A Seminar Report On VoIP TECHNOLOGY Submitted in partial fulfillment of award of degree of Bachelor of Technology In Electronics and Communication SUBMITTED BY: ROLL No- B.TECH 3rd YEAR SEMINAR IN-CHARGE Mr. PRADEEP KUMAR Department Of Electronics and Communication Kamla Nehru Institute of Physical & Social Sciences Faridipur, Sultanpur-228118 1 Session 2012-13 Department of Electrical & Electronic Engineering Kamla Nehru Institute of Physical & Social Sciences, Sultanpur CERTIFICATE This is certified that seminar entitled “VOIP TECHNOLOGY” which is submitted by Ms.GARIMA BHARDWAJ, B.Tech (Third Year) Electronics and communication, is a partial fulfillment towards the award of Degree of Bachelor of Technology in Electronics and Communication Engg. Ms.GARIMA BHARDWAJ, B.Tech (Third Year), Ec, has prepared the seminar under my guidance in the session 2012-13 and delivered it successfully. Date: March/19/2013 HOD SEMINAR INCHARGE Mr. R .K.YADAV Mr.PRADEEP KUMAR Electronics and communication Electronics and communication 2 ACKNOWLEDGEMENT I thank my seminar guide Mr. PRADEEP KUMAR, Lecturer, for his proper guidance, and valuable suggestions. I am indebted to Mr. R.K. YADAV, the HOD, Electronics and communication division & other faculty members for giving me an opportunity to learn and present the seminar on the topic "voip technology”. If not for the above mentioned people my seminar would never have been completed successfully. I once again extend my sincere thanks to all of them. 3 ABSTRACT The growing excitement surrounding the transport of telephony services over traditional data networks such as the Internet, corporate-enterprise intranets and new service provider extranets has led to the development of cost efficient gateway equipment based on embedded systems that converts analog telephony information such as voice and fax into packet data suitable for transport over IP, Frame Relay and ATM networks. As a result, the long-time promise of being able to replace or enhance the traditional PBX by combining voice and data services onto a single network can now finally is realized. In order to do so, a very low-cost telephony device capable of directly exchanging IP packets with the data network is required. Development of this 'VoIP Phone' will require the development of a 'system on a chip' which combines digital signal processing functions, microcontroller functions, analog interface, telephone user interface, network interface, and associated glue logic. Voice over IP (VoIP) is an alternative to traditional circuit-switched telephony that allows human voice and video to travel over existing packet data networks along with traditional data packets. This report looks at the functional requirements and features of an IP Telephone and examines the implementation issues that must be considered. 4 TABLE OF CONTENT Chap-1 Introduction 10-12 1.1 VoIP Overview 10 1.2 VoIP History 11 1.3 Consumer Market 11-12 Chap-2 VoIP Principles 13- 18 2.1 Analog to Digital Conversion 13 2.1.1 Resolution 14 2.1.2 Accuracy 5 14 2.1.3 Quantization Error 14 2.1.4 Sampling Rate 14-15 2.2 Processing of Digital Signal and Compression 15-16 2.3 Transport and Signaling Protocol in VoIP 16 2.3.1 SIP Signaling Protocol 16-17 2.3.2 H.323 Signaling Protocol 17 2.4 End to End VoIP Network 18 Chap-3 VoIP Advantages and Disadvantages 19-21 3.1 Advantages 19 3.2 Disadvantages 19-20 3.3 Quality of Service 20 6 Chap-4 Mobile VoIP 21-22 4.1 Technologies 21 4.2 Recent Developments 22 4.3 Software Clients 23 Chap-5 VoLTE 24 Chap-6 International VoIP Implementation 25-26 6.1 IP telephony in Japan 25-26 Chap-7 conclusion 27-28 References 29 7 LIST OF FIGURES 1. Residential Network Using VoIP 11 2. 4-Channel Stereo Multiplexed Analog-to-Digital converter 13 3. Analog-to-Digital conversion Signal 15 4. End to End VoIP Network 18 8 9 ABBREVIATIONS VoIP Voice over Internet Protocol PSTN Public Switch Telephone Network VoBB Voice over Broad Band IP Internet Protocol SMB Small to Medium Business SIP Session Initiate Protocol QoS Quality of Service IETF Internet Engineering Task Force TCP Transmission Control Protocol UDP User Datagram Protocol SCTP Stream Control Transmission 10 Protocol HTTP Hyper Text Transfer Protocol SMTP Simple Mail Transfer Protocol MIKEY Multimedia Internet Keying RAS Registration, Admission and status PC Personal Computer UMA Unlicensed Mobile Access VoLTE Voice over Long Term Evolution ITSP Internet Telephony Service Provider HSDPA High Speed Downlink Packet Access 11 12 Chapter-1 INTRODUCTION 1.1 VoIP Overview Voice over IP (VoIP, or Voice over Internet Protocol) commonly refers to the communication protocols, technologies, methodologies, and transmission techniques involved in the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Other terms commonly associated with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. Internet telephony refers to communications services —voice, fax, SMS, and/or voice-messaging applications— that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication, while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls 13 14 1.2 VoIP History Early providers of Voice over IP services offered business models (and technical solutions) that mirrored the architecture of the legacy telephone network. Second generation providers, such as Skype have built closed networks for private user bases, offering the benefit of free calls and convenience, while denying their users the ability to call out to other networks. This has severely limited the ability of users to mix-and-match third-party hardware and software. Third generation providers, such as Google Talk have adopted the concept of Federated VoIP - which is a complete departure from the architecture of the legacy networks. These solutions typically allow arbitrary and dynamic interconnection between any two domains on the Internet whenever a user wishes to place a call. 1.3 Consumer Market A major development that started in 2004 was the introduction of mass-market VoIP services that utilize existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing. Fig:1:- Residential network using VoIP 15 VoIP solutions aimed at businesses have evolved into unified communications services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as discrete units that can all be delivered via any means and to any handset, including cell phones. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market. 16 Chapter-2 VoIP Phone Principles VoIP works like that. First the A/DC (Analog to Digital Converter) to convert analog voice to digital signals. Now the bits have to be compressed in a good format for transmission. Then we have to insert our voice packets in data packets using a real- time protocol (RTP over UDP over IP). For signaling between customers terminal unit we need a signaling protocol ITU-T H.323 or SIP (Session Initiate Protocol). At Rx site we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card (or phone). All that must be done in a real time fashion cause we cannot waiting for too long a vocal answer. On voice transport via packet oriented network could be respect requirements of QoS (Quality of Service) 2.1 Analog to digital conversion An analog-to-digital converter (abbreviated ADC, A/D or A to D) is a device that converts a continuous quantity to a discrete time digital representation. An ADC may also provide an isolated measurement. The reverse operation is performed by a digital-to-analog converter (DAC). An ADC has an analog reference voltage or current against which the analog input is compared. The digital output word tells us what fraction of the reference voltage or current is the input voltage or current. 17 Fig:2:- 4-channel stereo multiplexed analog-to-digital converter Factors for analog to digital conversion 2.1.1 Resolution The resolution of the converter indicates the number of discrete values it can produce over the range of analog values. The values are usually stored electronically in binary form, so the resolution is usually expressed in bits. In consequence, the number of discrete values available, or "levels", is a power of two. For example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels, since 28 = 256. The values can represent the ranges from 0 to 255 (i.e. unsigned integer) or from −128 to 127 (i.e. signed integer), depending on the application. 2.1.2 Accuracy An ADC has several sources of errors. Quantization error and (assuming the ADC is intended to be linear) non-linearity are intrinsic to any analog-to-digital conversion.