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What is the difference between end-to-end delay and packet jitter? What are the causes of packet jitter?

Short Answer

Expert verified
End-to-end delay measures total travel time, while jitter measures arrival time variability; jitter is caused by congestion, routing changes, and varying path delays.

Step by step solution

01

Define End-to-End Delay

End-to-end delay refers to the total time taken for a packet to travel from the source to the destination across a network. It includes all types of delays, such as transmission delay, propagation delay, queuing delay, and processing delay that occur as the packet traverses the path to its destination.
02

Define Packet Jitter

Packet jitter, or simply jitter, refers to the variation in packet arrival time at the destination. If packets take different amounts of time to reach the destination due to network congestion or variable path routes, the difference between the expected arrival time and the actual arrival time of packets is termed as jitter.
03

Describe Difference Between End-to-End Delay and Packet Jitter

The main difference between end-to-end delay and packet jitter is that delay refers to the total time taken for the packet travel, whereas jitter measures the variability in packet arrival time. Delay is a measure of time taken, while jitter is a measure of inconsistency in timing.
04

Identify Causes of Packet Jitter

The causes of packet jitter include network congestion, routing changes, improper queuing of packets, and variations in network path delays. These factors can lead to different packets experiencing different delays, thereby causing variation in arrival times.

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Key Concepts

These are the key concepts you need to understand to accurately answer the question.

End-to-End Delay
End-to-end delay is a crucial concept in networking. It refers to how long it takes for a data packet to travel from its starting point to its endpoint in a network. This journey could include several stops along the way, such as routers or gateways.
To understand end-to-end delay thoroughly, consider the following components that contribute to the total time:
  • Transmission Delay: This occurs when data is placed on the transmission medium. It's influenced by the packet size and the data rate of the link.
  • Propagation Delay: This is the time it takes for the signal to travel through the transmission medium, usually at the speed of light, depending on the medium type.
  • Queuing Delay: This happens at network devices like routers where packets may need to wait in queues before they can be processed. Variations in queuing delays can significantly impact network performance.
  • Processing Delay: When packets reach intermediate devices like routers, there is some delay as they are processed and forwarded to the next hop.
Being aware of these types of delays helps in understanding how end-to-end delay impacts network communication and overall performance.
Packet Jitter
Packet jitter, often simply called jitter, refers to the variation in packet arrival times. In an ideal scenario, packets should arrive at consistent intervals. However, due to different factors in the network, this doesn't always happen.
If packets take varying lengths of time to reach their destination, this inconsistency is termed jitter. Jitter is especially problematic for time-sensitive applications like VoIP (Voice over Internet Protocol) or online gaming, which rely on a steady stream of data to function properly.
  • Why is Jitter Important? High jitter can lead to poor quality in voice and video calls, as data packets arrive out of order or too late, causing disruptions in service.
  • How is Jitter Measured? Jitter is measured as the difference in packet delay. High jitter indicates more variability and possible issues in network performance.
Managing jitter involves ensuring that the network is stable and packets flow smoothly with minimal variation.
Causes of Jitter
There are several reasons why jitter might occur in a network. Understanding these causes is essential for troubleshooting and optimizing network performance.
Key Causes of Jitter:
  • Network Congestion: This happens when too many packets are present in the network, which can lead to long queues and consequently variable delays.
  • Routing Changes: If the path that packets take changes mid-transmission, it may result in some packets taking longer paths, causing delays.
  • Improper Queuing: If packets are not queued correctly at network devices, some might experience longer waiting times than others.
  • Variable Path Delays: Different network paths have different lengths and types of infrastructure, leading to variability in time taken by packets to reach their destination.
It is important for network administrators to identify and mitigate these causes to ensure a smooth flow of packets and minimal jitter.
Network Congestion
Network congestion is a condition that arises when the demand for network resources is greater than the available capacity. It plays a significant role in causing various network performance issues, including jitter and increased end-to-end delays.
Why does Network Congestion Happen?
  • High Traffic Volume: An increase in users or devices trying to access the network can exceed the capacity of network components.
  • Limited Bandwidth: If the network's bandwidth is low, it can't handle a large amount of data traffic simultaneously, leading to slowdowns and congestion.
  • Traffic Bursts: Sudden spikes in data transmission can overwhelm the network, causing packets to line up in queues, resulting in delay and jitter.
Dealing with Network Congestion:
  • One effective strategy is implementing Quality of Service (QoS) policies to prioritize crucial data traffic.
  • Another approach is upgrading network infrastructure to support higher traffic loads and improve bandwidth capacity.
  • Monitoring network traffic continuously can help in identifying patterns of congestion and taking preemptive corrective actions.
Addressing congestion is vital for maintaining high-quality communications and effective network performance.

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Most popular questions from this chapter

True or false: a. If stored video is streamed directly from a Web server to a media player, then the application is using TCP as the underlying transport protocol. b. When using RTP, it is possible for a sender to change encoding in the middle of a session. c. All applications that use RTP must use port 87. d. If an RTP session has a separate audio and video stream for each sender, then the audio and video streams use the same SSRC. e. In differentiated services, while per-hop behavior defines differences in performance among classes, it does not mandate any particular mechanism for achieving these performances. f. Suppose Alice wants to establish an SIP session with Bob. In her INVITE message she includes the line: m=audio 48753 RTP/AVP 3 (AVP 3 denotes GSM audio). Alice has therefore indicated in this message that she wishes to send GSM audio. g. Referring to the preceding statement, Alice has indicated in her INVITE message that she will send audio to port 48753. h. SIP messages are typically sent between SIP entities using a default SIP port number. i. In order to maintain registration, SIP clients must periodically send REGISTER messages. j. SIP mandates that all SIP clients support G.711 audio encoding.

Consider a DASH system for which there are \(N\) video versions (at \(N\) different rates and qualities) and \(N\) audio versions (at \(N\) different rates and versions). Suppose we want to allow the player to choose at any time any of the \(N\) video versions and any of the \(N\) audio versions. a. If we create files so that the audio is mixed in with the video, so server sends only one media stream at given time, how many files will the server need to store (each a different URL)? b. If the server instead sends the audio and video streams separately and has the client synchronize the streams, how many files will the server need to store?

a. Consider an audio conference call in Skype with \(N>2\) participants. Suppose each participant generates a constant stream of rate \(r\) bps. How many bits per second will the call initiator need to send? How many bits per second will each of the other \(N-1\) participants need to send? What is the total send rate, aggregated over all participants? b. Repeat part (a) for a Skype video conference call using a central server. c. Repeat part (b), but now for when each peer sends a copy of its video stream to each of the \(N-1\) other peers.

Suppose an analog audio signal is sampled 16,000 times per second, and each sample is quantized into one of 1024 levels. What would be the resulting bit rate of the PCM digital audio signal?

List three disadvantages of UDP streaming.

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