Qos Quiz

Jitter, Delay, and Packet-loss are all examples of metrics for Quality of Service (QoS). Jitter refers to the variation in the delay of received packets, which can cause disruptions in real-time applications. Delay is the time it takes for packets to travel from the source to the destination, and it is important to minimize delay for efficient communication. Packet-loss is the percentage of packets that are lost during transmission, and it is crucial to keep this value low for reliable data transfer. Therefore, these three metrics are commonly used to measure and monitor the performance of a network's QoS.

QoS (Quality of Service) is not solely useful only if VOIP (Voice over IP) is used in the network. QoS is a network management technique that prioritizes certain types of network traffic to ensure a consistent and reliable user experience. While VOIP can greatly benefit from QoS to maintain voice quality and minimize latency, other applications such as video streaming, online gaming, and real-time data transfer can also benefit from QoS. Therefore, QoS is not exclusively dependent on the presence of VOIP in the network.

UDP (User Datagram Protocol) is the correct answer because it is commonly used for real-time applications such as voice traffic. UDP is a connectionless protocol that provides low-latency and fast communication, making it suitable for time-sensitive data like voice packets. Unlike TCP, UDP does not provide error checking or retransmission of lost packets, but for voice traffic, where a small delay is acceptable, UDP's speed and simplicity make it a preferred choice. XNS, TCP, and HTTP are not specifically designed for real-time communication and are less commonly used for voice traffic.

802.1X is a protocol used for network access control and authentication, not for Quality of Service (QoS) management. QoS is typically achieved using mechanisms such as DSCP bits and MPLS EXP bits, which prioritize and classify network traffic. 802.1P is also used for QoS, as it defines the priority levels for Ethernet frames.

Quality of Service (QoS) can alleviate the problem of dropped voice calls during periods of high network utilization. QoS ensures that voice traffic is given priority over other types of data traffic, such as file downloads or web browsing, in order to maintain a consistent and reliable connection for voice calls. By prioritizing voice traffic, QoS helps to minimize packet loss and latency, which are the main causes of dropped calls. This ensures that voice calls remain stable and clear even during times of high network congestion or utilization.

The MPLS header uses 3 bits for QoS. These bits are used to prioritize different types of traffic in a network. By assigning different levels of priority to packets, the network can ensure that high-priority traffic, such as voice or video data, is given preferential treatment and is delivered with minimal delay or loss. Using 3 bits allows for 8 different levels of priority to be assigned, providing enough granularity to effectively manage QoS in the network.

Packet classification is a QoS function performed on the incoming interface. It involves categorizing packets into different classes based on specific criteria such as source or destination IP address, protocol type, or port number. This classification allows for the application of different QoS policies and treatments to different classes of packets, ensuring that each packet receives the appropriate level of service based on its classification.

Traffic policing mechanisms use the token bucket mathematical model to meter traffic. In this model, tokens are placed in a bucket at a fixed rate. Each packet of traffic requires a certain number of tokens to be sent. If there are enough tokens in the bucket, the packet is allowed to be sent. If there are not enough tokens, the packet is either dropped or marked for a lower priority. This model helps in controlling the rate of traffic and preventing congestion in a network.

The three QoS requirements for voice are latency less than 150 ms, jitter less than 30 ms, and loss less than 1 percent. These requirements are necessary for ensuring a high-quality voice communication experience. Latency refers to the delay in transmitting voice packets, and a latency of less than 150 ms ensures real-time communication. Jitter is the variation in latency, and a jitter of less than 30 ms ensures consistent voice quality. Loss refers to the percentage of voice packets that are not successfully delivered, and a loss of less than 1 percent ensures minimal disruptions in the voice communication.

The correct answer is Best Effort and Expedited Forwarding. These two options are defined forwarding classes. Best Effort refers to the default forwarding class that provides no specific guarantees for packet delivery. Expedited Forwarding, on the other hand, is a forwarding class that ensures low latency and low loss for certain types of traffic, such as voice or video data. Both of these forwarding classes are commonly used in network communication to prioritize and manage different types of traffic.

ICMP Echo-Request and Echo-Reply packets are not treated as Network Control traffic. ICMP (Internet Control Message Protocol) is a network layer protocol used for diagnostic and error reporting purposes. While ICMP packets are used to send Echo-Request and Echo-Reply messages for network troubleshooting (commonly known as "ping"), they are not considered Network Control traffic. Network Control traffic refers to the traffic that is used for the management and control of the network itself, such as routing protocols and network management protocols. ICMP packets, including Echo-Request and Echo-Reply, are considered part of the data traffic.

Multiaction policing allows the setting of both Layer 2 and Layer 3 QOS markers at the same time. This means that it provides the advantage of being able to prioritize and manage network traffic at multiple levels simultaneously. By setting both Layer 2 and Layer 3 QOS markers, the network can prioritize different types of traffic based on their specific requirements, ensuring efficient and effective traffic management. This allows for better control and optimization of network resources, resulting in improved overall network performance.

Scheduler policies are only applied to egress interfaces. Egress interfaces are responsible for transmitting data from the device to the network, while ingress interfaces handle incoming data. By applying scheduler policies to egress interfaces, network administrators can control the rate at which data is transmitted and prioritize certain types of traffic. This allows for better management of network resources and ensures that critical traffic receives the necessary bandwidth.

A multifield classifier is used to classify network traffic based on multiple fields of a packet. In this case, the fields that can be matched on are the Source and Destination Address, as well as the Source or Destination Port. These fields provide information about the source and destination of the network traffic, as well as the specific port being used. By matching on these fields, the classifier can categorize the network traffic based on these criteria. The Advertising Router and IP TTL fields are not mentioned as options for matching in this question.

The QoS requirements on the CE and PE router differ depending on whether or not the CE router is managed by the service provider. If the CE router is managed by the service provider, they have control over the configuration and management of the router, allowing them to implement QoS policies according to their own requirements. However, if the CE router is not managed by the service provider, they have limited control over the router's configuration and management, making it more challenging to implement specific QoS policies.

Serialization delay is the time it takes to transmit a packet on a link. This delay occurs because data is transmitted sequentially, one bit at a time, over the link. The larger the packet size, the longer it takes to transmit the entire packet. Therefore, serialization delay is directly proportional to the packet size and the link's transmission rate.

The first statement is correct because the three least significant bits in the DSCP field are used to specify the drop probability. The second statement is incorrect because the DSCP field consists of the 6 most significant bits of the TOS (Type of Service) field, not all 8 bits. The third statement is correct because to convert DSCP to IP Precedence, the three highest bits in the DSCP field are matched. The fourth statement is incorrect because DSCP provides for up to 64 (not 8) different precedence levels, with four (not two) drop probabilities.

FRF.12 fragmentation is used on DLCIs configured for VoIP because FRF.12 is a fragmentation and interleaving technique specifically designed for VoIP traffic. It helps to reduce delay and jitter by breaking up large packets into smaller fragments.

FRF.11 Annex C fragmentation is used on DLCIs configured for VoFR because FRF.11 Annex C is a fragmentation and interleaving technique specifically designed for VoFR (Voice over Frame Relay) traffic. It helps to ensure the efficient transmission of voice packets over a Frame Relay network by breaking them into smaller fragments.