VOIP and its Application
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VOIP and its Application

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  • VoIP is the ability to make telephone calls and send faxes over IP-based data networks with a suitable quality of service and superior cost/benefit. Significant advances in technology have been made over the past few years that enable the transmission of voice traffic over traditional public networks such as Frame Relay (Voice over Frame Relay) as well as Voice over the Internet through the efforts of the Voice over IP Forum and the Internet Engineering Task Force (IETF). Additionally, the support of Asynchronous Transfer Mode (ATM) for different traffic types and the ATM Forum's recent completion of the Voice and Telephony over ATM specification will quicken the availability of industry-standard solutions. Alliance Datacom has new and refurbished Voice Over IP equipment. Let our Network Engineers help your compnay design and build a quality IP Network. Contact our team today. IP and Data Services from AT&T and Alliance Datacom. Get a quote on data service by filling out our fast VOIP form . Justifications for development of VoIP can be summarized as follows: Cost reduction. There can be a real savings in long distance telephone costs which is extremely important to most companies, particularly those with international markets. Simplification. An integrated voice/data network allows more standardization and reduces total equipment needs. Consolidation. The ability to eliminate points of failure, consolidate accounting systems and combine operations is obviously more efficient. Advanced Applications. The long run benefits of VoIP include support for multimedia and multiservice applications, something which today's telephone system can't compete with Frame Relay, IP and ATM are known as packet or cell switching technologies. This is in contrast to the public telephone network, which is a circuit switching technology, designed to carry voice transmissions. Frame Relay and IP insert data into variable-sized frames or packets. ATM chops data into small cells, which facilitates fast switching of data through the network.
  • 2.1 – Equipments developers and manufactures see a windows of opportunity to inovate and compete. They are busy developing new VoIP-enabled equipments attempting to break into narket in time 2.2 – Internet service providers see the possibility of competing with the PSTN for customers. 2.3 – Users are interested in the integration of voice and data applications in addition to the cost savings.
  • Gatekeeper: Manage a zone. Collection of H323 Devices). Gateway: Interoperability between different networks. Signaling converter MCU: Multipoint controller Unit: Conferencing UAC: User agent Client. Application that initiates and send send SIP request UAS: Receive and respond to sip requests on behalf of clients SIP terminal: provide and support two way communication with another sip entity. Contain UAC Proxy: Redirect server: map address into more new address and return those address to the client Location server: Provide information about caller possible location to redirect and proxy server
  • When the number is received from the PBX, the router compares the number to those mapped in the routing table. If a match is found, the call is routed to the IP host. After the connection is established, the corporate intranet connection is transparent to the subscriber.
  • Solution for processing delay: encapsulate several small packet into a single large frame. Process call concatenation Echo: Echo cancellation technique
  • Within the router the dial plan mapper maps the dialed digits to an IP address and signals a Q.931 call establishment request to the remote peer that is indicated by the IP address. Meanwhile, the control channel is used to set up the Real-Time Control Protocol (RTCP) audio streams, and the Resource Reservation Protocol (RSVP) is used to request a guaranteed QoS. When the remote router receives the Q.931 call request, it signals a line seizure to the PBX. After the PBX acknowledges, the router forwards the dialed digits to the PBX and signals a call acknowledgment to the originating router. In connectionless network architectures such as IP, the responsibility for session establishment and signaling is with the end stations. To successfully emulate voice services across an IP network, enhancements to the signaling stacks are required. For example, an 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 teardown between H.323 agents or end stations. RTCP is used to establish the audio channels themselves. A reliable session-oriented protocol, Transmission Control Protocol (TCP), is deployed between end stations to carry the signaling channels. Real-Time Transport Protocol (RTP), which is built on top of User Datagram Protocol (UDP), is used for transport of 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.
  • Packet voice appears to a network as data; thus it can be transported over networks normally reserved for data, where costs are often far less than in voice networks. Packet voice uses less transmission bandwidth than conventional voice, so more can be carried on a given connection. Whereas telephony requires as much as 64,000 bits per second (bps), packet voice often needs less than 10,000 bps. For many companies, there is sufficient reserve capacity on national and international data networks to transport considerable voice traffic, making voice essentially free.
  • 1.1 - In such applications, it is normally expected that some of the calls transported on the packet voice network will have originated in the public phone network. Such outside calling over packet voice is uniformly tolerated in a regulatory sense, on the basis that the calls are from employees, customers, or suppliers and represent the company’s business 1.2 - the application is uniformly tolerated in a regulatory sense. In such a situation, an outside call placed from a public phone network in one country and terminated in a company site within another via packet voice may be a technical violation of national monopolies or treaties on long-distance service. Where such a call is between company employees or between employees and suppliers or customers, such a technical violation is unlikely to attract official notice

VOIP and its Application VOIP and its Application Presentation Transcript

  • Voice over IP By : Adiel AKPLOGAN CAF E Infor matique S. A. - TOGO E-mail: adiel @ akplogan .net - Web: http://www. akplogan .net AFNOG 2001 Accra – Ghana 12 Mai 2001
  • Topics
    • Definition
    • Standard
    • Requirement
    • QoS
    • Application
    • CAFEnet S.A. (Togo) exemple
  • Introduction
    • Concern about reduction of communication coast. Voice, Fax and Data integration become a challenge and priority for many network manager.
    • Telecommunication deregulation and Multi-service access network development allow this challenge to be achieve.
    • Today voice over packet network (ATM, Frame Relay and IP) is the one most growing aspect of Multi-service access network
  • VoIP System Model
    • Voice and signaling transmission using packet network (IP)
    • ITU standard for signaling Model
      • H.323 ( ITU )- Packet-based Multimedia communication system.
    • SIP (IETF) – Session initiation protocol
    • The voice system resulting from these two standard follow a common Model
  • VoIP System Model
    • A packet voice system like VoIP follow this model.
      • A packet network (Internet in most case)
      • Voice agent at edge of the cloud. Convert voice information from it traditional telephony form to a suitable form for packet transmission.
      • the packet are send to another voice agent at the call destination.
  • VoIP system model
  • VoIP System Model
    • Two issue from the previous diagram
      • Voice coding: Voice  Packet, Packet  Voice
      • Signaling: who is called and where is the called party on the network?
  • The two models in brief UAC (user agent Client) UAS (User agent server) SIP Terminal Proxy Redirect Server Location server Gatekeeper Gateway H.323 terminal MCU SIP H.323
  • Architecture
  • Communication
  • VoIP Addressing
    • As Internet communication is based on IP address (layer 2), every equipment that want to communicate should have an IP address. For VoiP, the voice interface appear as additional IP host.
    • T ranslation of dial digits from the PBX to an IP host address is performed by the dial plan mapper. The destination telephone number, or some portion of the number, is mapped to the destination IP address.
  • Voice packet routing and Delay
    • VoIP take advantage of all strong and sophisticate routing protocols of IP (including best route calculation…)
    • But due to the way IP packets are send through the network, the big challenge is the Latency or delay that cause echo and talker overlap.
  • Voice packet routing and Delay
    • Source of delay include:
      • Accumulation delay: caused by need to collect a frame of voice samples to be processed by the voice coder (from some microsecond to many milliseconds.
      • Algorithmic Delay: caused by specific voice encoding delay.
      • Processing delay: result from the two previous delay plus collecting the sample in to packet for transmission.
      • Network delay: Processing that occurs as packets are sent across a network .(from protocol, medium, and buffer use to remove packet jitter on the receive side)
  • Voice packet routing and Delay
      • Echo: is generated toward the packet network from the telephone network…as it always greater than 50 ms, it is not acceptable for good audition.
      • Jitter: is the variable inter-packet timing cause by the fact that packets do not all cross the network at the same speed.
      • Lost packet : Under peak load and congestion, voice frame are dropped at the same rate as data frame.
  • VoIP Signaling
    • Three distinct area:
      • Signaling from PABX  router
        • Network seize the PABZ with any of the signaling used to seize a trunk (as the local network appear to the BABX as a trunk) – FXS or E&M.
      • Signaling from Router  router
        • Dial plan Mapper.
      • Signaling from Router  PABX
        • Line seizure signaling.
  • VoIP application
    • In today’s networking, there are several attractive alternatives both to conventional public telephony and to leased lines. Among the most interesting are networking technologies based on a different kind of voice transmission, called packet voice and in our case Voice over IP .
    • VoIP can be used in two broad context differentiated by geography or by the type of users to be served.
  • VoIP application
    • Within a national administration or telephony jurisdiction,
      • to support its own voice calling among its own sites .
      • to support the activities of a single company — to connect two or more company locations in multiple countries —
      • to connect public calls within a company , the packet voice provider is technically providing a local or national telephone service and is subject to regulation as such.
  • VoIP application
    • Between different administration or telephony jurisdiction
      • to connect public calls between countries , the packet voice provider is subject to the national regulations in the countries involved and also to any treaty provisions for international calling to which any of the countries served are signatories.
    • Be aware of the valid law applicable in your country until setting up VoIP application to avoid any inconvenience.
  • VoIP application
    • Example:
    • CAFE Informatique & Telecommunications S.A.
      • Two application:
        • International Communication
        • Call center
          • 30 local worker.
          • Tele-marketing for America and Canadian company
          • Data scramble for foreign company
  • VoIP application
  • VoIP
    • Question?