midterm_2013c

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TSM 6350G Midterm Exam
Spring Semester 2013: February 28, 2013
EXAM C
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NO CELL PHONES, PCS, COMPUTERS, OR ANY OTHER ELECTRONICS. BOOKS, CLASS NOTES, AND CALCULATORS ARE OK.
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Write clearly and IN INK. Think about your answer(s) before writing, or try writing your answers in a scratch blue book and
then transferring to the blue book your will hand in. (-25 pts if you do not write in ink).
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All problems are to be answered in a blue book. Put your name on the BACK of your blue book. (-10 points if you put your
name on the front). USE ONLY ONE BLUE BOOK! (-10 points if you use more than one Blue Book).
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PUT THE EXAM LETTER (A, B, OR C) ON THE FRONT OF THE BLUE BOOK. (-25 pts if you leave out the exam letter).
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The TAs will xxxx out all white space when you hand in your blue book(s).
Not surprisingly, the layer 1 and 2 protocol stacks in a 4G/LTE environment are different than what is
typically used in wired Ethernet environment. (The same is of course true for WiFi – the layer 1 and layer 2
stacks are also different than wired Ethernet). Another difference is that the codec used for Voice over LTE is
also different from what is typically used in the VoIP world. It’s called Adaptive Multi-Rate Wideband (AMRWB), and it was originally developed by Nokia (how the mighty have fallen), and a Canadian company that
spun out of Université de Sherbrooke in Quebec. Like Ethernet (and unlike WiFi or WiMAX), LTE is full
duplex. Both the protocol layers and the codec for LTE are standardized within the 3GPP consortium (which
is really a telecom dominated consortium, pretty different than IETF or IEEE).
Figure 1 shows the protocol stack for LTE. AMR-WB samples the analog voice waveform with 14 bits at 12.8
KHz and processes it into 20 ms frames at 12.7 kb/s. You can take L2 as about 80 bytes and L3 as about 100
bytes.
2.
3.
4.
5.
1. Please calculate the bandwidth occupied by a VoLTE conversation between two mobile phones on
the same carrier’s network (25 pts). What is the bandwidth efficiency (5 points)?
If the conversation were to be between two Ethernet VoIP phones that supported AMR-WB, do you expect that the
bandwidth efficiency on the Ethernet network to be HIGHER or LOWER than that of the VoLTE network? (Write down
one word in your blue book – HIGHER or LOWER). (10 pts)
If the designers of AMR-WB were to have picked 40 ms frames and 6 kb/s, would the latency for a call between two
mobile phones on the same carriers network be LONGER (MORE) or SHORTER (LESS) than what is described above?
(Write down one word in your blue book – LONGER or SHORTER) (10 pts)
Attached is a ladder diagram for a pretty standard call flow in a work environment (hint), along with a snippet of the
SIP debugs for these debugs. Each call in the overall call flow is in a different color. (Note: Make sure that your sip
debugs and ladder diagram have the same LETTER.) Based on the ladder diagram, sip debugs, or both:
a. Do the VoIP Phones here have to support more than 1 simultaneous call (10 pts, write down YES or NO)
b. Does the IP-PBX proxy the media, or is the media bypassing the IP-PBX (10 pts, write down PROXY, or BYPASS).
c. If you picked proxy in (4b), give two advantages for a media proxy. If you picked bypass, give two advantages for
media bypass. (10 PTS, WRITE DOWN YOUR ANSWER IN LESS THAN 15 WORDS)
d. In less than 10 lines in your blue book describe the overall calling scenario for your SIP debugs and ladder diagram.
(20 pts.) (note: -10 points for every line more than 10. -10 pints for writing each line that is small or un-intelligible).
Was Bach a German composer, or was he a Baroque composer (write down one word in your blue book – German or
Baroque). (2 pts).
1. First calculate the size of the IP payload. (12.7x103 bits/sec) X (20x10-3 sec) = 254 bits x (1 byte/8 bits) = 31.75 bytes = 32
bytes. Add the L2 and L3 overhead of 180 bytes (whew!) and you get 212 bytes. Now calculate the bit rate of sending out
212 bytes every 20 ms = (212 bytes) x (8 bits/byte) ÷ (20x10-3 sec) = 84.8 kbits/sec. The efficiency is (12.7 kb/s) ÷ (84.8
kbit/sec) = 15%.
a. Grading Rubric:
i. +25 if perfict.
ii. + 20 points if the math is screwed up, but the student completely understands the concept of calculating
the payload, adding the L2 and L3 encapsulation, and then dividing by the frame rate.
iii. +15 points if the understand calculating the payload and adding the L2 and L3 encapsulation, but screwed
up on dividing by the frame rate.
iv. +15 points if they screwed up either the payload or adding the L2 and L3 encapsulation, but understood to
divide by the frame rate.
v. + 5 points if they got the efficiency exactly correct.
vi. Note that the sampling rate and depth (12.8 KHz and 14 bits) is not needed for this problem. If I asked
about the maximum frequency that could be reconstructed, then it would be relevant.
2. HIGHER. The Ethernet network is clearly going to have better efficiency, since the payload is the same and L2/L3
encapsulation is only 40 bytes (compared to 212 bytes). +10 points or 0 points.
3. LONGER. The latency is dominated by the frame length (the time it takes to fill the bit bucket with bits). In this case, if the
frame length was 40 msec (instead of 20 msec) the latency would be 20 msec more! +10 points or 0 points.
4. Ladder Diagram.
a. YES +10 points or 0 points
b. BYPASS + 10 points or 0 points
c. BYPASS: 1) lower latency 2)less processing (fewer MIPS) for IP PBX +5 points for each correct answer
Proxy: 1)Transcoding possible, codec a  codec b 2) media forking (recording, lawful intercept).
d. This is hard. It’s a consultative transfer.
i. Party A calls Party B. They talk.
ii. Party B wants to transfer the call to Party C.
iii. Party B presses the TRANSFER button on the phone. Party B then talks to C, then hangs up and the call is
transferred to C. The Transfer sequence is:
1. Party A is placed on Hold
2. Party C is called. Party B and C talk.
3. Party B hangs up (or presses the transfer button again).
4. Party B (da phone) sends a REFER to the IP-PBX (with the TO: to the Party C). The IP PBX accepts
the REFER and SENDS a BYE to Party B.
5. The IP-PBX sends an INVITE to C, and an INVITE TO A (with the new SDP info in it), and the call is
established between A  C.
iv. + 20 points if the student gets that this is some sort of transfer. An extra +10 points if they understand it’s a
consultative transfer.
v. +10 points if they provide some other reasonable overall call flow or implementation.
vi. 0 otherwise
5. Both answers are correct. Bach was both a German composer (well, there really was no Germany at the time, he lived
before the German unification under Bismark) but he spoke German natively and he worked in what are either German or
Austrian cites), and a Baroque composer. +2 points
Spring 2013 TSM 6350G Midterm Exam C
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