VoIP (Voice over IP) Chapter 7 PDF

EGEC4120 – TELECOMMUNICATION NETWORKS and SWITCHING CHAPTER 7 - VOIP (VOICE OVER IP) Prepared by ANALENE MONTESINES NAGAYO And MOHAMED YOUSUF HASAN Outcome Covered OC 6: Demonstrate a broad understanding of Multimedia representation and the in...

EGEC4120 – TELECOMMUNICATION NETWORKS and SWITCHING CHAPTER 7 - VOIP (VOICE OVER IP) Prepared by ANALENE MONTESINES NAGAYO And MOHAMED YOUSUF HASAN Outcome Covered OC 6: Demonstrate a broad understanding of Multimedia representation and the interaction with telecommunications protocols including IP Telephony  What is VoIP? A technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Mostly, bypasses the existing telephone system. VoIP is a highly complex digital voice system that relies on high-speed internet connection, phone companies supplying DSL and broadband systems including Wireless. Based on ITU-U, Voice over Internet Protocol (VoIP) is referred to as and broadly includes Voice over Broadband (VoB),Voice over Digital Subscriber Line (DSL), Voice over Internet (VoI), Voice over Wireless Local Area, Network and Internet telephony.  What is VoIP? Some VoIP services may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long-distance, mobile, and international numbers. Also, while some VoIP services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter. The implementation of VoIP service is attractive for subscribers because it reduces the cost of international and long-distance calls, and it is also attractive to ISPs because it would increase the usage of Internet services. Cisco 7970 IP Phone How does VoIP / Internet Voice Works?  How does VoIP / Internet Voice Works? The voice is first amplified and digitized by an analog to digital converter(ADC) which is part of a coder-decoder(codec) circuit that also include DAC. ADC samples the voice signal at 8kHz and produces a 8-bit word for each sample, in a serial manner. A relatively more bandwidth is needed to transmit the bit stream(64kHz or more).  How does VoIP / Internet Voice Works? To reduce the data rate and bandwidth needed, the bit stream is processed and compressed by the DSP. G.729, narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. G.729 is officially described as coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS- ACELP). Because of its low bandwidth requirements, G.729 is mostly used in VoIP applications when bandwidth must be conserved. G.723.1 is Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s All the compression algorithms are stored in DSP memory and processed using DSP for providing echo cancellation.  How does VoIP / Internet Voice Works? At the transmitting phone, voice (analogue signal) is converted using codecs into digital data packets and sent along the internet highway. At the receiving phone, once the digital data packets arrive at their destination, they’re converted back into voice (analogue), allowing communication to happen. VoIP recovers the original packets. The compressed data is extracted, decompressed by a DSP, and send to CODEC. The CODEC converts the digital data into the original sound then it is heard in the handset.  Application for VoIP Case -I Case -II Case -III Case I to III shows the three possible ways to make telephone call over the Internet. Case -I 1. In the first application, a telephone subscriber dials the telephone number of the local gateway for an IP telephone service provider. 2. The call travels over the PSTN to the nearest gateway that acts as an access point to the Internet. 3. The service providers have their own telephone number prefix that connects a customer to the right gateway. 4. The caller enters the destination telephone number, and the gateway in the local office establishes a connection over the Internet to the gateway in his remote office closest to the destination. 5. Then, the gateway in the remote office calls the destination subscriber via the local PSTN.  Internet routing and speech processing is performed by the gateways and ordinary telephones can be used for the call. 6. Now the Internet, instead of PSTN, carries a long-distance section of the call.  In this way, international calls in particular can be provided at very attractive fees because only the local part of a telephone network is involved in the call. IP address to Phone Number VoIP look for IP address ◦ Translate Phone numbers to IP addresses The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. Soft switches know: ◦ Where the endpoint is on the network ◦ What phone number is associated with that endpoint ◦ The current IP address assigned to that endpoint If the soft switch does not have the information, the request is handled by another soft switch. Case -II The second application illustrates a customer surfing the Internet and a Web service provider with enhanced WWW service using VoIP. People surfing the Web can connect to a company’s call center by clicking a Call button located on the company’s Web page. Users can communicate with a customer service group, ordering department, or help desk by using their Web browser and a personal computer equipped with a compatible speech encoder. This is an important new feature as the commercial use of Internet technology. Case -III The third example shows a company with locations in multiple sites and an intranet. Intranet connections between offices A and B use the IP network. The IP network may carry secure VPN(Virtual Private Network) connections where IP packets are tunneled and ciphered, although firewalls. Now, transmission capacity is available between offices and in addition to data VoIP can carry voice via the same VPN connections. There is no need to lease separate channels from a PSTN service provider for speech only. In office B no Private Automatic Branch Exchange (PABX) equipment is required if the PCs contain suitable sound cards, headsets, and software for VoIP. External calls can be done via PABX in office A.  VoIP PROTOCOLS Real-Time Transport Protocol – UDP  One protocol is the Real-Time Transport Protocol, which operates on the top of UDP.  UDP protocol does not guarantee high QoS. Because it has no control on the lower-layer protocols.(flow control and Error control ). Resource Reservation Protocol (RSVP)  Another protocol designed for control of lower layers is the Resource Reservation Protocol.  An endpoint uses RSVP to request a simplex flow through an IP network with specified agreed QoS level for example, delay, throughput jitter and bandwidth.  But having signaling issues, that is, how a telephone call is established over an IP network.  VoIP/ PROTOCOLS The main two protocols to overcome signaling issues are: H.323 and SIP  H.323 recommendation of ITU-T ◦ Most widely used protocol ◦ provides specifications for real-time, interactive videoconferencing, data sharing, and audio applications (VoIP) ◦ H.323 is a complex protocol that was originally designed for video conferencing and later adapted for VoIP. ◦ primarily used for enterprise networks and requires more configuration to work with non-H.323 devices. ◦ It includes a large number of features and options, making it more difficult to set up and configure than SIP. ◦ uses a complex signaling mechanism that requires  VoIP PROTOCOLS  Session Initiation Protocol (SIP) of IETF (Internet Engineering Task Force). ◦ More streamlined protocol ◦ Developed specifically for VoIP ◦ SIP is a more flexible protocol that can be used with a wider range of devices and networks, including mobile devices and wireless networks. ◦ SIP uses a simple call setup process that is similar to making a phone call, where the caller sends an invitation to the recipient and waits for a response. ◦ SIP is more firewall-friendly than H.323, as it uses a simple request-response mechanism that can be easily translated by firewalls.  VoIP PROTOCOLS  Both support signaling for multimedia sessions including video in addition to ordinary telephone calls. They define signaling messages, which are exchanged for call establishment, maintenance, and clearing.  SIP PROTOCOL STACK CODEC, an abbreviation of coder-decoder or compression-decompression, is a standard used for compressing and decompressing digital media, especially audio and video. RAS (Registration Admission Status): Protocol that administers user authorization, monitoring, & permission levels. H.255.0 (Call Signaling): Sets up the connection between the H.323 endpoints after the handshake. H. 245 is a protocol for the transmission of call management and control signals in packet- based networks. Q. 931 (also called Q931) is a signalling protocol SIP Protocol for Integrated Services Digital Network (ISDN) Stack communications that is used in (VoIP).  Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) are used in combination making it possible to monitor data delivery for large multicast networks.  RTP carries the media streams, while RTCP is used to monitor transmission statistics and quality of service (see QoS).  VoIP ADVANTAGES For businesses: Security at a lower cost For consumers: Significantly lower costs For service providers: Lower investment, capital and operating costs for operators Less bandwidth requirements Low cost / no cost software and hardware Free VoIP to VoIP Low cost VoIP to Public Switch Telephone Network (PSTN)  VoIP ADVANTAGES oMobility  Any internet connection  Growing number of wireless broadband locations  Popular communication apps, like Skype, Zoom, FaceTime and WhatsApp, use VoIP for local and international telephone calls.  VoIP DRAWBACKS Quality ◦High quality PSTN ◦Variable VoIP dependent on connection Dependent on wall power Lost or delayed packets cause drop-out in voice Emergency Calls Hard to find geographic location Security ◦Most VoIP services do not support encryption  VoIP DRAWBACKS Takes relatively long time to transmit the voice signal Reassembling the packets at the destination takes time. Since the packets passes through numerous routers and servers each adding transit time or latency. Latency is the delay between the time the signal is transmitted and the time its is received. Maximum acceptable latency is 150 ms. INTERNET PHONE SYSTEM Two basic types of IP phones: 1.Home VoIP High Speed internet service VoIP interface  ATA(Analog Terminal Adapter)-This connects the standard home telephone to the existing broadband internet modem. 2. Enterprises IP phones DRILL PROBLEM 1. A special PCM system uses 16 channels of data, one whose purpose is for synchronization. The sampling rate is 3.5kHz.The word length is 6 bits. Find (a) The number of available data channels (b) The number of bits/frame (c) The serial data rate. DRILL PROBLEM-answer a.16(total channels)-1 (channel for synchr)=15 (for data) b.bits/frame=6*16=96 c.serial data rate=sampling rate*no of bits/frame=3.5 kHz*96=336kHz. DRILL PROBLEM 2. A VoIP system uses the G.711 codec (also known as u-law or a-law) with Packetization Interval is 0.02 ms and the number of concurrent Calls is 10. Calculate the bandwidth needed for the number of simultaneous calls.  DRILL PROBLEM - answer Calculation: a. Codec Bit Rate: G.711 typically uses an 8-bit PCM (Pulse Code Modulation) sample, and for a voice call, two samples (one in each direction, for full-duplex communication) are transmitted every packetization interval. Therefore, the codec bit rate can be calculated as follows: Codec Bit Rate = (8 bits/sample * 2 samples) / 0.02 ms Codec Bit Rate = 800 Kbps or 0.8 Mbps  DRILL PROBLEM - answer Calculation: b. Bandwidth for One Call: Each VoIP call requires the codec bit rate for both the sending and receiving directions. Therefore, the bandwidth for one call is: Bandwidth per Call = Codec Bit Rate * 2 Bandwidth per Call = 0.8 Mbps * 2 = 1.6 Mbps c. Total Bandwidth for Concurrent Calls: To calculate the total bandwidth for a given number of concurrent calls, simply multiply the bandwidth per call by the number of calls: Total Bandwidth = Bandwidth per Call * Number of Concurrent Calls Total Bandwidth = 1.6 Mbps * 10 = 16 Mbps  DRILL PROBLEM 3. Calculating packet loss rate is important in VoIP to ensure quality. If 200 packets were sent and 20 packets were lost, calculate the packet loss rate. Packet Loss Rate=(number of lost packets/ total number of packets) ×100 Packet Loss Rate=(20/200)*100=10%  DRILL PROBLEM 4. Jitter buffer size is a parameter that is considered to determine the network latency and jitter. If the maximum jitter is 20 ms, latency is 50 ms, and an additional 10 ms buffer, calculate the jitter buffer size. Jitter Buffer Size = maximum jitter + latency + additional buffer for safety Jitter Buffer Size=20ms+50ms+10ms=80ms REFERENCES 1. Principle of Electronics communication systems by Louis E.Frenzel Jr(3rd Edition). 2. Introduction to Telecommunications Network Engineering. ( second Edition Masader – )Oman by Tarmo Anttalainen Virtual Science Library https://www.masader.om/eds/detail?db=e000xww&an=87734 &isbn=9781580536165