What is meant by interactivity for streaming stored audio/video? What is meant by interactivity for real-time interactive audio/video?
Three "camps" were discussed for evolving the Internet so that it better supports multimedia applications. Briefly summarize the views of each camp. In which camp do you belong?
Figures 6.2-2, 6.2-3 and 6.2-4 present three schemes for streaming stored media. What are the advantages and disadvantages of each scheme?
What is the difference between end-to-end delay and delay jitter? What are the causes of delay jitter?
Why is a packet that is received after its scheduled playout time considered lost?
Section 6.3 describes two FEC schemes. Briefly summarize them. Both schemes increase the transmission of the stream by adding overhead. Does interleaving also increase the transmission rate?
How are different RTP streams in different sessions identified by a receiver? How are different streams from within the same session identified? How are RTP and RTPC packets (as part of the same session) distinguished.
Three RTCP packet types are described in Section 6.4. Briefly summarize the information contained in each of these packet types.
In Figure 6.4-9, which of the H.323 channels run over TCP and which over UDP? Why?
In Section 6.6, we discussed non-preemptive priority queuing. What would be preemptive priority queueing? Does preemptive priority queueing make sense for computer networks?
Give an example of scheduling discipline that is not work conserving.
Guaranteed Service provides an application no loss and firm bounds on delay. Referring back to Figure 2.1-2, are there any applications that require both no loss and firm bounds on delay?
What are some of the difficulties associated with the Intserv model and per-flow reservation of resources?
Surf the Web and find three products for streaming stored audio and/or video. For each product, determine: (a) whether meta files are used; (b) whether the audio/video is sent over UDP or TCP; (c) whether RTP is used; (d) and whether RTSP is used.
Write a poem, a short story, a description of a recent vacation, or any other piece which takes 2-5 minutes to recite. Recite and record your piece. Convert your recording to one of the RealNetworks audio formats using one of the RealNetworks free encoders. Upload the file to the same server that holds your personal homepage. Also upload the corresponding meta file to the server. Finally create a link from your homepage to the meta file.
Consider the client buffer shown in Figure 6.2-4. Suppose
that the streaming system uses the fourth option, that is,
the server pushes the media into the socket as quickly as
possible. Suppose the available TCP bandwidth >> d most
of the time. Also suppose that the client buffer can only
about one third of the media. Describe how x(t) and the contents of the client buffer will evolve over time.
Are the TCP receive buffer and the media player's client buffer the same thing? If not, how do they interact?
In the Internet phone example in Section 6.3, let h be the total number header bytes added to each chunk, including UDP and IP header.
Assuming an IP datagram is emitted every 20 msec, find the transmission in bits in second for the datagrams generated by one side of this application.
Consider the procedure described in Section 6.3 for estimating average delay di. Suppose that u = .1. Let r1 − t1 be the most recent sample delay, let r2 − t2 be the next most recent sample delay, etc.
For a given audio application suppose four packets have arrived at the receiver with sample delays r4 − t4 , r3 - t3 , r2 − t2 , r1 − t1. Express the estimate of delay d in terms of the four samples.
Generalize your formula for n sample delays.
For the formula in part (b) let n approach infinity and give the resulting formula. omment on why this averaging procedure is called an exponential moving average.
Repeat the above question for the estimate of average delay deviation.
Compare the procedure described in Section 6.3 for estimating average delay with the procedure in Section 3.5 for estimating round-trip time. What do the procedures have in common? How are they different?
Consider the adaptive playout strategy described in Section 6.3.
How can two successive packets received at the destination have timestamps that differ by more than 20 msecs when the two packets belong to the same talkspurt?
How can the receiver use sequence numbers to determine whether a packet is the first packet in a talkspurt? Be specific.
Recall the two FEC schemes for Internet phone described in Section 6.3. Suppose that the first scheme generates a redundant chunk for every four original chunks. Suppose the second scheme uses a low-bit-rate encoding whose transmission rate is 25% the transmission rate of nominal stream.
How much additional bandwidth does each scheme require? How much playback delay does each scheme add?
How do the two schemes perform if at most one packet is lost in every group of five packets? Which scheme will have better audio quality?
How do the two schemes perform if at most one packet is lost in every group of two packets? Which scheme will have better audio quality?
How is the interarrival time jitter calculated in the RTCP reception report? Hint: Read the RTP RFC.
Suppose in a RTP session there are S senders and R receivers. Use the formulas at the end of Section 6.4 to show that RTCP limits its traffic to 5% of the session bandwidth.
How is RSTP similar to HTTP? Does RSTP have methods? Can HTTP be used to request a stream?
How is RSTP different from HTTP. For example, is HTTP in-band or out-of-band? Does RTSP require state information about the client (consider the pause/resume function)?
What are the current Microsoft products for audio/video real-time conferencing. Do these products use any of the protocols discussed in this chapter (e.g., RTP or RTSP)?
Suppose that the WFQ scheduling policy is applied to a buffer that supports three classes, and suppose the weights are .5, .25 and .25 for the three classes.
Suppose that each class has a large number of packets in the buffer. In what sequence might the three classes be served in to achieve the WFQ weights? (For round-robin scheduling, a natural sequence is 123123123...).
Suppose that classes 1 and 2 have a large number of packets in the buffer, and there are no class 2 packets in the buffer. In what sequence might the three classes be served in to achieve the WFQ weights?
Consider the leaky bucket policer (discussed in Section 6.6) that polices the average rate and burst size of a packet flow. We now want to police the peak rate, p, as well. Show how the output of this leaky bucket policer can be fed into a second leaky bucket policer so that the two leaky buckets in series police the average rate, peak rate, and burst size. Be sure to give the bucket size and token generation rate for the second policer.
A packet flow is said to conform to a leaky bucket specification (r,b) with burst size b and average rate r if the number of packets that arrive to the leaky bucket is less than rt + b packets in every interval of time of length t for all t. Will a packet flow that conforms to a leaky bucket specification (r,b) ever have to wait at a leaky bucket policer with parameters r and b? Justify your answer.
Show that as long as r1 < R·wi/(∑wj), then dmax is indeed the maximum delay that any packet in flow 1 will ever experience in the WFQ queue.
How can a host use RTCP feedback information to determine whether problems are local, regional, or global?
Do you think it is better to stream stored audio/video on top of TCP or UDP?
In RSVP, are reservation sytles relevant for one-to-many multicast sessions?
Write a one-page report on prospects for Internet phone in the market place.
Can the problem of providing QoS guarantees be solved simply by "throwing enough bandwidth" at the problem, i.e., by upgrading all link capacities so that bandwidth limitations are no longer a concern?
An interesting emerging market is using Internet phone and a company's high-speed LAN to replace the same company's PBX (private branch exchange). Write a one-page report on this issue. Cover the following questions in your report:
What is a traditional PBX? Who uses them?
Consider a call between a user in the company and another user out of the company, who is connected to the traditional telephone network. What sort of technology is needed at the interface between the LAN and the traditional telephone network?
In addition to Internet phone software and the interface of question (b), what else is needed to replace the PBX?
Consider the four "pillars" of providing QoS support in Section 6.5. Describe the circumstances, if any, under which each of these pillars can be removed.
Use the Web to find three companies that manufacture H.323 gatekeepers. Describe their products.