Networked And Distributed Systems Computer Science Essay

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Abstract

This document is dealing with the VoIP (voice over internet protocol) operation. In the beginning it is discussed about the basic of VoIP (mobile and PSTN) service. It is discussed the working process with PSTN operators .It is also discussed the working process with mobile operators. Later it is discussed in details some technical issues of VoIP along with some important factors with illustration .In the end it is discussed about some VoIP service provider related to Mobile VoIP.

Introduction

Now a day's internet has greatly impacted the daily lives of people worldwide. It came to us with limitless possibilities, driving a lot of new ideas to make our life even more convenient. VoIP (voice over internet protocol) is one of the latest technological outcomes of internet, created a new era of transmitting voice communication using internet connection for making cheap local and international phone calls throughout the world instead of traditional phone networks. VoIP is complex network because it works with the voice and data networks altogether. Tradition voice network is circuit switched and data network is packet switched. It has become popular to consumers because of its some outstanding features like user friendly, cheap, convenient and seamless. Using VoIP can be done in just a few clicks on your computers or dials on a special VoIP phone. VoIP is also very easy to set up. Some VoIP company provide free services to make a call worldwide or some take a little amount charges for making a long distance calls. It is convenient because we can manage our VoIP account in any part of the world with stable internet connection. VoIP converts the voice signal from our telephone into a digital signal which can travel over the Internet.VoIP mainly does the transmission of voice traffic over the IP-based network. The internet protocol (IP) is mainly designed for the data networking. The internet protocol is now using very efficiently to voice networking for making a call because especially traditional phone system is very costly to make international phone call.

When VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).Calls from a VoIP device to a PSTN device are commonly called "PC-to-Phone" calls, even though the VoIP device may not be a PC. Calls from a VoIP device to another VoIP device are commonly called "PC-to-PC" calls, even though neither device may be a PC. Yahoo and Msn messengers can be an example of this type of VoIP service. Apart from this, there are some special types of software designed for this type of VoIP services. Now Calling card is frequently used for making international calls worldwide. There are some VoIP devices like gatekeeper, gateway used in the technology. Source and destination, both the parties uses the gateway to transmit and receive the calls. If VoIP service provider assigns a regular telephone number, then we can receive calls from regular telephones that don't need special equipment, and most likely we'll be able to dial just as we always have.

With the increasing demand of VoIP service, VoIP service providers to look into a new way using your mobile phone to route the calls over the internet. Mobile VoIP service has become a new challenge for providing seamless and cheap calls across the world. Technology has already established enough with the fixed phone operators. The first is easier, i.e. soft phone application that can be installed on mobile phone networks and data used to place and receive calls. Examples of the most important Yeigo, Fring, PeerMe and Truphone.  Skype, Fring and jajah are some common mobile VoIP services providers throughout the world.

Background

VoIP (voice over internet protocol), is a kind of software that you can use over the Internet to talk to people for a small fee (sometimes free of charge) of charge. VoIP phones use packet switching properties of the Internet which allows frequent calls to occupy the space used by a single call in circuit-switched network. It also uses the concept of data compression, which further reduces the size of the call.

The real conversation in VoIP has happened in some steps. To enable a smooth conversation all the steps are combined together. If we are using a traditional phone, a VoIP hardware called as a Analogue Telephone Adapter (ATA) allows you to connect your standard phone to broadband modem. The ATA converts the analogue signal from the standard phone to digital data for transmission. The telephone call will start with one of the parties (source and destination) when picking up the phone .This transmits a signal to the ATA. In return the ATA sends a dial tone in response to the signal. This is to ensure the Internet connection. 

To select the desired phone number, analogue sound is converted into digital data by the ATA device and temporarily stored. The data related to phone number is send to the VoIP service providers call processor to check the number format for a valid conversation. The next step is mapping the phone number in which the number is converted into an IP address. The devices on both the ends of the call are connected by the soft switch, and the party who has been called receives a signal on their ATA instructing it to ask the connected number to ring.

A session is established between your computer and the called party's system, once the other person answers the call. Both systems expect data from each other and they must use the same protocol to communicate. The packets of data are translated by the ATAs on each end into analogue audio signals, which both parties finally get to hear. Disconnecting the call will close the circuit between the VoIP phones and the ATAs. A signal is then sent to the soft switch by the ATA terminating the session. 

Packet switching technology in VoIP enables telephones with the ability to communicate the way computers do.

Mobile VoIP 

Mobile VoIP is an expansion of mobility to a Voice over IP network.

Several methods can be integrated with cell phone into a VoIP network. One method is turning the mobile device into a standard SIP client, and then uses a data network for sending and receiving SIP messages and to send and receive RTP for the voice path. This method requires mobile handset supports is turned into a standard SIP client along with high speed IP communication. In this application, standard VoIP protocols (typically SIP) are used over any broadband IP-capable wireless network connection such as EVDO rev A (which is symmetrical high speed - both high speed up and down), HSDPA, Wi-Fi or WiMAX

.

As i mentioned earlier, soft phone application that can be installed on mobile phone networks and data used to place and receive calls. Soft switch act as a gateway to bridge SIP and RTP into the mobile network's SS7infrastructure .In this method a sip application sever control the whole process and provide sip based service as well as mobile handset continues to operate as it has always (for example GSM or CDMA based device.

Mobile VoIP will be a compromise between economy and mobility. For example, the voice over service on Wi-Fi is free but only available within the area covered by Wi-Fi Access Point. High speed services from mobile operators using EVDO rev A or HSDPA, sound and better capabilities for citywide coverage including fast handoffs among mobile base stations yet, it will be more cost of service than typical Wi-Fi-based VoIP. 

Mobile VoIP will be an important service in the coming years as manufacturers uses the devices that has more powerful processors and cheaper memory to meet user needs for ever- more 'power in their pocket'. In Mid-2006 Smartphone can send and receive e-mail, browse the internet (although it is at low rates) and in some cases, allowing the user watching TV. 

It becomes a new challenge for the mobile operators to maintain the network services as well as to introduce new advantages and idea of IP for the user. Users like high speed internet to access some specific sites freely. Such a service challenges the most valuable service in the telecommunications industry - voice - and threatens to change the nature of the global communications industry.

VoIP Signalling Protocols

The International Telecommunications Union and the Internet Engineering Task Force Protocols for Governing VoIP

H.323, the International Telecommunications Union (ITU) standard for establishing VoIP connections

SIP (Session Initiation Protocol), the Internet Engineering Task Force (IETF) standard for establishing VoIP connections

Media Gateway Control Protocol, the first protocol developed by the IETF to signal control information between VoIP network components.

H.248; code-name Mega co, the protocol both the IETF and the ITU use to signal control information VoIP network elements

H.323

In H.323 networks gatekeeper is main responsible for all call authorization, bandwidth management and call signaling. Gateway also has separate call control and management function. A gateway connects the internet to the telephone networks. A gateway is five layer devices that can translate a message from one protocol to another. Here gateway transforms a telephone network message to an internet message. Call processing servers store information about network topology for routing calls to VoIP gateways and end user devices. Both the terminal should be register with the gatekeeper. Here the gatekeeper server plays the role of the registrar server.

SIP (Session Initiation Protocol)

SIP is the basis for the new IP Multimedia Subsystem (IMS) protocol; a joint development between the IETF and the Third Generation Partnership Project (3GPP). It is a application layer protocol that establishes, manages and terminates a session (call).it can be used to create two party, multi party session.

SIP client-server application supports user mobility with 2 modes

Proxy mode, SIP clients sends its signaling requests to the proxy server. The proxy server either handles the request or forwards it to other SIP servers.

Redirect mode, SIP clients send its signaling requests to the redirect server. The SIP redirect server then looks up the destination (IP) address and then returns it to the originator of the call. (8)

Real-time Transfer Protocol (RTP)

Real-time Transfer Protocol (RTP) is the protocol designed to handle real time traffic on the internet. RTP ensured the service quality and reliable data transmission. RTP provides end-to-end delivery services for data (such as interactive audio and video) with real-time characteristics. At first, it was designed to support multiparty multimedia conferences and it is used for different types of applications.RTP is a standard specified in RFC 1889. More recent versions are RFC 3550 and RFC 3551.

Signalling Control between Network Elements

Media Gateway Control Protocol (MGCP) segments the functionality of a traditional voice switch into three functional units:

The media gateway: informs the call agent of service events

The media gateway controller (call agent): manages signal control and informs the media gateways to start an RTP session between two endpoints.

The signaling gateway

Megaco/H.248: a new collaborative standard between IETF and ITU

Its primary focus is the promotion of standardized IP telephony equipment (such as Cisco and Siemens VoIP equipment) (8)

Codec

Codec is an important issue for the VoIP service. For acceptable voice quality service to select a codec that produces compressed audio. They are:-

A G.711 codec produces audio uncompressed to 64 Kbps.

A G.729 codec produces audio compressed to 8 Kbps.

A G.723 codec produces audio compressed to 5.3 to 6.3 Kbps.

PCM (Pulse Code Modulation)

The digitization of analog voice signals is a must to transmit voice over the digital IP network PCM is one of them. PCM modifies the pulses created by PAM to create a complete digital signal. PAM (Pulse amplitude modulation) takes an analogue signal , samples it and generate a series of pulses based on the results of the sampling. To do so, PCM first quantized the PAM pulses. Quantization is a method of assigning integral values in a specific range to sampled instances. Each value is translated into its -7bit binary equivalent. The eight bit indicates the sign. The binary digits are then transformed to a digital signal by using one of the line coding techniques.

Digital signal

Analog signal

Sampling

Coding

Quantizing

Figure. PCM Transmitter Block Diagram

Mobile VoIP Service provider

Some mobile operators are now drawing a growing amount of attention providing mobile VoIP service as well as with their regular services. There are some mobile VoIP service providers as follows:-

Skype Options

Service: Skype Mobile Platform/Network: This java bases application is very popular all over the world and it can runs all most all the phone and many mobile networks. This service is free with stable internet connection. It has some outstand features like Group chatting, presence settings (offline, online, do not disturb), and Skype-to-Skype calls (including Skype In).

Service: 3Skypephone Platform/Network: This service need a specialized handset which are available in the UK, Italy, Austria, Hong Kong, Australia, Ireland, Denmark and Sweden. The phone costs £49.99 (about $98) and can be used on a pre-paid basis. Calls cost nothing if they're made from Skype. Free Skype to Skype and also included some other features.

Service: iSkoot Platform/Network:

This service can provide only a specific number of phones like Windows Mobile , Nokia, BlackBerry and Palm OS models. It works on GSM network. its cost based on its usages.

Because iSkoot is a hybrid VoIP/GSM service, it uses SMS and mobile minutes when making and receiving calls or Skype IM messages. 

Service: Fring Platform/Network: Nokia/Symbian handsets, Widows Mobile, iPhone (pre-release beta) Cost: Free Features: Allows you to make VoIP calls on any SIP network, Skype or to other Fring users. Additionally, Fring is a multi-protocol IM client that will allow you to chat with your buddies on Skype, MSN, ICQ, Google Talk, Twitter, AIM and Yahoo. (5)

Conclusion

Mobile VoIP is now in its infancy. Some mobile operators are providing some limited services. VoIP is mainly depended on the Internet bandwidth. In that reason voice quality also vary from time to time. In most of the cases VoIP is an advantage for the users. As well as the mobile phones have some limitations to get the whole advantage from the mobile VoIP.

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