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CS 6250 LESSON 11 EXAM QUESTIONS WITH VERIFIED SOLUTIONS LATEST UPDATE 2026

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CS 6250 LESSON 11 EXAM QUESTIONS WITH VERIFIED SOLUTIONS LATEST UPDATE 2026 Continuous playout - Answers It should play out the same way it was recorded without freezing up in the middle. Differences between streamed audio and video - Answers Audio is more delay sensitive and has many simultaneous users What is the most delay sensitive? - Answers Conversational video and voice over IP Challenge of VoIP - Answers It's broadcasted over the internet 3 major categories of encoding schemes - Answers narrowband, broadband, and multimode Signaling protocol is responsible for what 4 major functions - Answers 1. User location 2. Session establishment 3. Session negotiation 4. Call participation management 3 metrics for quality of service for VoIP - Answers end-to-end delay jitter packet loss End-to-end delay sources - Answers 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? - Answers 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? - Answers 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 - Answers 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 - Answers It either never arrives OR if it arrives after its scheduled playout. 3 protocols for dealing with packet loss in VoIP - Answers FEC (forward error correction) Interleaving Error concealment Forward error correction - Answers Transmitting redundant data alongside the main transmission, which allows the receiver to replace lost data with the redundant data. Interleaving - Answers 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 - Answers Increases latency Error concealment - Answers "Guessing" what the lost audio packet might be What tech/trends have led to live and ondemand streaming? - Answers 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 - Answers Live and on demand Adaptive video content streaming steps - Answers 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? - Answers 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.

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CS 6250 LESSON 11 EXAM QUESTIONS WITH VERIFIED SOLUTIONS
LATEST UPDATE 2026


Continuous playout - Answers It should play out the same way it was recorded
without freezing up in the middle.
Differences between streamed audio and video - Answers Audio is more delay
sensitive and has many simultaneous users
What is the most delay sensitive? - Answers Conversational video and voice over IP
Challenge of VoIP - Answers It's broadcasted over the internet
3 major categories of encoding schemes - Answers narrowband, broadband, and
multimode
Signaling protocol is responsible for what 4 major functions - Answers 1. User
location
2. Session establishment
3. Session negotiation
4. Call participation management
3 metrics for quality of service for VoIP - Answers end-to-end delay
jitter
packet loss
End-to-end delay sources - Answers 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? - Answers 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? - Answers 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 - Answers 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 - Answers It either never arrives OR if it arrives after its
scheduled playout.
3 protocols for dealing with packet loss in VoIP - Answers FEC (forward error
correction)
Interleaving
Error concealment
Forward error correction - Answers Transmitting redundant data alongside the main
transmission, which allows the receiver to replace lost data with the redundant data.
Interleaving - Answers 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 - Answers Increases latency

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