voice

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Author:
Tjr31
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42855
Filename:
voice
Updated:
2010-10-19 03:04:25
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voice
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ch 7 to 13
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  1. what is delay or latency
    time it takes for speech to exit the speaker's mouth and reach the listener's ear.
  2. what is propagation delay
    length a signal must travel via light in fiber or electrical impulse in copper-based networks.
  3. what is handling delay
    • is caused by devices that forward the frame through the network
    • Includes queuing delay for congested networks.
  4. what is serialization delay
    amount of time it takes to actually place a bit or byte onto an interface
  5. What is one way delay for voip and total for satelite
    150 ms 500ms
  6. what is jitter
    • is a variation of packet interval time and only exist isn packet based networks
    • the difference between when the packet is expected and when it is actually received is jitter
  7. what is Packetizing Pulse Code Modulation
    sends two 10-ms G.729 speech frames in every packet.
  8. what is Packetizing Pulse Code Modulation
    the 54-byte header, multiple voice samples can be packed into a single Ethernet frame to transmit

    can increase the voice delay
  9. what is voice compression
    Two basic variations of 64 Kbps PCM are commonly used: µ-law and a-law
  10. adaptive differential pulse code modulation (ADPCM)
    encodes using 4-bit samples, giving a transmission rate of 32 Kbps
  11. waveform codecs
    compression techniques that exploit redundant characteristics of the waveform itself
  12. source codecs
    techniques that are grouped together
  13. g.711 voice coding
    .726
    .728
    .729
    .723.1
    • 64kbps
    • 40,32, 24 and 16
    • 16 low delay
    • 8 stream
    • 5.3 and 6.3
  14. iLBC (Internet Low Bitrate Codec)
    free speech codec suitable for robust voice communication over IP. The codec is designed for narrow band speech
  15. Mean Opinion Score (MOS)
    compare how well a particular codec works under varying circumstances, including differing background noise
  16. Perceptual Speech Quality Measurement
    developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.
  17. Echo
    developed to "hear" impairments caused by compression and decompression and not packet loss or jitter.
  18. Echo has two drawbacks
    loud and and long
  19. qos helps with what
    packetloss, 20 ms of speach is average
  20. g.729 rule of thumb
    5 percent of packet loss is average
  21. Voice Activity Detection (VAD)This fixed amount of time is what
    hangover and is typically 200 ms.
  22. detecting when speech begins
    beginning of a sentence is cut off or clipped or front end speach clipping
  23. Digital-to-Analog Conversion
    plague toll network
  24. Tandom Encoding
    The network is designed to put all the dial-plan information in the central-site PBX.
  25. how to avoid Tandem Compression
    simplifer the router configuration and use ios multimedia conference manager
  26. what are transport protocols
    RTP/UDP/IP
  27. what kind of traffic does RTP send
    transmitting delay-sensitive traffic across packet-based networks
  28. RTP consists of
    data part and a control part, the latter called RTP Control Protocol (RTCP).
  29. Reliable User Data Protocol (RUDP)
    • connectionless UDP protocol
    • enables reliability without the need for a connection-based protocol such as TCP
  30. Dial-Plan Design
    example of joining disparate networks is when two companies merge.
  31. Authentication
    provides a vehicle to identify
  32. Authorization
    • sets the process of determining whether the client is
    • allowed to perform or request certain tasks or operations
  33. Accounting
    process of measuring resource consumption
  34. Remote Authentication Dial-In User Service (RADIUS)
    • data-communications protocol designed to provide security management and statistics collection in
    • remote computing environments

    • •Transactions between the client
    • and RADIUS server are authenticated through the use of a shared secret, which is never sent over
    • the network.

    • User passwords are sent encrypted between
    • the client and RADIUS server
  35. Vendor-Specific Attributes (VSA)
    Each protocol has its own set of features and information fields
  36. call detail records (CDR)
    are the standards for every provider to offer billing-related information.
  37. Automatic Messaging Accounting (AMA
    • include typical informational elements such as calling number, called number, connect time and
    • date, call duration, and service characteristics.
  38. The VoIP network generates
    CDRs that
    contain data about which extension made or received calls to or from which number and for how long.
  39. records stored in
    plain text log files
  40. Prepaid Billing Applications
    • access to user funds before the call is important so that you can see how much cash is available for
    • the call to be made.
  41. 2 voip challenges
    volume and mixed ussage and billing records
  42. Mediation Services
    • •collects, correlates, and aggregates the
    • accounting messages generated by the various VoIP-enabled network elements
    • involved in a call.

    • –It converts these into standard
    • or proprietary CDR formats, such that one and only one CDR is generated for
    • each call.
  43. Security layering
    multiple technologies
  44. Confidentiality
    • a third party should not be able to read the
    • data that is intended for the recipient
  45. Integrity
    • recipient should receive the packets that the originator sends without any change to their content. A
    • third party should be unable to modify the packets in transit.
  46. Authenticity
    sender and recipient of VoIP signaling
  47. Availability
    protection from Denial-of-Service (DoS) attacks
  48. Shared Key
    Users share a single secret key

    • Symmetrical. Same key used to
    • encrypt and decrypt
  49. Public-Key cryptography
    Each user has a related public and private key

    Asymmetrical. Different key used to decrypt than was used to encrypt.
  50. Digital signature
    hash is created using the original message
  51. Certificates
    method to distribute public keys
  52. certificate authority (CA)
    issues a certificate validating the requestor’s identity and public key
  53. TLS
    –Evolved from SSL

    • –Typically used to secure
    • signaling

    –Sits on top of TCP
  54. TLS record protocol
    The lower-level layer that provides connection security and is the workhorse.

    • –It provides
    • privacy and integrity.
  55. IPSec 2 different modes
    • Transport mode—only the
    • payload of an IP datagram is protected.
    • Tunnel mode—, the entire IP packet is
    • protected.
  56. SRTP
    • provides integrity, authenticity, and privacy protection to the RTP traffic and to the control
    • traffic for RTP,
    • SRTP does not specify how the keys are exchanged between the sender and recipient
  57. Disabling Unused Ports/Services
    disable these unused services or ports for VoIP devices and IP infrastructure devices
  58. HIPS
    secure critical voice devices such as call processing elements.
  59. DHCP Snooping
    as a firewall between untrusted sources that send
  60. IP Source Guard
    All IP traffic on an untrusted port is blocked except for DHCP messages.
  61. Dynamic ARP Inspection (DAI)
    Can rate limit ARP requests to prevent flooding/DoS attacks.
  62. CAM Overflow and Port Security
    • Configuring a maximum number of MAC addresses per port. If a particular port encounters this limit,
    • the specified action is taken on that port.
  63. BPDU Guard and Root Guard
    Prevents malicious devices from sending STP BPDUs.
  64. NIPS
    • monitor and analyze network traffic to detect intrusion
    • Can be deployed on that VoIP side
    • and on the data side of the IP network.
  65. Transitive trust
    trust that is transmitted through another party.
  66. latest version of h.323
    h.323v5
  67. Terminals
    • Also called endpoints, terminals provide
    • point-to-point and multipoint conferencing for audio and, optionally, video and data
  68. Gateways
    • interconnect to Public Switched Telephone Network(PSTN) or ISDN networks for H.323 endpoint
    • interworking
  69. Gatekeepers
    provide admission control and address translation services for terminals or gateways.
  70. Multipoint control units (MCU)
    • Devices that allow two or more
    • terminals or gateways to conference with either audio and/or video sessions.
  71. Call Control Signaling
    Uses the Gatekeeper Routed Call Signaling (GKRCS) model
  72. Proxy Server
    • At the application layer
    • Can manage QoS for a terminal that doesn’t support RSVP
    • Can route H.323 traffic separate from data
    • using application-specific routing (ASR)
  73. H.323 Protocols
    most H.323 implementations today utilize TCP
  74. Registration, Admissions and Status (RAS) Signaling
    Provides pre-call control in H.323 gatekeeper-based networks.
  75. Call control signaling
    Used to connect, maintain, and disconnect calls between endpoints.
  76. RAS Signaling
    provides pre-call control
  77. Gatekeeper Discover
    • multicast address is 224.0.1 .41
    • Method for endpoints to determine which
    • gatekeeper to register with
  78. Gatekeeper Request (GRQ)
    multicast message sent by an endpoint looking for the gatekeeper
  79. Gatekeeper Confirm (GCF)
    The reply to an endpoint GRQ indicating the transport address of the gatekeeper’s RAS channel.
  80. Gatekeeper Reject (GRJ)
    gatekeeper does not want to accept its registration.
  81. Bandwidth Control
    • gatekeeper currently looks only at its static bandwidth table to determine whether to accept or reject the
    • bandwidth request.
  82. Call Control Signaling
    A reliable call control channel is created across an IP network on TCP port 1720
  83. Direct Endpoint Call Signaling
    Call signaling messages are sent directly between two endpoints
  84. GKRCS
    Call signaling messages are routed through a gatekeeper
  85. Session Initiation Protocol (SIP)
    • controls the initiation, modification, and termination of interactive multimedia sessions.
    • is a peer-to-peer protocol
  86. User Location
    • discover the location of the end user for the purpose of establishing a session or delivering a SIP
    • request
  87. User Capabilities
    enables the determination of the media capabilities
  88. User Availability
    determination of the willingness of the end user to engagein communication.
  89. Session Setup
    parameters for the parties who are involved in the session.
  90. Session Handling
    the modification, transfer, and termination of an active session
  91. User Agent (UA)
    logical function in the SIP network that initiates or responds to SIP transactions
  92. User Agent Client (UAC)
    A logical function that initiates SIP requests and accepts SIP responses
  93. User Agent Server (UAS)
    accepts SIP requests and sends back SIP responses
  94. Proxy
    forwarding SIP requests to the target UAS or another proxy on behalf of the UAC.
  95. Redirect Server
    that generates 300 class SIP responses to requests it receives
  96. Registrar
    • accepts SIP REGISTER requests and updates the information from the request message into a location
    • database.
  97. Back-to-back user agent (B2BUA)
    entity that processes incoming SIP requests as a UAS.
  98. DNS
    • resolve host or domainnames into routable IP addresses. DNS can also be used to load-share across
    • multiple servers in a cluster identified by a hostname.
  99. Session Description Protocol (SDP
    describe the parameters of the multimedia session.
  100. Realtime Transport Protocol (RTP)
    transports real-time data such as audio or video packets to the endpoints
  101. RSVP
    reserve network resources such as bandwidth prior to establishment of the media session
  102. TLS
    provide privacy and integrity of SIP signaling information over the network
  103. STUN
    discover the presence any type of Network Address Translation (NAT) between them and the public Internet.
  104. SIP Addresses
    • identify a user or a resource within a network domain
    • SIP URI sip:user@domain:portsip:user@host:port
  105. Address-of-Record (AOR).
    globally routable and points to a domain whose location service can map the AOR to another SIP URI,
  106. SIP Request Messages
    are sent from client to server to invoke a SIP operation
  107. INVITE
    recipient user or service is invited to participate in a session
  108. ACK
    ACK request confirms that the UAC has received the final response to an INVITE request
  109. OPTIONS
    capable of delivering a session to the user
  110. BYE
    the termination of a previously established session
  111. CANCEL
    enables UACs and network servers to cancel an in-progress request, such as INVITE
  112. REGISTER
    register its current location information corresponding to the AOR of the user with SIP servers.
  113. SIP Response Messages
    Sent from server to client
  114. Dialogs
    The establishment of a session also results in a SIP signaling relationship between the peers
  115. transaction
    the establishment, modification, or termination of a media session.
  116. Signaling SIP transactions
    • connection-oriented transport layer protocols such as TCP or Stream Control Transmission Protocol
    • (SCTP)
    • connectionless protocols such as UDP.
  117. B2BUA (Back-to-back user
    agent)
    • providing centralized call control and feature management in SIP networks.
    • initiate new SIP calls and modify and terminate existing calls.
  118. User Agent Discovering SIP Servers in a Network
    UA needs the IP address of the registrar or proxy server to register and provide SIP service.
  119. SIP Registration Process
    SIP endpoints register with a SIP registrar server.
  120. SIP Proxies
    are elements that route SIP requests to the UAS and SIP responses to the UAC.
  121. SIP Extensions
    that do not support the newer extensions, SIP defines the extension negotiation mechanism.
  122. SUBSCRIBE
    A SIP entity acts as a subscriber when it sends a SUBSCRIBE for a specific event type, such as message summary, to a SIP entity that the Request URI identifies.
  123. NOTIFY
    The UAS that is processing the SUBSCRIBE request acts as the notifier
  124. Presence
    enables users to publish their availability status and display messages or icons as a form of self-expression

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