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Voip

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Voip

  1. 1. INDEPENDENT SEMINAR ON BROADBANDDRONACHARYA COLLEGE OF ENGINEERING GURGAON SUBMITTED BY:- MOHIT ARORA 10191 E.C.E-1(b) 1
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  3. 3. Table of ContentsOverview of VoIP………………………………………………………………………...4VoIP Components:Terminals ............................................................................................................................5Gateways.............................................................................................................................6Gatekeepers.........................................................................................................................7Multipoint control unit……………………………………………………………………7VoIP Protocols:H-323 .................................................................................................................................8Session initiation protocol……………………………………………………………......10VoIP Signaling and routing …………………………………………………………....12Benefits and requirements of VoIP….………………………………………………....15Conclusion………………………………………………………………………………17References………………………………………………………………………………18 3
  4. 4. VOICE OVER INTERNET PROTOCOLOVERVIEW OF VoIPVoice over IP is the transport of voice using the Internet Protocol (IP) however this broad termhides a multitude of deployments and functionality. So voice over Internet Protocol is a methodfor taking analog audio signals and turning them into digital data that can be transmitted over theInternet.The following types of VoIP applications are in use: ATAThe simplest and most common way is through the use of a device called an ATA (analogtelephone adaptor). The ATA allows you to connect a standard phone to your computer or yourInternet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes theanalog signal from your traditional phone and converts it into digital data for transmission overthe Internet. IP PhonesThese specialized phones look just like normal phones with a handset, cradle and buttons. Butinstead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernetconnector. IP phones connect directly to your router and have all the hardware and softwarenecessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to makeVoIP calls from any Wi-Fi hot spot. Computer-to-computerThis is certainly the easiest way to use VoIP. You dont even have to pay for long-distance calls.There are several companies offering free or very low-cost software that you can use for this typeof VoIP. All you need is the software, a microphone, speakers, a sound card and an Internetconnection. 4
  5. 5. VoIP COMPONENTSTERMINALS IP PhonesAn IP phone uses Voice over IP technologies allowing telephone calls to be made over an IPnetwork such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet,or a private IP Network such as that of a company. The phones use control protocols such asSession Initiation Protocol, Skinny Client Control Protocol or one of various proprietaryprotocols such as that used by Skype. IP phones can be simple software-based Soft phones orpurpose-built hardware devices that appear much like an ordinary telephone or a cordless phone.Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA). H.323protocol and provide real-time, two-way multimedia communications. In the case of voice, theH.323 terminal is generally an IP telephone. Analog PhonesA telephone can be a basic push-button wall unit or an integrated system complete withanswering machine, stored-number dial, speaker phone, and 900MHz cordless operation. ComputersWith VoIP software such as Skype, yahoo, netmeeting and many more running on computersthey can also be used as communication devices. H.323 is also widely deployed on PCs. A verycommon application of the H.323 protocol can be found in the Microsoft NetMeeting softwarethat allows for both voice and video transmissions on a user’s PC. 5
  6. 6. GatewaysGateways work as a translator to allow communications between H.323 and non-H.323 entities(for instance, between H.323 terminals and telephones on the circuit-switched network). H.323gateways provide a means for an H.323 network to communicate to other networks, mosttypically the PSTN or PBX systems. In order to provide this interoperability, gateways providefor translation and call control functions between the two dissimilar network types. Encoding,protocol, and call control mappings occur in gateways between two endpoints. Gateways providemany functions, including: Translating protocols The gateway acts as an “interpreter,” allowing the PSTN and the H.323 network to talk to each other to set up and tear down calls. Signaling Gateway The Signaling Gateway is located in the service provider’s network and acts as a gateway between the call agent signaling and the SS7-based PSTN. It can also be used as a signaling gateway between different packet based carrier domains. It may provide signaling translation, for example between SIP and SS7 or simply signaling transport conversion e.g. SS7 over IP to SS7 over TDM. Trunking Gateway The Trunking Gateway is located in the service provider’s network and as a gateway between the carrier IP network and the TDM (Time Division Multiplexing)-based PSTN. It provides transcoding from the packet based voice, VoIP onto a TDM network. Typically, it is under the control of the Call Agent / Media Gateway Controller through a device control protocol such as H.248 (Megaco) or MGCP. 6
  7. 7. Access Gateway The Access Gateway is located in the service provider’s network. It provides support for POTS phones and typically, it is under the control of the Call Agent / Media Gateway Controller through a device control protocol such as H.248 (Megaco) or MGCP. Subscriber Gateway The Subscriber Gateway is located at the customer premises and terminates the WAN (Wide Area Network) link (DSL, T1, fixed wireless, cable etc) at the customer premises and typically provides both voice ports and data connectivity. Usually, it uses a device control protocol, such as H.248 (Megaco) or MGCP/NCS, under the control of the Call Agent. It provides similar function to the Access Gateway but typically supports many fewer voice ports.GatekeepersGatekeepers provide call control functions such as address translation and bandwidthmanagement and are often considered to be the most important component in the H.323 stack.Gatekeepers in H.323 networks are optional. However, if they are present, it is mandatory thatendpoints use their services. The H.323 standards define several mandatory services that thegatekeeper must provide and specify other optional functionality.Multipoint Control UnitsMCUs provide conference facilities for users who want to conference three or more endpointstogether. MCUs provide a unique function to the H.323 protocol in that they do not provide adirect interconnection to the H.323 protocol stack. Rather, they provide a method for H.323 tointerconnect voice and videoconferencing. MCUs provide conference support for three or moreendpoints. All terminals participating in the conference establish a connection with the MCU. Itmanages conference resources and negotiations between endpoints to determine which audio orvideo codec to use. 7
  8. 8. VoIP PROTOCOLSH.323H.323 is probably the most important standard supporting packetized voice technology. H.323 isan ITU-T recommendation umbrella set of standards that defines the components, protocols, andprocedures necessary to provide multimedia (audio, video, and data) communications over IP-based networks. Essentially, H.323 provides a method to enable other H.32X-compliant productsto communicate. In addition to control and call setup standards, H.323 encompasses protocols foraudio, video, and data as follows: Audio The compression algorithms H.323 supports for audio are all proven International Telecommunications Union (ITU) standards (G.711, G.723, and G.729). Because audio is the minimum service provided by the H.323 standard, all H.323 terminals must have support for at least one audio codec support, as specified by G.711. Video Video capabilities for H.323 are optional. However, any video enabled H.323 terminal must support the ITU-T H.261 encoding and decoding recommendation. Data H.323 references the T.120 specifications for data conferencing. An ITU standard, T.120 addresses point-to-point and multipoint data conferences. It provides interoperability at the application, network, and transport levels.The H.323 Protocol StackJust as with the TCP/IP protocol, the H.323 protocol is actually a suite of protocols that worktogether to provide end-to-end call functionality in a converged network. However, the H.323 8
  9. 9. protocol also relies heavily on the services provided by other protocols such as TCP, IP, andUDP as well as RTP. The protocols that make up the H.323 protocol are Registration,Admission, and Status (RAS), H.245, and H.225.CodecsCoder/decoders (codecs) are used by not only the H.323 protocol but by all VoIP protocols todefine the degree of compression and decompression algorithms that will be used to transporteither a voice or video transmission across a converged network.Speech codecs, sometimes called voice encoders or vocoders if source codecs areused, can be divided into three basic classes: waveform, source, and hybrid. Waveform codec These are older, operationally used high bit rates and provide very good quality speech reproduction. Source codec These operate at very low bit rates but tend to produce speech that sounds artificial or tinny. Hybrid codec These use techniques from both source and waveform coding, operate at intermediate bit rates, and provide good-quality speech. Fig No 8.1 - The H.323 Protocol Stack 9
  10. 10. Session Initiation ProtocolSIP is a simple signaling protocol used for Internet conferencing and telephony. SIP is fullydefined in RFC 2543. Based on the Simple Mail Transport Protocol (SMTP) and the HypertextTransfer Protocol (HTTP), SIP was developed within the IETF Multiparty Multimedia SessionControl (MMUSIC) working group. SIP specifies procedures for telephony and multimediaconferencing over the Internet. SIP is an application-layer protocol independent of theunderlying packet protocol (TCP, UDP, ATM, X.25). SIP is based on a client/server architecturein which the client initiates the calls and the servers answer the calls. Because it is an openstandard based protocol, SIP is widely supported and is not dependent on a single vendor’sequipment or implementation. However because of its simplicity, scalability, modularity, andease with which it integrates with other applications, this protocol is attractive for use inpacketized voice architectures.Some of the key features that SIP offers are: Address resolution, name mapping, and call redirection Dynamic discovery of endpoint media capabilities by use of the Session Description Protocol (SDP) Dynamic discovery of endpoint availability Session origination and management between host and endpoints SIP has learned from HTTP and SMTP and has built a rich set of extensibility and compatibility functions. SIP was designed to be highly modular. A key feature is its independent use of protocols. SIP has the capability to integrate with the Web, e-mail, streaming media applications, and other protocols..syngress.comSession Initiation Protocol ComponentsThe SIP system contains two components: User agents Network servers 10
  11. 11. A user agent (UA) is SIP’s endpoint, which makes and receives SIP calls. The client is called theuser agent client (UAC) and is used to initiate SIP requests.The server is called the user agent server (UAS), receiving the requests from the UAC andreturning responses for the user. There are three kinds of SIP servers: Proxy server Proxy servers decide to which server the request should be forwarded and then forward the request. The request can actually traverse many SIP servers before reaching its destination. The response then traverses in the reverse order. A proxy server can act as both a client and server and can issue requests and responses. Redirect server Unlike the proxy server, the redirect server does not forward requests to other servers. Instead, it notifies the calling party of the actual location of destination. Registrar server Provides registration services for UACs for their current locations. Registrar servers are often placed with proxy and redirect servers. Fig No 8.2 - SIP Components 11
  12. 12. VoIP SIGNALING AND ROUTINGIn telephony, the signaling information is used to exchange information between endpoints on anetwork to set up, control, and end calls. The signaling method thats used depends on the type ofdevice thats being used and the type of signaling method thats used by the telephone company.On the PSTN local loop An open circuit with no current flowing indicates an on-hook condition (telephone handset placed in the cradle). Offhook (telephone receiver off the cradle) is indicated by a closed circuit with current continuously flowing. DP and DTMF are the address-signaling methods implemented from telephone to switch in the telephone network. Earth and magnet (E&M) signaling is the most commonly utilized method of analog trunking.VoIP SignalingIn connectionless network architectures such as IP networks, the responsibility for sessionestablishment and signaling resides in the end stations. To successfully emulate voice servicesacross an IP network, enhancements to the signaling stacks are required. Some are: H.323 agent is added to the router for standards-based support of the audio and signaling streams. The Q.931 protocol is used for call establishment and tear-down between H.323 agents or end stations. Real-Time Control Protocol (RTCP) provides for reliable information transfer once the audio stream has been established. A reliable session-oriented protocol such as TCP is deployed between end stations to carry the signaling channels. 12
  13. 13. RTP, which is built on top of UDP, is used to transport the real-time audio stream. RTP uses UDP as a transport mechanism because it has lower delay than TCP and because actual voice traffic, unlike data traffic or signaling, tolerates low levels of loss and cannot effectively exploit retransmission. H.245 control signaling is used to negotiate channel usage and capabilities. H.245 provides for capabilities exchange between endpoints so that codecs and other parameters related to the call are agreed upon between the endpoints. It is within H.245 that the audio channel is negotiated.wyngress.coVoIP signaling is most commonly used in three distinct areas: signaling from the PBX to the router signaling between routers signaling from the router to the PBX.Signaling Between Routers and PBXsWhen signaling from PBX to router, the user picks up the handset, signaling an off-hookcondition. The connection between the PBX and router appears as a trunk line to the PBX, whichsignals the router to seize the trunk. Once a trunk is seized, the PBX forwards the dialed digits tothe router in the same manner the digits would be forwarded to a telephone company switch oranother PBX. The signaling interface from the PBX to the router may be any of the commonsignaling methods used to seize a trunk line, such as FXS, FXO, E&M, or T1/E1 signaling. ThePBX then forwards the dialed digits to the router in the same manner as the digits would beforwarded to a telco switch. The PBX seizes a trunk line to the router and forwards the dialeddigits. Within the router, the dial plan mapper maps the dialed digits to an IP address andinitiates a Q.931 call establishment request to the remote peer router that is indicated by an IPaddress. 13
  14. 14. Fig No 9.1- PBX-to-Router Signaling Fig No 9.2 - Router-to-Router SignalingWhen the remote router receives the Q.931 call request, it signals a line seizure to the PBX. Afterthe PBX acknowledges this seizure, the router forwards the dialed digits to the PBX and signalsa call acknowledgment to the originating router. Fig No 9.3 - Router-to-PBX signaling 14
  15. 15. BENEFITS AND REQUIREMENTS FOR VoIPFor service providers examining the business case for VoIP, the ubiquity of IP as a networkingtechnology at the customer premises opens the possibility of deploying a vast range of innovativeconverged voice and data services that simply cannot be cost effectively supported over today’sPSTN infrastructure. IP-based internet applications, such as email and unified messaging, may be seamlessly integrated with voice application IP centrex services allow network operators to provide companies with cost effective replacements for their ageing PBX infrastructure VoIP services can be expanded to support multimedia applications, opening up the possibility of cost effective video conferencing, video streaming, gaming or other multi- media applications. The flexibility of next generation platforms allows for the rapid development of new services and development cycles are typically shorter than for ATM or TDM-based equipment. VoIP products based on the MSF architecture, unlike legacy TDM switches, often support open service creation environments that allow third party developers to invent and deliver differentiated services. VoIP leverages data network capacity removing the requirement to operate separate voice and data networks. IP equipment is typically faster and cheaper than ATM or TDM-based equipment – a gap that is increasing rapidly every few months. Re-routing of IP networks (e.g. with MPLS) is much cheaper than, say, SDH protection switching.Whatever the justifications, most service providers recognize that VoIP is the direction of thefuture – however when looking at a future PSTN scale solution service providers must ensurethat the following key requirements are met to provide equivalence with the PSTN: Security Quality of Service 15
  16. 16. Reliability Migration path OSS support Billing Network InterconnectionThese issues are by no means simple and in many cases have delayed roll out of VoIP services.This white paper will look in more detail at these problems and consider at a high level how theymight be addressed. 16
  17. 17. CONCLUSIONVoice over IP is quickly becoming readily available across much of the world, however manyproblems still remain. For the time being transmission networks involve too much latency ordrop too many packets, this effects quality of service sometimes severely deteriorating thequality of the call. Also VOIP contains many security risks, sending out packets that any personmay intercept. Although VOIP may offer cheaper solutions for many the PSTN offers a highQoS and greater security that makes up for its higher prices. It is my belief that the telephonemarket will continue to be dominated by the PSTN until quality of service and security issuescan be addressed. 17
  18. 18. REFRENCES 1. www.wikipedia.org 2. www.britanica.com 3. www.computer.howstuffworks.com 4. Data Communications and Networking by Behrouz A. Forouzan 18

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