net, talk over the phone and work on other applications at the same time and without having to switch between devices. This idea is also part of the convergence advantage that was brought up here in this section.

ü Evolution towards better communication services – For all the reasons mentioned above, VoIP is an evolution towards better communication services. We can combine voice with streaming video for conference calls, allow better multimedia by using all sorts of web applications and offer customers with better communication services (such as the smart net) in order to get a communication package that will be adjustable and will be configured to supply every customer needs.

Main Issues of VoIP For VoIP to become popular, some key issues need to be resolved. Some of these issues stem from the fact that IP was designed for transporting data while some issues have arisen because the vendors are not conforming to the standards [3].

The key issues are discussed below.

Interoperability In a public network environment, products from different vendors need to operate with each other if Voice over IP is to become common among users. To achieve interoperability, standards are being devised and the most common standard for VoIP is the H.323 standard.

Security This problem exists because in the Internet, anyone can capture the packets meant for someone else. Using encryption and tunneling can provide some security. The common tunneling protocol used is Layer 2 Tunneling protocol and the common encryption mechanism used is Secure Sockets Layer (SSL).

Integration with Public Switched Telephone Network (PSTN) While Internet telephony is being introduced; it will need to work in conjunction with PSTN for a few years. We need to make the PSTN and IP telephony network appear as a single network to the users of this service.

Scalability As researchers are working to provide the same quality over IP as normal telephone calls but at a much lower cost, so there is a great potential for high growth rates in VoIP systems. VoIP systems need to be flexible enough to grow to large user market and allow a mix of private and public services.

VoIP technology can yield big cost savings to both corporations and consumers. It is more efficient than the plain old telephone service (POTS) and is poised to undergo huge growth. Before that growth can occur, however, designers have to address the issues listed above.

Along with the issues listed above, providing better voice quality to the customer is another major challenge. VoIP introduces a number of potential impairments that can impact voice quality adversely, such as the use of lossy low-bit-rate codec’s, the effects of tandem encoding/transcoding, longer delays, and packet loss. Most of these impairments are either not present or are negligible in circuit switched networks. Thus new techniques for delivering and maintaining voice quality are needed for VoIP networks. The impairments that a voice call experiences can be classified as either architectural or load dependent.

Architectural components include IP phone codec’s and their configuration parameter settings as well as fixed components of delay such as processing delays at each network element along the path and the end-to-end propagation delay. These architectural components define an upper bound on the best voice quality that could be achieved in a given network. If the upper bound is unacceptable, then changes in equipment and configurations will be required. In general, if the architecture is satisfactory, then low packet loss and delay are sufficient to ensure good voice quality.

Load dependent impairments include packet loss, queuing delay, and jitter. As load increases, these parameters deteriorate and begin to degrade voice quality. The voice quality a user experiences depends on the behavior of the entire end-to-end connection. This connection may cross multiple network domains each with its own set of controls and management methods. Since impairments across the connection are cumulative, it is possible that each network domain delivers acceptable voice quality while the end-to-end connection does not.

The networks service offering to the end applications can be measured quantitatively and qualitatively by means defining network Quality of Service. Managing voice Quality of Service across multiple domains requires SLAs (Service Level Agreements) between service providers and use of signaling protocols to indicate the desired QoS.

In the next chapter we have discussed about QoS, and requirements of QoS for voice. The minimum QoS requirements needed for better voice quality in VoIP networks.


Source by Raja Shekar

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