1. What is a converged network?
    It is a single, packet-based IP network supporting multiple services and carries voice, video and data traffic.
  2. What are Real-Time application?
    They are applications that need the data in each packet by a certain time.
  3. Give 2 examples of streaming video service.
    Video-On-Demand and Scheduled Multicasts are two types of streaming video service are commonly deployed.
  4. What is the real-time, two-way transmission of voice information over the Internet?
    Voice over IP 
  5. List the benefits of VoIP.
    •Reduce long-distance charges - By substituting fixed-rate Internet access charges for the per-minute/per-mile charges of switched phone service, VoIP can significantly reduce long-distance charges.

    •Cut phone costs using branch office connectivity - Companies who have already invested in connecting branch offices and other remote locations through an intranet, can piggyback voice traffic on top of these data networks to cut phone costs.

    •Integrated messaging - With Voice over IP, it is possible to have and integrated messaging solution that provides access to e-mail, voicemail, and faxes from a single source.

    •“Click to call” Sale and Customer service – Company can include a "click to call" feature on their Web site so that visitors can speak with a live representative directly through their computers.
  6. What are the 2 classes of protocols VoIP and video conferencing requires?
    • Singnalling Protocols:
    • Voip => H.323, SCCP, SIP & MGCP
    • Videocon. => H.323

    Transport protocol:  RTP/RTCP

    *Voice over IP and video conferencing require two classes of protocols: a signaling protocol that is used to set up, disconnect and control the calls and telephony features; and a transport protocol to carry data with real-time characteristics.
  7. What is ITU H.323?
    It stands for International Telecommunications Union (ITU) H.323, which defines an international standard for multimedia over IP.
  8. What protocol defines all aspects of call transmission, from call establishment to capabilities exchange to network resource availability?
    ITU H.323
  9. Describe RTP.
    RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services
  10. Describe RTCP.
    RTP control protocol (RTCP) provides monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTCP provides out-of-band control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself.
  11. What is Codecs, and list the benefits of it.
    Audio and Video Codecs define the format of audio and video information and represent the way audio and video are compressed and transmitted over the network.

    Codecs significantly reduce the bandwidth used for audio and video transmission over the network
  12. What are the factors to consider when evaluating whether a network is ready to support real-time application?
    They are bandwidth, packet loss, delay and jitter.
  13. How are call quality reported?
    Call quality was traditionally reported as a Mean Opinion Score (MOS) on a scale from 1-5 where 1 is the lowest and 5 the highest.
  14. What is bandwidth?
    Bandwidth (usually expressed in kilobits per second) describes the rated throughput capacity of a given medium, protocol, or connection. The bandwidth of a path determines the rate at which information can be send through a channel.
  15. What is packet delay?
    Packet delay (usually in milliseconds) is the time it takes for a packet to cross a network connection from sender to receiver.
  16. What is forwarding delay?
    It is the time it takes for a device to receive a packet, make a forwarding decision and start transmitting a packet through an uncongested port.
  17. What is queuing delay?
    It is the time the packet waits in a buffer for its turn to be transmitted. It depends on the serialization delay for the packets served ahead, the dimension of the buffers, the amount of congestion, and the configuration of the router or switch scheduling policies.
  18. What is serialization delay?
    It is the time it takes for a device to clock a packet at the given output rate. Serialization delay depends on the link’s bandwidth as well as the size of the packet being clocked.
  19. What is propagation delay?
    It is the time it takes for a transmitted bit to get from the transmitter to a link’s receiver. This delay is a function of the distance and the media but not of the bandwidth.
  20. Formula for calculating serialization delay at an interface.
    Packet Size(bits) / Link Speed (bps)
  21. What is Packet Jitter?
    Also called inter-packet delay, it is the variation in packet delay due to buffering of packets in network devices.
  22. What is Packet Loss?
    It is generally specified as a percentage of packets lost while transmitting a certain number of packets over some time interval.

    Packet loss is caused by buffer exhaustion at network congestion points, network downtime and corrupted packets on the transmission wire.
  23. What causes congestion?
    Congestion occurs when the rate at which traffic is directed to an interface exceeds the rate at which the interface can forward the traffic.
  24. How to calculate total delay when there is no congestion?
    Total Delay = Minimum Delay of the network = Forwarding delay + Serialization Delay + Propagation Delay.
  25. How to calculate total delay when congestion occurs?
    Total Delay = Forwarding delay + Queuing delay + Serialization delay + Propagation delay
  26. What happens when congestion persists?
    The software queue may become full.

    The network element will start to drop new arriving packet (tail drop) because there is no room in the queue for it.This may result in packet loss if the application does not do a packet retransmission.
  27. What are the steps for capacity planning?
    Calculate or measure the minimum bandwidth requirements for each major application.

    Sum of step 1 = minimum bandwidth requirement for the link 

    Link capacity = 4/3 * step 2
  28. The Min B/W Requirement for Interactive Video (Video conferencing).
    For capacity planning, it is recommended to allow for a 20% overhead for the H.323 signaling traffic on top of the media (audio, video, and T.120 data).
  29. What is Point to Point & Multi Point calls.
    Video conferencing calls can be between two parties (a point-to-point call) or involving more then two parties (a multi-point call).
  30. What does the bandwidth requirement per call for Voice-over-IP bearer traffic depends on?
    It depends on the codec, sampling rate (packetization interval) and the Layer 2 protocol.
  31. What other link optimization techniques are there?
    There is the use of TCP optimization, application-specific optimization, compression and caching.

    TCP and application-specific optimization are techniques to overcome the design limitations of TCP or a specific application that cause poor performance in WAN environments.

    Compression and caching reduce redundant and unnecessary data travel across the link. 
  32. What is achieved with the implementation of Quality of Service?
    The network will have an ability to provide improved service to selected network traffic over others.

    It enables the network to control and predictably service a variety of networked applications and traffic types.
  33. List the benefits of QoS.
    Control over resources being used.

    Provide service differentiation to users or customers.

    have co-existence of mission-critical applications, voice and video applications on a single WAN link.
Card Set
Network Management Chapter 6