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Voice over IP Foundations

Price: $2,995.00

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Voice over IP is rapidly moving from a tactical, cost-saving effort to a more long-term strategy of productivity improvements and reduced cost of network ownership. The ability to service voice communications needs over your existing data networks is now a reality. You can maximize your savings through the critical elements of proper evaluation and design. This course provides real-world, multi-vendor options for integrating voice and data communication applications. You will analyze cost versus call quality issues and understand the key standards and technologies that make VoIP a reality. In our intensive hands-on labs, you will evaluate public Internet calling, bandwidth considerations, echo control, jitter, voice compression, softswitch function, and more. Whether you are considering fully deploying VoIP or implementing IP Telephony in a hybrid approach, this course will help you understand the options you face, the problems you will encounter, and the security and performance issues you'll need to consider. This course will help you prepare for the Convergence Technologies Professional (CTP) certification.

Course Overview

By the end of the class students will be able to:

  • Core concepts of how IP (Internet Protocol) carries a VoIP packet
  • The benefits and capabilities of SIP (Session Initiation Protocol)
  • How to implement RTP and QoS to ensure the highest voice quality over your IP networks
  • The essentials of signaling and when to use SIP, MEGACO, H.323, or MGCP
  • Understand how SIP establishes, modifies and terminates "sessions" over IP networks
  • The importance of QoS (jitter, packet delay, packet loss)
  • Compare IP, ATM, and Frame Relay voice/data networks
  • Decipher the call setup procedure under the H.323 standard
  • Protocol flows and how to analyze code negotiation using network trace tools
  • How a VoIP gatekeeper acts as a virtual telephony switch
  • Security issues to consider when setting up your VoIP
  • The missing pieces of VoIP implementation, including signaling, call accounting, and billing
  • The effect of line jitter, call latency, compression, and sockets
  • Evaluate PC-to-PC, PC-to-phone, and phone-to-phone calls

Prerequisites

No prerequisites require for this course.

Course Outline

  • Voice over IP Foundations

    • VoIP Fundamentals
    • Key architectural VoIP components
    • End-to-end voice transmission
    • Packetizing voice (encapsulation)
    • Transmission time allocation
    • QoS and capacity considerations
    • Sources of delay
    • Coder processing delay (Think Time)
    • Algorithmic delay (Look Ahead)
    • Packetization delay
    • Serialization delay
    • Queuing delay
    • Jitter buffer function
    • VoIP QoS requirements: Packet Loss, Latency, Jitter
  • VoIP in the LAN

    • MAC address
    • IP address and ARP
    • Ethernet switching
    • Logical and physical segmentation
    • VLAN - 802.1Q/P
    • 802.3af - Power over Ethernet (POE)
  • IP Networking

    • IP addressing
    • Static routing
    • OSPF
    • EIGRP
  • TCP and UDP

    • Transmission Control Protocol (TCP)
    • VoIP protocols that use TCP
    • User Datagram Protocol (UDP)
    • VoIP protocols that use UDP
  • IP Services

    • DNS
    • How SIP uses DNS
    • DHCP
    • How IP telephony uses DHCP
  • Voice Encoding and Compression

    • G.711 u-law and A-law
    • G.729
    • G.723.1
  • Real-Time Transport Protocol (RTP)

    • Dealing with Packet Loss, Latency, Jitter
    • How various protocols defines the RTP session
      • Session Description Protocol
      • H.245 Terminal Capabilities
      • The RTP profile
      • The RTP payload type field
      • RTP telephony events (RFC 2833)
    • How RTP removes jitter
    • How RTP handles packet loss
    • How RTP identifies the talking party
    • How RTP handles silence suppression
    • How RTP is used to mix voice (conference calls)
    • The RTP header
    • RTP Control Protocol (RTCP)
      • SDES
      • Sender/receiver reports
      • Bye reports
  • SIP Architecture

    • SIP architecture
      • Proxy: stateful, stateless, call stateful, Session Border Controller
    • SIP methods: INVITE, ACK, BYE, CANCEL, REGISTER, INFO, PRACK, etc.)
    • SIP response codes: 1xx, 2xx, 3xx, 4xx, 5xx, 6xx
    • SIP headers (To:, From:, Call-ID:, Allows: Required, Via)
    • Session Description Protocol (SDP)
    • SIP Addressing, Session Control, and Call Setup
  • SIP Uniform Resource Indicators (URIs)

    • Understand the format of SIP URIs and how URIs interoperate with PSTN dialing plans, email systems, and web pages
    • Generic URI information (RFC 2396)
    • Direct or Proxy
    • PSTN number (RFC 2808)
    • Instant messaging
    • Presence
    • In registrations
  • SIP Call Flow Examples Review how SIP calls are set up for applications like PSTN, instant messaging, VoIP, and more in this technical, in-depth analysis of the protocol.

    • Call attempt - unsuccessfulli>
    • Presence subscription
    • Registration
    • Presence notification
    • Instant Message Exchange
    • Call setup - successful
    • Cancel
    • Vacant number
    • 100rel
    • www authenticate
  • Media Gateway Control Protocol (MGCP)

    • Architecture
    • Verbs: CRCX, MDCX
    • Responses
    • Packages (DTMF, Line, Trunk, Generic, etc.)
    • Parameter lines
    • Sample call flow protocol analysis
  • H.323

    • ASN.1 primer
    • H.323 architecture
      • Gatekeeper
      • Gateway
      • MCU
      • Terminal
    • H.323 versions
    • H.323 gatekeeper controlled call flow example
  • Queuing

    • Priority queuing
    • Weighted fair queuing
    • Weighted precedence
    • Traffic policing and traffic shaping
    • Low latency queuing
    • The effects of data traffic and fair queuing on VoIP
    • Mixing voice and data traffic effectively
    • Determining bandwidth needs for voice traffic
    • Assessing the impact of voice on data networks
      • Low speed links
      • High speed links

    QoS Related Networking Protocols

    • Differentiated Services (DiffServ)
    • Call Admission Control