voip lec.ppt

Download voip lec.ppt

Post on 21-Jul-2016

11 views

Category:

Documents

1 download

Embed Size (px)

DESCRIPTION

ppt

TRANSCRIPT

  • Introduction of VOIP

  • DefinitionsInternet Telephony:" In the beginning, Internet telephony simply meant the technology and techniques to let you make voice phone calls local, long distance, and international over the Internet using your PCthe definition of Internet telephony is broadening day by day to include all forms of media (voice, video, image), and all forms of messaging and all variations of speed from real-time to time-delayed.

    IP Telephony: As defined by Microsoft) IP Telephony is an emerging set of technologies that enables voice, data, and video collaboration over existing IP-based LANs, WANs and the Internet. Specifically, IP Telephony uses open IETF and ITU standards to move multimedia traffic over any network that uses IP (the Internet Protocol).

    Voice over IP (VoIP): The technology used to transmit voice conversations over a data network using the Internet Protocol. Such data network may be the Internet or a corporate Intranet, or managed networks typically used by long and local service traditional providers and ISPs that use VoIP.

  • What is VOIP?VoIP is a term used in IP telephony to describe a set of facilities for managing the delivery of voice information using the Internet Protocol.

    This means sending voice information in digital form in discrete packets rather than in the circuit committed protocols of the Public Switched Telephone Network (PSTN).

  • Understanding VOIPVoIP stands for Voice over the Internet Protocol and is also referred to as IP telephony. This technology enables users to :

    Make calls by moving packets of information over the Internet using broadband network connection to make phone calls to other VoIP and regular phones users rather than traditional public switched telephony circuits also known as Public Switched Telephone Network (PSTN) or analog telephones

  • VOIP OverviewIP Telephony is a process that enables the transfer of voice data over a packet switched network as opposed to the traditional circuit switched network.

    The transmission of voice packets over the Internet Protocol (IP) is known as VoIP and is implemented properly holds the promise of converged networks and unified communications.

  • VOIP NetworkMajor components of the VOIP network include:

    Signaling Protocols and Standards: Initiate and control communications of voice, video, and data. - H.323: Network, platform, and application independent standard that allows interoperability between H.323-compliant devices. - SIP: Protocol that is based upon request-response or INVITE model. SIP is used to establish conferencing, telephony, multimedia, and other Internet communication sessions. SIP uses a media-description language, such as Hypertext Transfer Protocol (HTTP). Media Gateway: - Converts one media stream to another; for example, converts voice packets to analog. - Can interact with call controllers, proxies, and soft switches via proprietary or standard protocols such as SIP.

  • H.323 Gateway: - Transforms audio received from a telephone device or telecommunications system into a format that the data network can use. - Acts as a bridge to the IP network. - Generally has built-in intelligence to select the voice compression Codec's and adjust the protocols and timing between two dissimilar computer systems or voice over data networks.

    Call Server: - Receives call setup request messages. - Determines the status of destination devices and checks the authorization of users to originate or receive calls. - Creates and sends the necessary messages to process the call requests.

  • Gatekeeper: - Provides call control, media access, and bandwidth management between endpoints. - Performs address translation, admissions control, bandwidth management, and zone control. - Coordinates access to other servers and manages call routing. - Receives requests from clients, determines the destination server that it needs to communicate with, and coordinates access with that server. - Maps destination telephone number to destination endpoint IP address. IP Terminals and Clients: - Include end points on the network, such as hard or soft telephones (portable or stationary) and wireless devices (802.11a and 802.11b). - Bring voice and data communications to the end user. A Call Server completes the call processing. IP Backbone Network: - Provides the universal communication language and foundation to allow dissimilar networks and equipment from a variety of vendors to interconnect.

  • H.323 Recommendation published by ITU Ties together a number of protocols to allow multimedia transmission through an unreliable packet-based network 1996: approved by ITU 2003: Version 5H.323 Architecture

  • H.323 Protocol Stack for VoIP

  • H.323 Protocol SuiteVideoAudioDataTransportH.261 H.263G.711 G.722 G.723.1 G.728 G.729T.122 T.124 T.125 T.126 T.127H.225 H.235 H.245 H.450.1 H.450.2 H.450.3 RTP X.224.0

  • G.7xx Speech (De)Coding H.323 systems must support G.711: PCM, 64kbps Other codecs: G.729, G.726,

    RTP Realtime Transport Protocol(RFC 3550, July 2003) Application layer protocol for transmitting realtimedata (audio, video, ...) Includes payload type identification, sequencenumbering, timestamping, delivery monitoring Mostly over UDP Supports multicast & unicast

  • Control Protocol - RTCP RTP Control Protocol (RFC 3550, July 2003) Periodic transmission of control packets to all participants in the session Main functions:- provide feedback on quality of data distribution- carries a persistent transport-level identifier for an RTP source (CNAME)- each participant sends control packets to all others which independently observe the number of participants

  • More Control Protocols in H.323 H.225 (RAS)- protocol between terminal and gatekeeper (if present)- allows terminals to join/leave zone, request/returnbandwidth, provide status updates, H.245 (Call Control)- Media Control Protocol- Allows terminals to negotiate connection parameters (codec, bit rate, ..) Q.931 (Call Signalling)- Manages call setup and termination

  • SIP Session Initiation Protocol Developed by IETF since 1999 RFC 2543, March 1999 (obsolete) RFC 3261, June 2002 Target: develop simpler and more modular protocol for VoIP than the large and complex H.323 by ITUSIP is a text-based protocol similar to HTTP and SMTP, for initiating interactive communication sessions between users SIP is an application-layer control (signalling) protocol for creating, modifying and terminating sessions with one or more participants Sessions include Internet Multimedia conferences, Internet Telephone calls and Multimedia distribution

  • SIP (2) SIP can be used with different transport protocols, it doesn't even require reliable transport protocols A simple SIP client can be implemented using only UDP

  • SIP (3)

  • Comparison of H.323 and SIP

  • Comparison of H.323 and SIP (2)

  • CODECsSelecting the appropriate speech CODEC is essential. CODEC performance includes the baseline quality (that is, without impairments) and the performance with impairments present, such as background noise and lost or late packets. The table below shows some CODECs that are used for voice traffic. Bandwidth requirements are estimates.

  • CODEC SelectionIt is important to select a CODEC that meets the bandwidth and voice quality requirements.

  • G.711 is the preferred choice when bandwidth and cost are not an issue and is generally the default CODEC for Local Area Networks (LANs) because G.711 does not compress the audio.

    G.729 A/B is generally the default CODEC for Wide Area Networks (WANs) because it requires less bandwidth than G.711 and delivers near-toll voice quality; for example, a G.729 A/B with a 30 ms sample size is an effective technique to reduce bandwidth limitations, while delivering acceptable voice quality.

    G.723.1 is used when bandwidth, not voice quality, is the customers objective.

  • ImportantSpeech CODECs (compression algorithms), such as G.729 A/B and G.723.1, are designed to reduce the bandwidth required; however, when using these CODECs, consider parameters such as end-to-end delay (latency) and distortion in voice quality. Although the G.726 CODEC has less processing delay than the G.729 A/B CODEC, G.729 A/B is generally used for VoIP because it delivers better voice quality.

    Tip: Remember, when selecting a CODEC, consider the customersvoice quality targets and bandwidth requirements. For example, theG.723.1 CODEC might be appropriate if bandwidth consumption, notvoice quality, is the primary consideration. However, if voice quality,not bandwidth consumption, is the primary consideration, anotherCODEC, such as G.711 is appropriate. If bandwidth consumption ANDvoice quality are important, consider G.729 A/B.

  • VoIP & QoS

    Voice quality characteristics- Clarity: fidelity, clearness, and intelligibility of signal- Delay: effect on interactivity- Echo: distracting and confusing

    Latency- Components: Encoding, Packetisation, Network delay,Receiver buffering, Decoding- ITU-TG.114 recommends 150msOne-way Delay Effect on perceived Quality

  • Jitter- Smoothed by playback buffers- Receivers adapt the depth of these buffersSudden changes in jitter may cause loss

    Figure: Playback bufferBandwith- Generally modest (64 kbps or less)- Depends on codec and use of silence suppressionCodec Rate (kbps)G.729 (A/B) 8G.722 48-64G.711 64Packet loss- Should be less then 5%