CS 6250 Quiz 11 (All Quizzes) 2026/2027 Latest Update | Verified Questions and Correct Answers | Complete Study Guide – Georgia Institute of Technology

CS 6250 Quiz 11 (All Quizzes) 2026/2027 Latest Update | Verified Questions and Correct Answers | Complete Study Guide – Georgia Institute of Technology 2026/2027 | GRADED A+ | 100 out of 100 Question: Compare the bit rate for video, photos, and audio. Answer Video - 2 Mbps Facebook photo gallery browsing (1 photo/s) - 320 kbps Music - 128 kbps Question: What are the characteristics of streaming stored video? Answer - Streamed - video starts playing within a few seconds of receiving data, instead of waiting for the whole file - Interactive - user can pause, fast forward, skip ahead, move back - Continuous playout - should play out the same way it was recorded Question: What are the characteristics of streaming live audio and video? Answer - Many simultaneous users - Delay sensitive Question: What are the characteristics of conversational voice and video over IP? Answer - Delay sensitive - Loss tolerant - the occasional glitch is okay Question: How does the encoding of analog audio work (in simple terms)? Answer Generally speaking, audio is encoded by taking many (as in, thousands) of samples per second, and then rounding each sample's value to a discrete number within a particular range. (This "rounding" to a discrete number is called quantization.) Question: What are the three major categories of VoIP encoding schemes? Answer 1. Narrowband 2. Broadband 3. Multimode Question: What are the functions that signaling protocols are responsible for? Answer 1. User location - the caller locating where the callee is. 2. Session establishment - handling the callee accepting, rejecting, or redirecting a call. 3. Session negotiation - the endpoints synchronizing with each other on a set of properties for the session. 4. Call participation management - handling endpoints joining or leaving an existing session. Question: What are three QoS VoIP metrics? Answer - end-to-end delay - jitter - packet loss Question: What kind of delays are included in "end-to-end delay"? Answer - the time it takes to encode the audio - the time it takes to put it in packets, - all the normal sources of network delay that network traffic encounters such as queueing delays - "playback delay," which comes from the receiver's playback buffer (which is a mitigation technique for delay jitter) - decoding delay, which is the time it takes to reconstruct the signal Question: How does "delay jitter" occur? Answer Between all the different buffer sizes and queueing delays and network congestion levels that a packet might experience, different voice packets can end up with different amounts of delay. One voice packet may be delayed by 100 ms, and another by 300 ms. We call this phenomenon "jitter," "packet jitter," or "delay jitter." Question: What are the mitigation techniques for delay jitter? Answer The main VoIP application mechanism for mitigating jitter is maintaining a buffer, called the "jitter buffer" or the "play-out buffer." This mechanism helps to smooth out and hide the variation in delay between different received packets, by buffering them and playing them out for decoding at a steady rate. There's a tradeoff here, though. A longer jitter buffer reduces the number of packets that are discarded because they were received too late, but that adds to the end-to-end delay. A shorter jitter buffer will not add to the end-to-end delay as much, but that can lead to more dropped packets, which reduces the speech quality. Question: Compare the three major methods for dealing with packet loss in VoIP protocols. Answer - FEC (Forward Error Correction) - transmitting redundant data alongside the main transmission, which allows the receiver to replace lost data with the redundant data - Interleaving - mixing chunks of audio together so that if one set of chunks is lost, the lost chunks aren't consecutive - Error concealment - "guessing" what the lost audio packet might be Question: How does FEC (Forward Error Correction) deal with packet loss in VoIP? What are the tradeoffs of FEC? Answer Lost data is replaced with the redundantly-transmitted data. Tradeoffs: the more redundant data transmitted, the more bandwidth is consumed. Also, some of these FEC techniques require the receiving end to receive more chunks before playing out the audio, and that increases playout delay. Question: How does interleaving deal with the packet loss in VoIP/streaming stored audio? What are the tradeoffs of interleaving? Answer Not having consecutive loss chunks, so as to minimize the impact of a packet loss. Tradeoff: the receiving side has to wait longer to receive consecutive chunks of audio, and that increases latency. Unfortunately, that means this technique is limited in usefulness for VoIP, although it can have good performance for streaming stored audio. Question: How does the error concealment technique deal with packet loss in VoIP? Answer "Guessing" what the lost audio packet might be. This is possible because generally, with really small audio snippets (like between 4 ms and 40 ms), there's some similarity between one audio snippet and the next audio snippet. Could be repeating the previous packet or interpolating using the previous and next packets. Question: What developments lead to the popularity of consuming media content over the Internet? Answer 1. The bandwidth for both the core network and last-mile access links have increased tremendously over the years. 2. Video compression technologies have become more efficient. This enables streaming high-quality video without using a lot of bandwidth. 3. The development of Digital Rights Management culture has encterm-16ouraged content providers to put their content on the Internet. Question: Provide a high-level overview of adaptive video streaming. Answer The video content is first created -- say in a professional studio like this lesson or using a smartphone by a user. The raw recorded content is typically at a high quality. It is then compressed using an encoding algorithm. This encoded content is then secured using DRM and hosted over a server. Typically content providers have their own data centers such as Google or use third-party Content delivery networks to replicate the content over multiple geographically distributed servers. This makes sure that the content can be delivered in a scalable manner. The end-users download the video content over the Internet. The downloaded video is decoded and rendered on the user's screen. Question: Which protocol is preferred for video content delivery - UDP or TCP? Why? Answer TCP, as data loss would make it difficult to reliably decode video data. TCP also provides congestion control, which is required for effective bandwidth sharing. Question: What was the original vision of the application-level protocol for video content delivery, and why was HTTP chosen eventually? Answer The original vision was to have specialized video servers that remembered the state of the clients. These servers would control the sending rate to the client. In the event that the client paused the video, it would send a signal to the server and the server would stop sending video. Thus, all the intelligence would be stored at a centralized point and the clients, which can be quite diverse, would have to do a minimal amount of work. However, all this required content providers to buy specialized hardware. Another option was to use the already existing HTTP protocol. In this case, the server is essentially stateless and the intelligence to download the video will be stored at the client. A major advantage of this is that content providers could use the already existing CDN infrastructure. Moreover, it also made bypassing middleboxes and firewalls easier as they already understood HTTP. Question: Summarize how progressive download works. Answer The client sends byte-range requests for part of the video instead of requesting the entire video. Once the video content has been watched, the client sends a request for more content. Ideally, this should be enough for streaming without stalls. To account for throughput variations, the client pre-fetches some video ahead and stores it in a playout buffer. Question: How to handle network and user device diversity? Answer Content providers encode their video at multiple bitrates chosen from a set of pre-defined bitrates. Specifically, the video is chunked into segments which are usually of equal duration. Each of these segments is then encoded at multiple bitrates and stored at the server. The client request while requesting for a segment also specifies its quality. How does the bitrate adaptation work in DASH? Answer A video in DASH is divided into chunks and each chunk is encoded into multiple bitrates. Each time the video player needs to download a video chunk, it calls the bitrate adaptation function, say f. The function f that takes in some input and outputs the bitrate of the chunk to be downloaded. What are the goals of bitrate adaptation? Answer 1 .Low or zero re-buffering: users typically tend to close the video session if the video stalls a lot 2. High video quality: the better the video quality, better the user QoE. A higher video quality is usually characterized by a high bitrate video chunk. 3. Low video quality variations: a lot of video quality variations are also known to reduce the user QoE. 4. Low startup latency: startup latency is the time it takes to start playing the video since the user first requested to play the video. What are the different signals that can serve as an input to a bitrate adaptation algorithm? Answer - Network Throughput: The first signal that can facilitate the selection of bitrate is the network conditions, or more specifically, the network throughput. - Video Buffer: The amount of video in the buffer can also inform the decision of the video bitrate for the next chunk. Explain buffer-filling rate and buffer-depletion rate calculation. Answer The buffer-filling rate is essentially the network bandwidth divided by the chunk bitrate. The buffer-depletion rate is simply 1 because 1 s of video content gets played in 1 s. What steps does a simple rate-based adaptation algorithm perform? Answer 1. Estimation: The first step involves estimating the future bandwidth. This is done by considering the throughput of the last few downloaded chunks. Typically, a smoothing filter such as moving average, or the harmonic mean is used over these throughputs to estimate the future bandwidth. 2. Quantization: In the second step, the continuous throughput is mapped to a discrete bitrate. Basically, we select the maximum bitrate that is less than the estimate of the throughput, including a factor in this selection. Explain the problem of bandwidth over-estimation with rate-based adaptation. Answer TL;DR Under the case when the bandwidth changes rapidly; the player takes some time to converge to the correct estimate of the bandwidth. As observed in this specific, this can sometimes lead to an overestimation of the future bandwidth. Explain the problem of bandwidth under-estimation with rate-based adaptation. Answer This was terribly explained... Continuous playout Answer It should play out the same way it was recorded without freezing up in the middle. Differences between streamed audio and video Audio is more delay sensitive and has many simultaneous users What is the most delay sensitive? Conversational video and voice over IP Challenge of VoIP It's broadcasted over the internet 3 major categories of encoding schemes narrowband, broadband, and multimode Signaling protocol is responsible for what 4 major functions 1. User location 2. Session establishment 3. Session negotiation 4. Call participation management 3 metrics for quality of service for VoIP end-to-end delay jitter packet loss End-to-end delay sources The time it takes to encode the audio The time it takes to put it in packets Normal sources of network delay that network traffic encounters such as queueing delays "playback delay" decoding delay Why is jitter problematic? It interferes with reconstructing the analog voice stream. With large jitter, we end up with more delayed packets that end up getting discarded, and that can lead to a gap in the audio. Too many dropped sequential packets can make the audio unintelligible How do you mitigate jitter? Maintaining a buffer, called the "jitter buffer" or the "play-out buffer." This mechanism helps to smooth out and hide the variation in delay between different received packets, by buffering them and playing them out for decoding at a steady rate. Tradeoff for jitter buffer A longer jitter buffer reduces the number of packets that are discarded because they were received too late, but that adds to the end-to-end delay. A shorter jitter buffer will not add to the end-to-end delay as much, but that can lead to more dropped packets, which reduces the speech quality. Packet loss for VoIP It either never arrives OR if it arrives after its scheduled playout. 3 protocols for dealing with packet loss in VoIP FEC (forward error correction) Interleaving Error concealment Forward error correction Transmitting redundant data alongside the main transmission, which allows the receiver to replace lost data with the redundant data. Interleaving Doesn't transmit ANY redundant data, and so it doesn't add extra bandwidth requirements or bandwidth overhead. It works by mixing chunks of audio together so that if one set of chunks is lost, the lost chunks aren't consecutive. Tradeoff for interleaving Increases latency Error concealment "Guessing" what the lost audio packet might be What tech/trends have led to live and ondemand streaming? 1. The bandwidth for both the core network and last-mile access links have increased tremendously over the years. 2. The video compression technologies have become more efficient. Two types of content streamed over the internet Live and on demand Adaptive video content streaming steps 1. The video content is first created 2. It is then compressed using an encoding algorithm. 3. This encoded content is then secured using DRM and hosted over a server. 4. Content delivery networks to replicate the content over multiple geographically distributed servers. This makes sure that the content can be delivered in a scalable manner. 5. The end-users download the video content over the Internet. 6. The downloaded video is decoded and rendered on the user's screen. What protocol is chosen for video delivery? Why? TCP bc it provides reliability. An additional benefit of using TCP was that it already provides congestion control which is required for effectively sharing bandwidth over the Internet. What application-layer protocol should be used for video delivery? Why? http because the server is essentially stateless and the intelligence to download the video will be stored at the client. A major advantage of this is that content providers could use the already existing CDN infrastructure. Moreover, it also made bypassing middleboxes and firewalls easier as they already understood HTTP. What are the disadvantages to using HTTP get for video? 1. Users often leave the video mid-way. Thus, downloading the entire file can lead to a waste of network resources. 2.The video content that has been downloaded but not played so far would have to be stored. Thus, we will need a video buffer at the client to store this content in memory Playout buffer Number of seconds of video that can be downloaded in advance or in terms of size in bytes. 2 streaming states Filling state and steady state Filling state This happens when the video buffer is empty and the client tries to fill it as soon as possible. Steady state After the buffer has become full, the client waits for it to become lower than a threshold. After which, the client sends a request for more content. Bitrate adaption Once the available bandwidth reduces due to a background download, you can reduce the video quality and stream the video lower, avoiding any stalls. DASH/Dynamic streaming over http The client dynamically adjusts the video bitrate based on the network conditions and device type. Dynamic streaming signifies the dynamic bitrate adaptation. 4 conditions for good quality of experience for bitrate adaption Low or zero re-buffering High video quality Low video quality variations Low startup latency Network throughput The first signal that can facilitate the selection of bitrate 2 types of input to bitrate algorithms Network throughput and video buffer Rate based adaption steps Estimation and quantization Quantization Estimation Considering the throughput of the last few downloaded chunks Quantization The continuous throughput is mapped to discrete bitrate. Basically, we select the maximum bitrate that is less than the estimate of the throughput, including a factor in this selection. Why do we add a factor during quantization? We want to be a little conservative in our estimate of the future bandwidth to avoid any re-buffering If the chunks are VBR-encoded, their bitrate can exceed the nominal bitrate Finally, there are additional application and transport-layer overheads associated with downloading the chunk and we want to take them into account Bitrate for videos, photos, and audio High for video, audio has lower bit rate, photos are somewhat in the middle Streaming stored video characteristics Can play within a few seconds of getting data, interactive, continuous playout out, on a CDN Streaming live audio and video characteristics Many simultaneous users, delay sensitive Conversational voice and video over IP Highly delay sensitive and loss tolerant How is analog video encoded? By taking many (as in, thousands) of samples per second, and then rounding each sample's value to a discrete number within a particular range. Tradeoffs of FEC The more redundant data transmitted, the more bandwidth is consumed. Also, some of these FEC techniques require the receiving end to receive more chunks before playing out the audio, and that increases playout delay. Progressive download Thus, instead of downloading the content all at once, the client tries to pace it. This can be done by sending byte-range requests for part of the video instead of requesting the entire video. Once the video content has been watched, it sends request for more content. Ideally, this should be enough for streaming without stalls. How to handle network and user device diversity? Bitrate adaption Buffer filling The network bandwidth divided by the chunk bitrate Buffer depletion The output rate is simply 1 Overestimation with bitrate adaption Bandwidth changes rapidly, the player takes some time to converge to the right estimate of the bandwidth. Underestimation with bitrate adaption Note that this problem happens because of the ON-OFF behavior in DASH. Had it been two competing TCP-flows they would have gotten their fair share. While we have seen this problem for DASH and a competing TCP flow, it can also happen in competing DASH players leading to an unfair-allocation of network bandwidth.