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VoIPVoice Over Internet ProtocolVoIPis short forVoiceoverInternetProtocol.

Voice over Internet Protocol is a category of hardware and software that enables people to use theInternetas the transmission medium for telephone calls by sending voice data in packets usingIPrather than by traditional circuit transmissions of thePSTN.

IP Network

Multimedia PC

Multimedia PCInitially, PC to PC voice calls over the Internet PSTN (DC)GatewayPSTN (NY)Gateways allow PCs to also reach phones or phones to reach phones Public Switched Telephone Network

Gateway3Circuit-Switched TelephonyTraditional PSTN ApproachSignaling NetworkClass 5SwitchTypically analog loop, conversion to digital at local switchCircuit-based TrunksClass 5SwitchClass 4Switch64 kb/s digital voiceMedia streamSignalingSCPMost service logic in local switches, rest in SCPsData travels over a parallel (but separate) networkPSTN vs. VoIPVoice network use circuit switching.

Dedicated path between calling and called party.

Bandwidth reserved in advance.

Cost is based on distance and time. Data network use packet switching.

No dedicated path between sender and receiver.

It acquires and releases bandwidth, as it needed.

Cost is not based on distance and time.How does it work?Compression voice is compressed typically with one of the following codecs, G7.11 64k, G7.29AB 8k, G723.1 6.3kEncapsulation the digitized voice is wrapped in an IP packetRouting the voice packet is routed thru the network to its final destinationVoIP protocols are:H.323 protocol suiteMedia Gateway Control Protocol (MGCP)Session Initiation Protocol(SIP)H.248(Media Gateway Control (Megaco))Real-time Transport Protocol(RTP)Real-time Transport Control Protocol(RTCP)Secure Real-time Transport Protocol(SRTP)Session Description Protocol(SDP)

Device Control Protocolslike H.248 (more popularly known as Megaco),

Media Gateway Control Protocol (MGCP), NCP, Real-time Transport Protocol (RTP)

Access Service Signalling protocolslike Session Initiation Protocol (SIP) and H.323

Network Service Signalling Protocolslike SIP, SIP-T, CMSS, BICC etc.

Call Agent SIP ServerService BrokerApplication ServerMedia ServerSignaling Gateway

Trunking gatewayAccess Gateway and subscriber gatewayAccess concentratorBandwidth ConcentratorBandwidth managerEdge routers

International Voice MarketCalls Terminated on PSTNSource: Telegeography 2010(2001 figures were projections)H.323 ArchitectureITU-T

H.323 ZoneH.323TerminalH.323 GatekeeperH.323GatewayPSTN

H.323Multipoint Control UnitTelco-centric multimedia, multiparty conferencing (initially for LANs)Gatekeeper for network control, heavy-weight protocolsWidely deployed in first wave of VoIP standardization

3 stages of signaling: RAS to Gatekeeper H.225 call signaling H.245 media stream control(can be simplified for VoIP)RAS(Registration/Admission/Status): defined as H.225 (RAS) protocol in the standard. RAS messages are originating by the gateway at the moment a terminal initiates a call. These messages are transported using UDP (user datagram protocol) to the gatekeeper.

H.225:also known as H.225.0 call control signaling. This protocol specifies the use and support of Q.931 signaling messages. This protocol is responsible for establishing and releasing connections.

H.245:used after establishment of the connection, between the two gateways, to negotiate parameters of the call such as codecs to be used, video or conference support, timer values, etc. Moreover, IP logical channel ports are exchanged for the RTP sessions and eventually the transmission of data and/or voice traffic.

RTPis the protocol used for transporting voice packets and it is managed by RTCP.

SIP (Session Initiation Protocol)The SIP consists of the following entities:User Agent Client (UAC): Caller application that initiates and sends SIP requests.User Agent Server (UAS):Receives and responds to SIP requests: accepts, redirects, or refuses calls.Proxy Server:Contacts one or more clients or next-hop servers and passes the call requests further. It contains UAC and UAS.Redirect Server:Does not initiate SIP requests or accept calls. Accepts SIP requests, maps the address into new addresses and returns those addresses to the client.

Methods of SIP are:INVITE:User or service is invited to participate in a session.ACK:Client has received a final response to an INVITE request.OPTIONS:Server being queried about capabilities.REGISTER:Client registers address with a SIP server.BYE:User Agent Client indicates to server to release the call.

Advantages of VOIP (Voice over Internet Protocol)

Decrease the cost of telephone calls in long distance.Multi-functionality of the communication link to the user.Alternatives to Internet telephony are DSL technology and ISDN technology.Disadvantages of VOIPQuality of service (Latency, Jitter and packet loss )Number portabilityEmergency callsFax supportPower requirementsSecurityCaller ID