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  1. 1. VoIP www.bestneo.com1.INTRODUCTIONThe public telephone network and the equipment that makes it possible are takenfor granted in most parts of the world. Availability of a telephone and access to alow cost ,high quality worldwide network is considered to be essential in modernsociety .Anything that would jeopardize this is usually since more and morecommunication is in digital format and transported via packet networks such asIP,ATM cells etc. Since data traffic is growing much faster than telephone traffic,there has been considerable interest in transporting voice over data networks.Support for voice communication using the Internet Protocol (IP), which isusually just called “voice over IP” or VoIP, has been become especially attractivegiven the cost, flat rate pricing of the public internet. In fact, toll quality telephoneover IP has now become one of the key steps leading to the convergence of thevoice, video, and data communication industries. The feasibility carrying voiceand call signaling messages over the Internet has already been demonstrated butdelivering high quality commercial products, establishing public services, andconvincing users to buy in to the vision are just beginning.VoIP can be defined as the ability make telephone calls and to send facsimiles orIP based data networks with suitable quality of service (QoS) and much superiorcost / benefit. Equipments producers see VoIP as a new opportunity to innovatecompete. The challenge for them is turning this vision in to reality by quicklydeveloping new VoIP enabled equipments. For Internet service providers the
  2. 2. VoIP www.bestneo.compossibility of introducing usage based pricing and increasing their traffic volumesis very attractive. Users are seeking new types of integrated voice / dataapplication as well as cost benefits.Success fully delivering voice over packet networks precedence a tremendousopportunity; however, implementing the products is not as straight forward atask as it may first appear.2. INTERGRATION OF IP & PSTNUsing VoIP we can communicate from PC to any other device – PC, phone, internetphone, PDA, etc.1.PC-PC: we can community from one PC to another PC using VoIP i.e. voice mailor voice chat. For this PC with headphone, a microphone and sound card arerequired.2.PC- Phone: Allows PC user to establish a call with conventional phone. Thisfacility enables us to make long distance a call with rate cheaper thanconventional telephone call.3.Phone – Internet Phone: Extension of PC to phone architecture. The IP phoneconnects to the Internet through a cable or DSL modem using an RJ-45 Ethernetjack.3. PSTN vs. INTERNET.To understand Internet telephony or VoIP, it is necessary to be familiar with thefundamental principles behind the Internet and how it compares to PSTN.
  3. 3. VoIP www.bestneo.comAlthough the Internet shares some characteristics of the PSTN, it also hassignificant differences.Computer to computer data communication was far cheaper than voicecommunication through Internet. One of the main reasons for these is the basicdifference in the technologies used for voice and data communication. Voicenetworks use circuit switching while data network use packet switching. Whatthis means is that for a voice conversation to happen, one complete andcontinuous circuit has to be held open between the two parties during the fullduration of conversation, while for a data transmission such a continuous connectis unnecessary. In data transmission, the data is split into independent packetsthat can take alternate rules to the intended destination, where they arereassembled. In voice communication, circuit switching is used because it is ableto handle data in real time. Packet switching, on the other hand, can lead todelays, which can lead to considerable degradation in the quality of theconversation.So the networks when their separate ways till the Real time Transport Protocol(RTP) was developed. RTP provides support for streaming audio and video overcomputer networks. Unlike what its name seems to suggest, RTP does not ensurereal time delivery of data. Instead it provides mechanisms for times stamping thepackets and methods for synchronizing data streams with time properties. Thusit became technically possible to send voice over the net.PSTN INTERNET
  4. 4. VoIP www.bestneo.com1.circuit switching 1. packet switching2.dedicated path between calling and 2. There is no dedicated Calledparty. path between Sender and. receiver3.Has a guaranteed QoS . 3. Cannot guarantee QoS4. Reserve required bandwidth in advance 4.It acquires and releases bandwidth, as it is needed.5. Cost is based on distance and time. 5.Cost mechanism for Internet telephony is not dependent on time and distance.4. HOW VoIP WORKS?In VoIP, for transmitting voice, it is pocketsize and then transmitted throughpacket switching networks. This process is described in following steps.1. Our voice is converted into analog electrical signal using micro phone2. These electrical signals are digitized using PCM-Pulse Code Modulation PCM samples analog signal at a rate of 8000 samples/sec and coded into 64 kb/sec. Each sample there fore represents 125 microseconds of a voice stream, and is 8 bits, or 1byte long.3. These digital voice samples are buffered on IP gate way- it converts PCM data into IP packets using DSP. DSP s(Digital Signal Processors ) are responsible for converting from analog to digital as well as compression. This has following steps.
  5. 5. VoIP a. Remove line echoes: Echo becomes a problem when the round trip delay is more than 50milli seconds. Since round trip delay for VoIP is always greater than 50 ms, echo cancellation is a requirement. Therefore line echoes are cancelled using a digital filter. b. Silence cancellation using Voice Activity Detector (VAD):It checks the speaks for all the moments of silence .The length and beginning of the pauses is noted, while the remaining silence is removed from data set. Similarly redundant data is also removed, making the data set more compact so that it takes up much less band width. c. This digitized voice signals are compressed and framed on the basis of ITU standards G.729. d. This voice frame is converted into IP packets. First voice frame is converted into RTP packet by adding 12 byte header, for sequencing the data packet. Then 8- byte UDP header with source and destination number added. Then20-byte IP header containing the gateway IP addresses added.4. Voice packet is sending on the Internet, it finds its way to the destination just like any other data packet. It passes through various routers and switches to reach the destination gate way.5. At receiver VoIP system reverses process for voice play back. System extracts IP packet- UDP packet – RTP packet-compressed voice frame-analog form of voice.
  6. 6. VoIP Figure showing 5 steps of Internet Telephony
  7. 7. VoIP www.bestneo.com5. VoIP: COMPONENTS Sender’s end:The PCM Interface, which receives samples from the telephony interface andforward s them to the VoIP software module for processing (and vice versa).The Echo Cancellation Unit, which performs echo cancellation on sampled, fullduplex voice port signals in accordance with the ITU G.165 or G.168 standards.The Voice Activity Detector, which monitors the input device for device so that theprocess of sending it over the network occurs only in case of voice activity at themouthpiece.The Tone Detector, which detects the reception of DTMF tones and discriminatebetween voice and facsimile signalsVoice Encoder, which encoded the voice signals from the PCM Interface and sentover to the host interface that is the gateway to the carrier like Internet Protocol
  8. 8. VoIP www.bestneo.comComfort Noise Encoder, which measures idle noise level and reported to thedestination so that comfort noise can be inserted into the call (so that the listenerdoes not get dead air on their telephone).Voice fax classifier, which separates voice and fax signals, if we are using thesame set up for fax also.Modem /Fax Demodulator, an optional element for processing fax data. Receiver’s End:Comfort Noise Generator, which generates a comfort noise at the receivers end, asper the data received from the comfort noise encoder at the sender’s end.Voice Decoder, which decodes the encoded signal.Tone /DTMF Generator, which generates DTMF tones and call progress toneunder command of the operating system.
  9. 9. VoIP www.bestneo.comBad Frame Handler, which detects the corrupted packet and may ask for it to beretransmitted.6. IMPLEMENTING VoIP.Most of the VoIP implementation follows the ITU H.323 standard. The H.323standard was originally a multimedia standard meant for transferring audio andvideo data over network. Basic element of H.323 architecture 1. The terminals. 2. Gate way 3. Gate keepers 4. Multipoint control unit (MCU)Out of these, the first two are key elements, while the other two are optionalcomponents.Terminals- is an end user device. These could be PCs with headphones and amicrophone, or, IP telephone or telephone.
  10. 10. VoIP www.bestneo.comGateway-is an intermediate device to provide inter operation between H.323device and non-H.323 device. The gateway can be designed inside a PBX (privatebranch exchange) or stand alone device such as a router. One side of this gatewaywill be the PSTN or a company’s IP WAN, while the other side would have thecompany’s internal network.Gate keepers – is a piece of software controlling gateways. Simply speaking, agatekeeper acts as a routine device for voice calls. They do address translations todetermine the calling terminals. They can also control access given to the otherVOIP devices such as terminals, gateways etc. They determine how muchbandwidth is required for a voice call, and can perform other functions like callauthorization, bandwidth management, call management, and a directory control.Multipoint control unit (MCU) – This is meant to provide conferencing facilitiesbetween three or more H.323 terminals or gateways.7. STANDARDSOne important consideration of VoIP software is interoperability. Many Internettelephony products require all communicating parties to use the same application.All the vendors begin to support an international can implement thistelecommunications union (ITU-T) standard for multimedia communicationssystems known as H.323.It is important standard set of procedures that provide aspecification for voice, data and video communications over packet switchednetwork (or TCP/IP networks). Using H.323, a device or application from onevendor can place a call to another. It tries to provide real time audio, video anddata communications by making use of Real time Transport Protocol (RTP) which
  11. 11. VoIP www.bestneo.comwas proposed by IETF (Internet Engineering Task Force) as a standards for realtime data transmission over internet .The H.323 recommendation is independentof network topology.H.323 offers reduced bandwidth requirement for conversions from 64Kbps/eachto under 10Kbps /each. This is nearly an order of magnitude improvement .netmeeting and Vocal Tec Internet Phone supports H.323. The most widely acceptedstandard now is the H.323 set of protocol. However recently Session LimitationProtocol (SIP) is coming up an alternative for H.323 signaling8. HOW VOICE GOES OVER IP?When we talk of sending voice over Internet connections, the first question thatpops up is “how can you do that, when the connections are not even broad enoughto handle regular data?”. Traditionally, voice is a lot of data and if sent as aregular analog data, it would simply clog the networks and won’t be able to getthrough. Instead the voice is sampled, compressed and packetized to send it overan IP network. This may need much less bandwidth.
  12. 12. VoIP www.bestneo.comThe compression technique used in VoIP is similar to the compression concept ofMP3s.Regular music or audio is compressed by checking out the noise, silence andcertain in audible frequencies, so that it takes up much less space.There was a speech codecs, also called voice coders or ‘vocoders’ are speechcompression algorithms that let us drastically reduce the amount of data that goesinto the network, while still preserving voice quality. The devices that send andreceive audio and video work on the H.323 protocol Vocoders are useddepending on the network conditions. When we talk of sending voice over IP, theoverall realism depends on sound quality and latency. Latency is the delaybetween the times; the voice is spoken at the originating end and the time it’sperceived at the receiving end. There needs to be a trade-off between the two,which means that if we go in for higher speech quality (less compression), wemight have more latency. So VoIP solutions come with standard vocoders.The amount of network traffic is not the same at all times. So, VoIP systems aregenerally dynamic, that is, they support multiple vocoders with the codec beingautomatically switched to match the network conditions, that is, if the networktraffic increases the system switches to a high compression rate vocoders and viceversa. This however increases the system complexity and resource requirements.9. LIMITATIONS AND SOLUTIONSThere are various problems associated with VoIP.
  13. 13. VoIP 1. Packet-delay –The voice packets may get delayed in reaching the destination. If a router is free, it sends the packet quickly and if it is under heavy load of traffic the voice packet will get delayed, leading to latency. The route taken by a packet can also take delay. If it takes a short path, it will reach quickly. And if it reaches the destination after the multiple hops the delay would be longer. 2. Packet error & Packet loss. If a voice packet encounters a bad router, it might get corrupted or get lost all together. If it is the former, the packet that reaches the destination is of no value. If it gets lost, then the router may ask for it to be re-transmitted.Solutions: 1. Traffic priorization-Routers can be configured to give preference to certain type of packets over others. So voice packet can be given higher priority over normal data packets. 2 Weighted fair queuing-Here; a minimum amount of bandwidth is allocated to certain traffic, in this case voice. This can be done using the Resource Reservation Voice Protocol (RSVP).10. ADVANTAGES: 1. Cost reduction-Long distance voice telephony is where telcos make most of their money. But using VoIP we can able to make long distance calls with cheaper rate. 2. Simplification-an integrated infrastructure that support all form of communication allows more standardization and reduces the total
  14. 14. VoIP equipment complement. This combined infrastructure can support dynamic bandwidth optimization and fault tolerant design. These differences between the patterns of voice and data offer further opportunities for significant efficiency improvement. 3 Consolidation-since people are among the most significant cost elements in an network, any opportunity to combine operations, to eliminate points of failure, to consolidate accounting system would be beneficial. 4. Advanced applications-even though basic telephony and facsimile are the initial applications for VoIP, longer benefits are expected to be derived from multimedia and multi-service applications.11. THE DEATH OF DISTANCELong distance voice telephony is where telcos make most of their money. Thefurther away our call is to the more we are charged. This means that we thinktwice before making that long distance call, and even when we make the call wekeep the conversation short. Traditional economics says that if the cost of the callwere reduced, the more people would make the call, and if properly done, thevoice service provider could end up making more money than before. So telcoshave extended the distance we can call for the same amount.When we send mail over the internet we do not pay different rates for differentdestinations of the mail .we do not pay more for our internet connection when weare downloading software from an FTP server in Iceland as against when we areaccessing a web page on a sever in Mumbai. The fact that we can do both
  15. 15. VoIP www.bestneo.comsimultaneously, from different windows of our browsing makes a switch over to adistance-based model impossibly complex. We do not even pay for the amount ofdata transferred. We just pay for the bandwidth we opted for and the time westayed when voice traffic moves over completely to IP based networks, it just possiblethat we would pay the same for an international callas we would for a local call . inshort , the concept of long distance calls just disappear .12. APPLICATON AND SERVICES OF INTERNET TELEPHONY. 1. Integration of data voice and fax. 3. Sound grading. 4. Video telephony. 5. Unified messaging. 6. A virtual second line. 7. Web based call centers. 8. Low cost voice calls. 9. Real time billing. 10. Remote Teleporting. 11. Enhanced teleconferencing.13. CONCLUSIONWith the advances in computer technology, phenomenal growth in Internet useand declining cost of computer hardware, there has been growing interest inrecent years in developing real time voice communication software for Internet
  16. 16. VoIP www.bestneo.comtelephony. In spite of the numerous Internet telephony systems available today,these are still at infancy and far from being substitutes of conventional telephonysystems. While the Internet telephony may have difficulty matching the reliabilityof switched voice networks, it is dirt-cheap and is growing rapidly. Moreover withadvances in compression algorithms, standards, network technology and higherbandwidth in future, these problems will diminish overtime.