760 Chapter 10: LAN QoS
Q&A
As mentioned in the Introduction, you have two choices for review questions. The questions that follow next give you a more difficult challenge than the exam itself by using an open-ended question format. By reviewing now with this more difficult question format, you can exercise your memory better, and prove your conceptual and factual knowledge of this chapter. You can find the answers to these questions in Appendix A.
The second option for practice questions is to use the CD included with this book. It includes a testing engine and more than 200 multiple-choice questions. You should use this CD nearer to the end of your preparation, for practice with the actual exam format. You can even customize the CD exam to include, or not include, the topics that are only on the CCIP QoS.
LAN QoS Concepts
1What is instantaneous buffer overrun?
2What is meant by the term “1p1q4t?”
3What packet or frame marking(s) can a Layer 2 switch use to prioritize traffic?
4What packet or frame marking(s) can a Layer 3 switch use to prioritize traffic?
5What must be done for a Layer 2 switch to properly classify a packet received from a host across the WAN?
6For a Layer 2 switch to receive a CoS value from a router, what must exist?
7What type of router interface supports an 802.1Q trunk to a Layer 3 switch?
8What is a drop threshold?
9What is a trust boundary, and where should it be set?
Catalyst 6500 Series of Switches
10Name three of the five fields that the Policy Feature Card (PFC) rewrites to perform Layer 3 switching.
11Explain why the first packet in a flow may not retain the proper CoS values when a Catalyst 6500 is configured with a PFC.
12What is the difference between Hybrid mode and Native mode?
13What command is used on a Catalyst 6500 running in Hybrid mode to place ports 2/1 through 2/10 in VLAN 10?
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14What command enables QoS on a 6500 running in Native mode?
15What queue and threshold is recommended for call control traffic on a Catalyst 6500?
16Which series of Ethernet line cards are preferred in the Catalyst 6500 series for QoS? Why?
Catalyst 4500/4000 Series of Switches
17Which supervisor engine is preferred on the Catalyst 4500/4000 series? Why?
18In which queue number does the priority queue reside on a Catalyst 4500 or 4000 with a Supervisor II Engine?
19In which queue number does the priority queue reside on a Catalyst 4500 or 4000 with a Supervisor III Engine?
20What keyword(s) is used to enable the priority queue on a Catalyst 4500 or 4000 with a Supervisor III module?
21Name the four methods a Catalyst 4500 with a Supervisor III uses to classify traffic.
22What command enables you to rewrite a the CoS value of a PC attached to an IP Phone with Catalyst IOS switches?
23What trust state are the ports if a 4500 with a Supervisor III in when QoS is enabled?
24What is dynamic buffer limiting?
Catalyst 3550/3524 Series of Switches
25In which queue number does the priority queue reside on a Catalyst 3550?
26What command enables QoS on the Catalyst 3550?
27What trust state are the ports of a 3550 in when QoS is enabled?
28Name the four methods a Catalyst 3550 can use to classify traffic.
29What command enables QoS on the Catalyst 3524?
30When is a GigaStack GBIC module an acceptable QoS choice for cascading Catalyst 3524 switches together?
A P P E N D I X A
Answers to the “Do I Know This Already?” Quizzes and Q&A Sections
Chapter 1
“Do I Know This Already?” Quiz
QoS: Tuning Bandwidth, Delay, Jitter, and Loss
1List the four traffic characteristics that QoS tools can affect.
Bandwidth, delay, jitter, and loss are the four characteristics that affect QoS tools.
2Describe some of the characteristics of video traffic when no QoS is applied in a network.
Picture displays erratically; picture shows jerky movements; audio not in sync with video; movements slow down; video stream stops.
3Define bandwidth. Compare and contrast bandwidth concepts over point-to-point links versus Frame Relay.
Bandwidth refers to the number of bits per second that can reasonably be expected to be successfully delivered across a network. With point-to-point networks, bandwidth is equal to the speed of the link—the clock rate. With Frame Relay, the actual bandwidth is difficult to define. Typically, the minimum bandwidth equals the committed information rate (CIR) of a virtual circuit (VC). However, engineers at the provider and at the customer typically expect more than CIR to get through the network. The maximum bandwidth would be bounded by the slower of the two access rates on the access links.
4List the categories of delay that could be experienced by all three types of traffic: data, voice, and video.
Serialization, propagation, queuing, forwarding/processing, shaping, network. Note that coder/decoder (codec), packetization, and de-jitter delays are unique to voice and video, so technically these delays should not have been part of your answer for this question.
764 Appendix A: Answers to the “Do I Know This Already?” Quizzes and Q&A Sections
5Define jitter. Give an example that shows a packet without jitter, followed by a packet with jitter.
Jitter measures the change in delay experienced by consecutive packets. For instance, if a PC sends four packets one after the other, practically at the same time, say 1 ms apart, so the departure times are T=0, T=1, T=2, and T=3. The packets arrive at T=70, T=71, T=80, T=81, respectively. The second packet was sent 1 ms after the first, and arrived 1 ms after the first packet—so no jitter was experienced. However, the third packet arrived 9 ms after the second packet, after being sent 1 ms after the second packet—so 8 ms of jitter was experienced.
6Define packet loss, and describe the primary reason for loss for which QoS tools can help.
Packet loss means that a packet, which has entered the network, does not get delivered to the endpoint—it is lost in transit. Routers and switches drop packets for many reasons. However, QoS tools can affect the behavior of loss when packets will be lost due to queues being too full. When a queue is full, and another packet needs to be added to the queue, tail drop occurs.
Traffic Characteristics of Voice, Video, and Data
7Describe the contents of an IP packet carrying the payload for a G.729 Voice over IP (VoIP) call.
The IP packet contains an IP header, a UDP header, an RTP header, and the voice payload. With G.729, the payload uses 20 bytes, with an 8-byte UDP header, and a 12-byte RTP header. The IP header is 20 bytes long.
8Describe the amount of bandwidth required for G.711 and G.729 VoIP calls, ignoring data-link header/trailer overhead.
G.711 consumes 64 kbps for payload, for an 80-kbps stream with IP, UDP, and RTP headers. G.729 consumes 8 kbps for payload, plus another 16 kbps for IP, UDP, and RTP headers, for a total of 24 kbps.
9Define the meaning of the term “packetization delay” in relation to a voice call.
Voice must be converted from sound waves to analog electrical signals, and finally to digital signals, and then placed into a packet. Before 20 ms of voice digital payload can be placed into a packet, the speaker must speak for 20 ms. Packetization delay refers to the (default) 20 ms of delay—waiting for the speaker to speak long enough to fill the packet with the correctly sized payload.
10Define the term “codec delay” and discuss the two components when using a G.729 codec.
Voice calls incur codec delay when the codec converts the analog signal into digital voice payload. Every codec requires some time to process the incoming signal, which adds delay. With G.729, because it is predictive, it must also wait for some additional
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incoming voice to arrive, because it is algorithm-processing the voice sample to be encoded, plus a part of the next sample that will be encoded. The delay waiting for the additional voice is called “look-ahead” delay.
11Describe a typical video payload flow in terms of packet sizes and packet rates.
Video payloads use variable-length packets. The packet rates are also typically variable.
12Contrast the QoS characteristics needed by interactive data applications to the QoS needs of voice payload flows.
Answering such a question requires one to understand that QoS requirements for data applications are more subjective than those for voice. Generally, interactive data wants consistent delay (low jitter); relative to voice, however, more jitter is tolerable. Bandwidth demands vary greatly for data applications, whereas a single voice call uses a static amount of bandwidth. Delay for interactive data can be relatively longer than for voice, but the key measurement for data is application response time, which includes round-trip packet delays. Finally, data applications are much more tolerant of packet loss—because either the application will resend the data, or rely on TCP to resend the data, or just not care if some data is lost.
Q&A
1List the four traffic characteristics that QoS tools can affect.
Bandwidth, delay, jitter, and loss.
2Describe some of the characteristics of voice traffic when no QoS is applied in a network.
Voice is hard to understand; voice breaks up, sounds choppy; calls are disconnected; large delays make it difficult to know when the other caller has finished talking.
3Describe some of the characteristics of video traffic when no QoS is applied in a network.
Picture displays erratically; picture shows jerky movements; audio not in sync with video; movements slow down; video stream stops.
4Describe some of the characteristics of data traffic when no QoS is applied in a network.
Data arrives too late to be useful; erratic response times cause users to stop using application; customer care agents waiting on screen refresh, so customer waits.
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10List the categories of delay that could be experienced by all three types of traffic: data, voice, and video.
Serialization, propagation, queuing, forwarding/processing, shaping, network. Note that codec, packetization, and de-jitter delays are unique to voice and video, so technically these delays should not have been part of your answer for this question.
11Define, compare, and contrast serialization and propagation delay.
Serialization delay defines the time it takes to encode a frame onto the physical link. For instance, on a point-to-point link of 56 kbps, a bit is encoded every 1/56,000 seconds; therefore, a frame that is 1000 bits long takes 1000/56000 seconds to encode on the link. So, serialization delay is a function of link speed and length of the frame.
Propagation delay defines the time taken for a single bit to be delivered across some physical medium, and is based solely on the length of the physical link, and the speed of energy across that medium. If that same point-to-point link were 1000 km (approximately 620 miles) in length, the propagation delay would be 1,000,000m/2.1 * 10^8 ms, or 4.8 milliseconds.
12Define network delay.
Network delay refers to the delay incurred by a packet inside a packet network, like ATM, Frame Relay, or MPLS networks. Because the customer does not know the details of these networks, and because many customers’ packets share the carrier network, variable delays occur.
13List the QoS tool types that affect delay and give a brief explanation of why each tool can affect delay.
Queuing, link fragmentation and interleaving, compression, and traffic shaping. Queuing methods use an algorithm to choose from which queue to take the next packet for transmission, which can decrease delay for some packets and increase delay for others. LFI tools break large frames into smaller frames, so that smaller delay-sensitive frames can be sent after the first short fragment, instead of having to wait for the entire large original frame to be sent. Compression helps delay because it reduces the overall load in the network, reducing congestion, reducing queue lengths, and reducing serialization delays. Finally, traffic shaping actually increases delay, but it can be applied for one type of traffic, allowing other traffic to be sent with less delay.
14Define jitter. Give an example that shows a packet without jitter, followed by a packet with jitter.
Jitter measures the change in delay experienced by consecutive packets. If a PC sends four packets one after the other, practically at the same time, say 1 ms apart, so the departure times are T=0, T=1, T=2, and T=3, for instance, packets arrive at T=70, T=71, T=80, T=81, respectively. The second packet was sent 1 ms after the first,
768 Appendix A: Answers to the “Do I Know This Already?” Quizzes and Q&A Sections
and arrived 1 ms after the first packet—so no jitter was experienced. However, the third packet arrived 9 ms after the second packet, after being sent 1 ms after the second packet—so 8 ms of jitter was experienced.
15List the QoS tool types that affect jitter and give a brief explanation of why each tool can affect jitter.
Queuing, link fragmentation and interleaving, compression, and traffic shaping. These same QoS tools can be used for addressing delay issues. Queuing can always be used to service a jitter-sensitive queue first if packets are waiting, which greatly reduces delay and jitter. LFI decreases jitter by removing the chance that a jittersensitive packet will be waiting behind a very large packet. Compression helps by reducing overall delay, which has a net effect of reducing jitter. Traffic shaping may actually increase jitter, so it should be used with care—but if shaping is applied to jitter-insensitive traffic only, jitter-sensitive traffic will actually have lower delays and jitter.
16Define packet loss and describe the primary reason for loss for which QoS tools can help.
Packet loss means that a packet, which has entered the network, does not get delivered to the endpoint—it is lost in transit. Routers and switches drop packets for many reasons. However, QoS tools can affect the behavior of loss when packets will be lost due to queues being too full. When a queue is full, and another packet needs to be added to the queue, tail drop occurs.
17List the QoS tool types that affect loss and give a brief explanation of why each tool can affect loss.
Queuing and RED. Queuing tools allow definition of a longer or shorter maximum queue length; the longer the queue, the less likely that drops will occur. Also by placing traffic into different queues, more variable traffic may experience more loss, because those queues will be more likely to fill. RED tools preemptively discard packets before queues fill, hoping to get some TCP connections to slow down, which reduces the overall load in the network—which shortens queues, reducing the likelihood of packet loss.
18Describe the contents of an IP packet carrying the payload for a G.729 VoIP call.
The IP packet contains an IP header, a UDP header, an RTP header, and the voice payload. With G.729, the payload uses 20 bytes, with an 8-byte UDP header, and a 12-byte RTP header. The IP header is 20 bytes long.
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19Describe the amount of bandwidth required for G.711 and G.729 VoIP calls, ignoring data-link header/trailer overhead.
G.711 consumes 64 kbps for payload, for an 80-kbps stream with IP, UDP, and RTP headers. G.729 consumes 8 kbps for payload, plus another 16 kbps for IP, UDP, and RTP headers, for a total of 24 kbps.
20List the delay components that voice calls experience, but which data-only flows do not experience.
Codec delay, packetization delay, and de-jitter initial playout delay.
21Define the meaning of the term “packetization delay” in relation to a voice call.
Voice must be converted from sound waves to analog electrical signals, and finally to digital signals, and then placed into a packet. Before 20 ms of voice digital payload can be placed into a packet, the speaker must speak for 20 ms. Packetization delay refers to the (default) 20 ms of delay, waiting for the speaker to speak long enough to fill the packet with the correctly sized payload.
22List the different one-way delay budgets as suggested by Cisco and the ITU.
The ITU in document G.114 suggests a budget of up to 150 ms for quality voice calls; Cisco suggests a delay budget of up to 200 ms one-way if you cannot meet the 150-ms goal.
23Define the term “codec delay” and discuss the two components when using a G.729 codec.
Voice calls incur codec delay when the codec converts the analog signal into digital voice payload. Every codec requires some time to process the incoming signal, which adds delay. With G.729, because it is predictive, it must also wait for some additional incoming voice to arrive, because it is algorithm-processing the voice sample to be encoded, plus a part of the next sample that will be encoded. The delay waiting for the additional voice is called “look-ahead” delay.
24Describe the affects of a single lost packet versus two consecutive lost packets, for a G.729 voice call.
Lost voice packets result in the receiver having a period of silence corresponding the length of voice payload inside the lost packet(s). With two consecutive G.729 packets lost, 40 ms of voice is lost; the G.729 codec cannot predict and generate replacement signals when more than 30 ms of consecutive voice is lost. A single lost G.729 packet would only cause a 20-ms break in the voice, which could be regenerated. So, a single lost packet is not perceived as loss in a G.729 call.
25Describe a typical video payload flow in terms of packet sizes and packet rates.
Video payloads use variable-length packets. The packet rates are also typically variable.