Sheet no.(27)VoIPVoice Over IP

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Sheet no.(27)IP TelephonyInstead of using traditional circuit switch systems for voice communications, IP Telephony uses a packet protocol originally designed for data communications.

Circuit Switched - PSTNPacket Switched Data Network

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Sheet no.(27)DefinitionIP TelephonyTransmission of voice, fax, and related services over packet-switched IP- based networks.Internet TelephonySpecific sub-set of IP TelephonyIP Telephony in which the principal transmission network is the public Internet. Voice-over-the-Net(VoN) ; Internet Phone ; Net TelephonyVoice-over-IP (VoIP)Specific sub-set of IP Telephony IP Telephony, in which the principal transmission network(s) is (are) private, managed IP-based network(s).Voice-over-frame relay ; Voice-over-cable ; Voive-over-DSL (VoDSL)

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Sheet no.(27)Models of IP TelephonyThree mains models to use IP Telephony

PC-to-PC over IP.PC-to-Phone over IP.Phone-to-Phone over IP.

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Sheet no.(27)PC-to-PCUSER AUSER BServer

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Sheet no.(27)PC-to-PhoneUSER AUSER BUSER BGatewayIP Telephony Provider

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Sheet no.(27)Phone-to-Phone (1)USER AUSER BUSER BGatewayNetwork of IP Telephony Service ProviderUSER BUSER AGatewayManagement IP Network

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Sheet no.(27)Phone-to-Phone (2)USER AUSER BUSER AServer

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Sheet no.(27)The IP Network Problem Congestion continues to grow over the Internet Traffic expands = or > bandwidth Best-effort" performance dictated by the very design of the Internet Protocol (IP)

IP Telephony is a Mission critical applications, require prioritization.

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Sheet no.(27)IP Telephony: QoS

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Sheet no.(27)QoS: DelaysTLoss PacketT"#T#TInversionDelay Variation : T#T JitterIP NetworkNetwork DelaySenderNetworkReceiverSender Delay:Coding delayPacketing delayTransmission delay

Receiver Delay:Decoding delayDePacketing delayReceiver delay

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Sheet no.(27)QoS : Delay ComponentsOne of the most important performance measures of a data network is the average delay required to deliver a packet from origin to destination. Furthermore delay considerations strongly influence the selection of the networking algorithms and their performance, such as routing and flow control. The packet delay within the communication subnet ( network layer) is the sum of the delays on each subnet link traversed by the packet. Each delay may in turn consists of four components:Processing Delay : The delay between the time the packet is correctly received at the head of the node and the time packet is assigned to an outgoing link queue for transmission.Queuing Delay :The delay between the time packet is assigned to an outgoing queue for transmission and the time it starts being transmitted.Transmission Delay: Delay between times that the first and last bits of the packet are transmitted.Propagation Delay : Delay between the time the last bit is transmitted at the head node of the link and the time last bit is received at the tail node. The propagation delay is independent of the amount of the traffic. It is dependent on the physical characteristics of the media.The Processing delay is independent of the traffic. It is basically a function of the processor and architecture of the node.

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Sheet no.(27)QoS TechnologiesReservationAllocates resources on a per-flow basisFlows include information such as transport protocol, source address & port, destination address and port IntServ/RSVPPrioritizationTraffic flows are aggregated and categorized by "class of serviceDiffServ and MPLS.

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Sheet no.(27)DiffServ: Differentiated ServicesDefined in RFCs 2474, 2475Creates classes of service for traffic flows with different priorities:Aggregates large numbers of individual flows at the edge of the network into small numbers of aggregated flows through the core of the network.Flows are marked at network edge in the IPv4 ToS field (DS field). Services applied through the core.

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Sheet no.(27)IP Telephony ProtocolsH.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.SIP supports TCP and UDP.SIP, H.323 and MGCP

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Sheet no.(27)SIP: Session Initiation ProtocolSession Initiation Protocol - An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users.IETF RFC 2543 Session Initiation Protocol

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Sheet no.(27)SIP Distributed Architecture

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Sheet no.(27)SIP MessagesSIP Methods:INVITE Initiates a call by inviting user to participate in session.ACK - Confirms that the client has received a final response to an INVITE request.BYE - Indicates termination of the call.CANCEL - Cancels a pending request.REGISTER Registers the user agent.OPTIONS Used to query the capabilities of a server.INFO Used to carry out-of-bound information, such as DTMF digits.SIP components communicate by exchanging SIP messages:

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Sheet no.(27)SIP HeadersSIP borrows much of the syntax and semantics from HTTP.A SIP messages looks like an HTTP message message formatting, header and MIME supportSIP Addressing The SIP address is identified by a SIP URL, in the format: user@host.

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Sheet no.(27)SIP: Communication EstablishmentEstablishing communication using SIP usually occurs in six steps:Registering, initiating and locating the user.Determine the media to use involves delivering a description of the session that the user is invited to.Determine the willingness of the called party to communicate the called party must send a response message to indicate willingness to communicate accept or reject.Call setup.Call modification or handling (eg call transfer (optional)).Call termination.

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Sheet no.(27)SIP: Registering

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Sheet no.(27)Simplified SIP Call Setup

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Sheet no.(27)IP Telephony Signaling Protocols: H.323ITU-T Recommendation H.323 Version 4Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications.

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Sheet no.(27)H.323 ComponentsGatewayMultipoint Control UnitGatekeeperTerminal

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Sheet no.(27)H.323 : Communication EstablishmentEstablishing communication using H.323 may occur in five steps:Call setup.Initial communication and capabilities exchange.Audio/video communication establishment.Call services.Call termination.

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Sheet no.(27)Simplified H.323 Call SetupNote: This diagram only illustrates a simple point-to-point call setup where call signaling is not routed to the gatekeeper. Refer to the H.323 recommendation for more call setup scenarios.Both endpoints have previously registered with the gatekeeper.Terminal A initiate the call to the gatekeeper. (RAS messages are exchanged).The gatekeeper provides information for Terminal A to contact Terminal B.Terminal A sends a SETUP message to Terminal B.Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission.Terminal B sends a Alerting and Connect message.Terminal B and A exchange H.245 messages to determine master slave, terminal capabilities, and open logical channels.The two terminals establish RTP media paths.

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