Lesson 11 (T/F):
When streaming stored multimedia applications, the user must download first the entire content before it can start playing.
False
- First, streaming stored video is…streamed!
- The video starts playing within a few seconds of receiving data, instead of waiting for the entire file to download first.
- It’s also interactive, which means that the user can pause, fast forward, skip ahead or move back in the video, and then see the response within a few seconds.
Lesson 11 (T/F):
With streaming stored multimedia applications, the user can pause, fast forward, skip ahead the audio/video.
True
- streaming stored video is also interactive, which means that the user can pause, fast forward, skip ahead or move back in the video, and then see the response within a few seconds.
Lesson 11:
Which of the following applications is the least sensitive to network delays?
Streaming video
VoIP
Interactive games
File Transfer
File Transfer
Lesson 11:
Which of the following applications is the least tolerant to packet losses?
Streaming video
VoIP
Interactive games
File Transfer
File Transfer
Lesson 11:
What are the consequences of using TCP instead of UDP for VoIP applications? Select all that apply.
Group of answer choices
Less packet loss
More packet loss
Less end-to-end delay.
More end-to-end delay
Less packet loss
More end-to-end delay
Lesson 11:
Which of the following are QoS metrics for VoIP applications? Select all that apply.
End-to-end delay
Jitter
Packet loss
Available bandwidth
End-to-end delay
Jitter
Packet loss
Lesson 11 (T/F):
In VoIP applications, increased packet jitter results in increased end-to-end delay.
Group of answer choices
True
False
True
Lesson 11:
Which of the following has an impact on packet jitter?
Encoding delay
Decoding delay
The type of multimedia application
Network conditions such as buffer sizes, queueing delays, network congestion levels
Network conditions such as buffer sizes, queueing delays, network congestion levels
- “jitter,” “packet jitter,” or “delay jitter”: 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 100ms, and another by 300ms.
Lesson 11 (T/F):
In VoIP applications, we have a harsher definition for packet loss, as we consider a packet to be lost if it never arrives or if it arrives after its scheduled playout.
True
What we consider packet loss for VoIP.
- We end up with a different definition for packet loss.
- With VoIP, a packet is lost if it either never arrives OR if it arrives after its scheduled playout. This is a harsher definition than for other applications (such as file transfers), but thankfully, VoIP can tolerate loss rates of between 1 and 20 percent, depending on what voice codec is used and other factors.
Lesson 11 (T/F):
With Forward Error Concealment we also transmit redundant data that can be used for reconstructing the stream at the receiver’s side. This approach to error recovery can lead to more bandwidth consumption.
True
But in general, FEC works by transmitting redundant data alongside the main transmission, which allows the receiver to replace lost data with the redundant data.
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.
Lesson 11 (T/F):
With interleaving we mix chunks of audio together so we avoid scenarios where consecutive chunks are lost. This approach can lead to increased latency.
True
- Interleaving works by mixing chunks of audio together so that if one set of chunks is lost, the lost chunks aren’t consecutive. The idea is that many smaller audio gaps are preferable to one large audio gap. (We note that the human ear is pretty intolerant of audio gaps, and ideally audio gaps should be under 30ms.)
- The tradeoff for interleaving is that 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.
Lesson 11:
Which transport-level protocol is preferred for video content delivery?
UDP
TCP
TCP
Thus, between UDP and TCP, content providers ended up choosing TCP for video delivery as 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.
Lesson 11:
What are the characteristics of good quality of experience from the user’s perspective? Select that apply.
Low or zero re-buffering
High video quality
Low video quality variations
Low start up latency
Low or zero re-buffering
High video quality
Low video quality variations
Low start up latency
- Low or zero re-buffering: users typically tend to close the video session if the video stalls a lot
- High video quality: Better the video quality, better the user QoE. A higher video quality is usually characterized by high bitrate video chunk.
- Low video quality variations: A lot of video quality variations are also known to reduce the user QoE.
- Low startup latency: Startup latency is the time it takes to start playing the video since the user first requested to play the video. Players typically fill up the video buffer a little before playing the video. For this lesson, we will skip considering startup latency and focus on the other three metrics.
Lesson 11 (T/F):
With throughout-based rate adaption, our goal is to have a buffer-filling rate that is greater than the buffer-depletion rate.
True
In order to have a stall-free streaming, clearly the buffer-filling rate should be greater than the buffer-depletion rate. In other words, C(t)/R(t) > 1 or C(t) > R(t).
Lesson 11 (T/F):
With rate-based adaption, when the bandwidth changes rapidly, the player takes some time to converge to the right estimate of the bandwidth, which can lead to overestimation of the future bandwidth.
True
- Thus, we can see under the case when the bandwidth changes rapidly, the player takes some time to converge to the right estimate of the bandwidth. As observed in this specific, this can sometimes lead to overestimation of the future bandwidth.
