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Why is a packet that is received after its scheduled playout time considered lost?

Short Answer

Expert verified
Packets received after their playout time are considered lost because they disrupt the flow of playback, degrading quality.

Step by step solution

01

Understanding Playout Time

In network communications, playout time refers to the scheduled time for rendering or processing the packet data, such as in streaming audio or video. This ensures a smooth experience without interruptions for the end-user.
02

The Definition of Lost Packet

A packet is considered lost if it fails to reach the destination before a given deadline. The deadline is typically the scheduled playout time in real-time applications, which is critical for maintaining smooth quality.
03

Impact of Delayed Packets

Packets arriving after the playout time do not serve their intended purpose, as the data they carry is out of sync with the required sequence for playback. Processing these delayed packets would disrupt the stream's continuity and degrade the quality.
04

Conclusion Based on Real-time Needs

In real-time communication, the timely arrival of packets is crucial. Delayed packets are neglected to preserve the overall timing and quality of the stream, hence they are considered lost when received after playout time.

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Key Concepts

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

Playout Time
In the world of network communications, playout time is essential for delivering content smoothly. Imagine you are streaming your favorite show. Each packet of data has a predetermined time to be "played out" or rendered on your screen. This precise timing helps to create a seamless viewing experience without any annoying lags or breaks.
  • Playout time ensures that audio and video data are shown in sync.
  • The timing is crucial to maintain continuity in the stream.

If packets arrive on time, your device processes them and keeps the stream fluid. However, if a packet arrives after its designated playout time, it cannot be integrated properly. It's like a puzzle piece arriving too late; the picture has already been completed without it. This is why precise synchronization using playout times is at the heart of smooth streaming experiences.
Lost Packet
In network terms, a packet is considered lost when it doesn't reach its destination in time for its intended use. This doesn't necessarily mean the packet was never received, but rather it was received too late to be useful.
  • The concept of packet loss is linked with deadlines, vital in real-time data transmission.
  • Even if a packet arrives but misses its deadline, it won't be used.

In real-time applications like video conferencing or live streaming, data packets need to be processed in a specific order and time frame. A lost packet disrupts this chain, often resulting in skipped frames or gaps in audio. So, the importance is not just about packet delivery, but timely delivery.
Real-Time Applications
Real-time applications are those that must respond to input or updates instantly or almost instantly. The performance of these applications relies heavily on the punctual delivery of data packets.
  • Examples include live streaming, online gaming, and video conferencing.
  • Real-time systems require strict timing to avoid latency issues.

In these situations, delays can lead to serious performance hiccups. For instance, in a video call, if packets are consistently late, the audio may fall out of sync with the video, causing confusion and a poor user experience. Thus, the effectiveness of real-time applications hinges on the flawless and immediate execution of tasks, emphasizing the need for fast packet delivery with minimal delay.

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Most popular questions from this chapter

a. Suppose we send into the Internet two IP datagrams, each carrying a different UDP segment. The first datagram has source IP address A1, destination IP address B, source port P1, and destination port T. The second datagram has source IP address A2, destination IP address B, source port P2, and destination port T. Suppose that A1 is different from A2 and that P1 is different from P2. Assuming that both datagrams reach their final destination, will the two UDP datagrams be received by the same socket? Why or why not? b. Suppose Alice, Bob, and Claire want to have an audio conference call using SIP and RTP. For Alice to send and receive RTP packets to and from Bob and Claire, is only one UDP socket sufficient (in addition to the socket needed for the SIP messages)? If yes, then how does Alice's SIP client distinguish between the RTP packets received from Bob and Claire?

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.

CDNs typically adopt one of two different server placement philosophies. Name and briefly describe these two philosophies.

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?

Suppose that the WFQ scheduling policy is applied to a buffer that supports three classes, and suppose the weights are 0.5, 0.25, and 0.25 for the three classes. a. Suppose that each class has a large number of packets in the buffer. In what sequence might the three classes be served in order to achieve the WFQ weights? (For round robin scheduling, a natural sequence is 123123123 . . .). b. Suppose that classes 1 and 2 have a large number of packets in the buffer, and there are no class 3 packets in the buffer. In what sequence might the three classes be served in to achieve the WFQ weights?

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