VOICE OVER INTERNET PROTOCOL Y.Mallikarjunareddy, Finalyear E.C.E Branch, Roll No: 082J1A0429, AVR&SVR Engg College, Kurnool,AndhraPradesh, Email Id: email@example.com ABSTRACT VoIP (voice over IP - that is, voice delivered using The first VoIP application was introduced in 1995the Internet Protocol) is a term used in IP telephony - an "Internet Phone". An Israeli company by thefor a set of facilities for managing the delivery of name of "Vocal Tec" was the one developing thisvoice information using the Internet Protocol (IP). application. The application was designed to run on aVoIP is therefore telephony using a packet based basic PC. The idea was to compress the voice signalnetwork instead of the PSTN (circuit switched). The and translate it into IP packets for transmission overapplication was designed to run on a basic PC. The the Internet. This "first generation" VoIP applicationidea was to compress the voice signal and translate it suffered from delays (due to congestion),into IP packets for transmission over the Internet. disconnection, low quality (both due to lost and outThe most significant development is gateways that act of order packets) and incompatibility.as an interface between IP and PSTNnetworks. Vocal Tec’s Internet phone was a significant Keywords-VoiceProcessingModule;PCM breakthrough, although the applications manyInterface;EchoCancellation;IdleNoiseDetection; problems prevented it from becoming a popularTone Detector;Packet VoiceProtocol;Voice Playback product. Since this step IP telephony has developedModule; Call Signaling Module; Packet Processing rapidly. The most significant development isModule;Network Management Protocol. gateways that act as an interface between IP and PSTN networks.1. Introduction 2) WHAT IS VoIP? VoIP (voice over IP - that is, voice delivered usingthe Internet Protocol) is a term used in IP telephony Voice over IP (VoIP) is a blanket description forfor a set of facilities for managing the delivery of any service that delivers standard voice telephonevoice information using the Internet Protocol (IP). In services over Internet Protocol (IP). Computers togeneral, this means sending voice information in transfer data and files between computers normallydigital form in discrete packets rather than in the use Internet protocol.traditional circuit-committed protocols of the publicswitched telephone network (PSTN). A majoradvantage of VoIP and Internet telephony is that itavoids the tolls charged by ordinary telephone service VoIP is therefore telephony using a packet basednetwork instead of the PSTN (circuit switched). During the early 90s the Internet was beginningits commercial spread. The Internet Protocol (IP),part of the TCP/IP suite (developed by the U.S.Department of Defense to link dissimilar computersacross many kinds of data networks) seemed to havethe necessary qualities to become the successor of thePSTN.
"Voice over IP is the technology of digitizing The gateway allows the calls to transfer from onesound, compressing it, breaking it up into data network to the other by converting the incomingpackets, and sending it over an IP (internet protocol) signal into the type of signal required by the networknetwork where it is reassembled, decompressed, and it is required to send it on. For example, A PC userconverted back into an analog wave form...” The wishes to call someone using a conventional phone.transmission of sound over a packet switched The PC sends the IP packets containing digitizednetwork in this manner is an order of magnitude more voice to the gateway.efficient than the transmission of sound over a circuitswitched network. 3) REQUIREMENT OF VoIP As mentioned before, VoIP saves bandwidth alsoby sending only the conversation data and not The requirements for implementing an IPsending the silence periods. This is a considerable Telephony solution to support Voice Over IP variessaving because generally only one person talks at a from organization to organization, and depends ontime while the other is listening. By removing the the vendor and product chosen. The following sectionVoIP packets containing silence from the overall aims to identify the fundamental requirements in theVoIP traffic we can reach up to 50% saving. In a general case and is split into 3 sections:circuit switched network, one call consumes theentire circuit. That circuit can only carry one call at a Software Requirementstime. Hardware Requirements In a packet switched network, digital data is Protocol Requirementschopped up into packets, sent across the network, andreassembled at the destination. This type of circuitcan accommodate many transmissions at the same 3.1 Software Requirementstime because each packet only takes up whatbandwidth that is necessary.. Internet Telephony The software package chosen will reflect thesimply takes advantage of the efficiencies of packet organizational needs, but should contain theswitched networks. following modules as defined in the Technology Gateways are the key component required to Guide Series - Voice over IP Publication, and otherfacilitate IP Telephony. sources. A gateway is used to bridge the traditional circuitswitched PSTN with the packet switched Internet. Voice Processing Module. This aspect of the software is required to prepare voice samples for transmission. The functionality provided by the voice processing module should support: A PCM Interface is required to receive samples from the telephony interface (e.g. a voice card) and forward them to the Voice Over IP software for further processing. Echo Cancellation is required to reduce or eliminate the echo introduced as a result of the round trip exceeding 50 milliseconds. Idle Noise Detection is required to suppress packet transmission on the network when there are no voice signals to be sent. This helps to reduce network traffic as up to 60% of voice calls are silence and there is no point in sending silence. A Tone Detector is required to discriminate between voice and fax signals by detecting DTMF (Dial Tone Multi frequency) signals. The Packet Voice Protocol is required to encapsulate compressed voice and fax data for transmission over the network. A Voice Playback Module is required at the destination to buffer the incoming packets before they are sent to the Codec for decompression.
Call Signaling Module. This is required to serve G.723.1 defines how an audio signal with aas a signaling gateway which allows calls to be bandwidth of 3.4 KHz should be encoded forestablished over a packet switched network as transmission at data rates of 5.3Kbps and 6.4Kbps.opposed to a circuit switched network (PSTN for G.723.1 requires a very low transmission rate andexample). delivers near carrier class quality. The VoIP Forum Packet Processing Module. This module is as the baseline Codec for low bit rate IP Telephonyrequired to process the voice and signaling packets has chosen this encoding technique.ready for transmission on the IP based network. G.711. The ITU standardized PCM (Pulse Code Network Management Protocol. Allows for Modulation) as G.711. This allows carrier classfault, accounting and configuration management to quality audio signals to be encoded for transmissionbe performed. at data rates of 56Kbps or 64Kbps. G.711 uses A- Law or Mu-Law for amplitude compression and is3.2 Hardware Requirements the baseline requirement for most ITU multimedia communications standards. Real-Time Transport Protocol (RTP) is the The exact hardware, which would be required, standard protocol for streaming applicationsagain, depends on organizational needs and budget. developed within the IETF (Internet EngineeringThe list below highlights the most general hardware Task Force).required. Resource Reservation Protocol (RSVP) is the The most obvious requirement is the existence (or protocol which supports the reservation of resourcesinstallation) of an IP based network within the branch across an IP network. RSVP can be used to indicateoffice gateway is required to bridge the differences the nature of the packet streams that a node isbetween the protocols used on an IP based network prepared to receive.and the protocols used on the PSTN. The gateway takes a standard telephone signal anddigitizes it before compressing it using a Codec. The 4) HOW VoIP WORKS?compressed data is put into IP packets and thesepackets are routed over the network to the intendeddestination. The PCs attached to the IP based network requirethe voice/fax software outlined above. They alsorequire Full Duplex Voice Cards which allow bothcommunicating parties to speak at the same time - asoften happens in reality. As an alternative to installing Voice Cards, IPTelephones can be attached to the network tofacilitate Voice Over IP. A secondary gateway shouldbe considered as a backup in the event of the failureof the primary gateway.3.3 Protocol Requirements There are many protocols in existence but themain ones are considered to be the following: H.323 is an ITU (InternationalTelecommunications Union) approved standardwhich defines how audio /visual conferencing data istransmitted across a network. H.323 relies on theRTP (Real-Time Transport Protocol) and RTCP(Real Time Control Protocol) on top of UDP (UserDatagram Protocol) to deliver audio streams acrosspacket based networks.
VoIP over the Internet This is probably the best known and most publicized, talking PC to PC. Basically free telephone calls. The call is only free if both parties to the call have access to the public Internet at zero cost... Advantage... free calls regardless of distance or length of call. Disadvantage.... often the voice quality is bad due to the lack of bandwidth available for the call. Other factors. Have to use a PC or other computer running VoIP software. Office to Office A large multinational company will have offices across the whole country. They have a fixed data network connecting all the offices together. This allows every computer access to every other computer in the company. By installing a VoIP Gateway in each office and connecting it to the office legacy PBX and to the data network, employees use the data network for voice calls between offices. Advantages. Interoffice calls are free, since the company already has the bandwidth between offices. The technology is transparent to the user, and requires minimum training. The only new equipment required is a gateway at each office. Voice quality is good, because the company has control over the4.1 How VoIP works: Part 1 bandwidth. Disadvantage. Extra bandwidth may be required Let us look at very simple VoIP call. Consider two between offices, which offset the savings.VoIP telephones connected via an IP network .In this Other factors... The carrier providing theexample both VoIP telephones are connected to a interoffice bandwidth will almost certainly offer anlocal LAN. Sally’s phone has an IP address of alternative solution including management of the192.168.1.1, Bill’s phone is 192.168.1.2, the IP internal telephone traffic.addresses uniquely identify the telephones. Both our IP PBX A traditional Private Branch Exchangephones are configured to use a widely used VoIP (PBX) connects all the phones within an organizationstandard called H.323. to the public telephone network. Essentially IP PBX Bill wants to talk to Sally and his phone knows the replaces all the internal phones with VoIP telephones.IP address of Sally’s phone. Bill lifts the handset and The IP PBX has standard telephone trunkdials Sally, the phone sends a call setup request connections to the public telephone network. The IPpacket to Sallys phone, Sally’s phone starts to ring, PBX is a PBX with VoIP, but it also has the ability toand responds to Bills phone with a call proceeding support VoIP over the Internet and Office to Officemessage. When Sally lifts the handset the phone VoIP.sends a connect message to Bills phone. The twophones will now exchange the data packetscontaining the speech. At the end of the call Billreplaces his handset and phone stops sending voicedata sends a disconnect message and Sallys phoneresponds with a release message. The call is nowcomplete. all the messages contain the Q931 ISDNprotocol. Having introduced VoIP I will now talk aboutthree main types of VoIP installed in the marketplace today. Main ‘types’ VoIP VoIP has broadly three main branches, which canand do overlap.
4.2 How VoIP works part 2: The Protocols. As such, H.323 addresses the core Internet- telephony applications by defining how delay- sensitive traffic, (i.e., voice and video), gets priority I have made an assumption that both ends of a transport to ensure real-time communications serviceVoIP telephone conversation are compatible. This over the Internet. (The H.324 specification definescompatibility only happens if both ends agree to use the transport of voice, data, and video over regularthe same protocol. All manufacturers who claim to be telephony networks, while H.320 defines theproducing industry standard voice over IP either protocols for transporting voice, data, and video oversupport SIP or H.323 protocol. integrated services digital network (ISDN). What is H.323? 4.3 How VoIP works part 3: Encoding Over the next few years, the industry will addressthe bandwidth limitations by upgrading the Internetbackbone to asynchronous transfer mode (ATM), the The call control part of H.323 sets up theswitching fabric designed to handle voice, data, and parameters for the full duplex voice path betweenvideo traffic. Such network optimization will go a source telephone and destination telephone. I willlong way toward eliminating network congestion and continue with my analogies to explain how yourthe associated packet loss. The Internet industry also voice gets transported across the Internet.is tackling the problems of network reliability and In terms of H.323 there is a trade off between callsound quality on the Internet through the gradual quality and bandwidth, in general the higher theadoption of standards. Standards-setting efforts are quality the greater the bandwidth requiredfocusing on the three central elements of Internet During the call setup portion of H.323 the phonestelephony: the audio codec format; transport have to decide which speech encoder/decoder to useprotocols; and directory services. when they send the speech to the other phone, Bill and Sally both have phones that support G.723.1, H.323 Call Sequence: G.711 and G729. The main difference between each of these encoders is the amount of bandwidth they use, G.711 uses 64kbit/s and G.723.1 can use as little as 5.3kbit/s. Although it would seem obvious to use the encoder with the lowest bandwidth, there is a loss of quality with a lower bandwidth.. At the same time a stream of G723.1 encoded voice data starts being sent from each phone to the other phone. 4.4 How VoIP works part 4: Hear the Quality. The performance of the speech encoders at each end, the number of packets lost on route, Latency and Jitter. I have already talked about the encoders in the previous section. I also bundle into the encoding process echo suppression. In the early days of voice calls via satellite there would be an annoying echo. As the technology improved the echo disappeared. Echo suppression is very key to good quality VoIP calls. I do not dwell on the subject since the mathematics is beyond my comprehension. Good echo suppression makes for quality calls. Be warned that because a manufacturer has a G.723.1 encoder it may not sound the same as another manufacturer who claims to have G.723.1, quality does vary. As a general rule the occasional lost packet will not affect too drastically the quality of a call, but lose 5 in a row and an entire word is lost
and this will be a problem. So if you are going to result in considerable degradation of quality. Longhave lost packets make sure they are only lost in a delays in transit can affect quality so much that theregular distributed manner. 5% lost packets technology can become unusable, though manydistributed evenly will not result in the loss of words vendors do have solutions which aim to negate thelose 5% of the words by clustering the packets and degradation suffered due to transit delays.the effect is bad. While some standards have been set by the ITU, the technology is not fully standardized and5) Advantages and Weaknesses of VoIP: there is no guarantee that products from different vendors will be interoperable. Some vendors are trying to resolve this problem by forming groups and5.1 Advantages: making guarantees about the products in the group but this is only a partial solution - vendor’s out with There are many advantages to be gained from the group cannot guarantee interoperability.implementing an IP Telephony solution within the Heavy congestion on the network can result inorganization. The following list aims to highlight considerable degradation of service as IP is notsome of the advantages of such a strategy: good at providing Quos (Quality of Service) Single network infrastructure. When installing guarantees. Feedbacks to Lucent TechnologiesVoIP in the office only a single cable is required to customers reflect this worry. Major companies arethe desk, for both telephone and data. Eliminating planning to install IP Telephony capabilities at someseparate telephone wiring. point and have carried out initial investigations, VoIP uses "soft" switching which eliminates however:most of the legacy PBX equipment. Reducing the Since only one physical network for both data andcost of installing a communications infra-structure voice/fax transmissions is required, failure of theand the maintenance cost once installed. network could be catastrophic, as all communications Simple upgrade path. The VoIP PBX capabilities are lost.technology is software based. It is easier to expand,upgrade and maintain than its traditional telephony 6) Conclusion:counterparts. Bandwidth efficiency. VoIP can compress more Without a doubt, the data revolution will only gainvoice calls into available bandwidth than legacy momentum in the coming years, with more and moretelephony.. IP Telephony helps to eliminate wasted voice traffic moving onto data networks. Vendors ofbandwidth by not transporting the 60% of normal voice equipment will continue to develop integratedspeech which is silence voice and data devices based on packetized IP - the underlying protocol - is supported by technology. Users with ubiquitous voice and datamost platforms and is independent of the transport service integrated over one universal infrastructureprotocol used. will benefit from true, seamless, transparent Only one physical network is required to deal interworking between voice and all types of data.with both voice/fax and data traffic instead of twophysical networks. Having only one physical networkhas the following advantages: 7) References: Lower physical equipment cost, lowermaintenance costs. 1. Computer Networks by Andrew S.Tanenbaum5.2 Weaknesses: 2. Internetworking with TCP/IP by Douglas E.comer 3. www.iec.org.com While there are many aspects of VoIP which 4. www.telogy.comprovide considerable benefits, the technology is still 5. www.rad.comvery young and problems remain. The following 6. www.mailto:firstname.lastname@example.org looks at some of the weaknesses of thistechnology and their consequences. The Internet is not the best medium for realtime communications. Individual packets can takedifferent routes and varying delays can beencountered and packets lost in transit. Waiting fordelayed packets or retransmission of lost packets can