Comp 346/488: Intro to Telecommunications

Tuesdays 7:00-9:30, Lewis Towers 412

Class 10: Nov 9

Reading:
    Chapter 9, Spread Spectrum and CDMA
    Chapter 14, Cellular networks
    Chapter 11, ATM

Exam problem 2, and working with units

Program 2


Cellular telephony

9.4:  CDMA

Here, each bit becomes k "chips" (k ~ 128). The chips are sent using FHSS or DSSS. Each user has an individual "chipping code".

As with DSSS, one consequence is that data is spread out over a range of frequencies, each with low power

basic idea:
K users: each user transmits k chips on k frequencies at the same time. Ideally we use K=k, but in practice K can be somewhat larger. All the transmissions "add" together in the airwaves. The transmissions are done in such a way that each receiver can extract its own signal, even though all the signals overlap.

Trivial way to do this: ith sender sends on frequency i only, and does nothing on other frequencies. This is just FDM. We don't want to use this, though.

Example from Table 9.1:
CA∙CB = 0, CA∙CC = 0, CB∙CC = 2 (using ∙ for "dot product")

A sends (+1)*CA or (-1)*CA

suppose D = data = a*CA + b*CB, a,b = +-1

Then D∙CA = a*CA∙CA + b*CB∙CA = a*6 + b*0 = 6a; we have recovered A's bit!


Cell phones

analog
uses FDMA (Freq Division Multiple Access): allocates a 30kHz channel to each call (2 channels!), held for duration of call.

TDMA
digitizes voice, compresses it 3x, and then uses TDM to send 3 calls over one 30kHz channel.

GSM
form of TDMA with smaller cells, more dynamic allocation of cells. Note that if everyone talks at once, there might not be enough cells for all!  Data form: GPRS (General Packet Radio Service)

CDMA
sprint PCS, US Cellular, others
code division multiple access: kind of weird. This is a form of "spread spectrum". Signals are encoded, and spread throughout the band. Any one bit is sent as ~128 "chips", but chips may be used by multiple senders. Demultiplexing is achieved because codes are "orthogonal", as above.

This strategy gives good gradual degradation as # of calls increases, and good eavesdropping prevention (though not as good as strong encryption). It also gives excellent bandwidth utilization (3x TDMA).

There is a deep relationship in CDMA between power and bandwidth, but in a good way: phones negotiate to use the smallest power for their needs

second reason for CDMA: allows gradual performance falloff with oversubscription

Chapter 14

Fig 14.1 on cell geometries: a cell diameter is typically a few miles, though in cities it can be much less
Fig 14.2 on frequency reuse (for TDMA, FDMA)
Fig 14.6 on finding phones: note that paging is pretty rare; phones instead "check in" regularly.

Fast (half-wavelength, or 6") fading v slow fading
fastfade.pdf (from the IEEE 802.11 "wi-fi" specification)

AMPS: FDMA, analog
channel spacing 30kHz!!! So big because of multipath problem

Multipath distortion:
Figure 14.7

phase distortion: multipath can lead to different copies arriving sometimes in phase and sometimes out of phase (this is the basic cause of "fast fading").

Intersymbol interference (ISI) and Fig 14.8

Note that CDMA requires good POWER CONTROL
Phones all adjust signal strength downwards if received base signal is stronger than the minimum.

Go through Andrews presentation at http://users.ece.utexas.edu/~jandrews/publications/cdma_talk.pdf
SMS (text messaging)
This originated as a GSM protocol, but is now universally available under CDMA as well.
Basically, the message piggybacks on "system" packets.


Erlang model

Suppose we know the average number of simultaneous users at the peak busy time (could be calls using a trunk line, could be calls to help desk). We would probably figure that at least some of the time, the demand might be higher than average. How many lines do we actually need?

To put this another way, suppose we flip 200 coins. The average number of heads is 100. What are the odds that in fact we get less than or equal to 110 heads? Or, to put it in a slightly more parallel way, suppose we keep doing this. We want to reward every head-getter, 99% of the time. How many rewards do we have to keep on hand?

One parameter is the acceptable blocking rate: the fraction of calls that are allowed to go unconnected. As this rate gets smaller, the number of lines needed will increase.

Example: how many lines do we need to handle a peak rate of 100 calls at any one time (eg 2,000 users each spending 5% of the day on the phone), and a maximum blocking rate of 0.01%?

(From http://erlang.com/calculator/erlb and my own program)

We assume that calls for which a line is not available are BLOCKED, not queued.


rate
 allowed blocking
 lines needed
 excess
10
 0.01
 18
 180%
20
 0.01
 30
 150%
50
 0.01
 64
 128%
100
 0.01
 118
 118%
180
 0.01
 201
 111%
200
 0.01
 221
 110%
500
 0.01
 526
 105%
1000
 0.01
 1029
 103%

To a reasonable approximation, if we have 200 lines, we can handle ~180 calls.

Here's a table where the rate is 0.001:

rate
 allowed blocking
 lines needed
 excess
10  
0.001 21 210%
20
 0.001 35 175%
50
 0.001 71 142%
100
 0.001 128 128%
180
 0.001 216 120%
200
 0.001 237 119%
500
 0.001 554 111%
1000
 0.001 1071 107%


# lines needed is always greater than the average peak rate. The question is by how much.

As allowed blocking rate goes down, the # of lines needed goes up, for the same rate.

Also, as the rate goes up, the % of extra lines needed goes DOWN, due to "law of large numbers"

            

ATM: Asynchronous Transfer Mode

versus STM: what does the "A" really mean?

Basic idea: virtual-circuit networking using small (53-byte) fixed-size cells

Review Fig 10.11 on large packets v small, and latency. (Both sizes give equal throughput, given an appropriate window size.)

basic cellular issues
delay issues:
    10 ms echo delay: 6 ms fill, ~4 ms for the rest
    In small countries using 32-byte ATM, the one-way echo delay might be 5 ms
    echo cancellation is not cheap!
   
150 ms delay: threshhold for "turnaround delay" to be serious

packetization: 64kbps PCM rate, 16kbps compressed rate. Fill times: 6 ms, 24 ms respectively

buffering delay: keep buffer size >= 3-4 standard deviations, so buffer underrun will almost never occur. This places a direct relationship between buffering and jitter.

encoding delay:
time to do sender-side compression; this can be significant but usually is not.


voice trunking (using ATM links as trunks) versus voice switching (setting up ATM connections between customers)

voice switching requires support for signaling

ATM levels:

A few words about how TCP congestion control works


ATM
basic argument (mid-1980s): Virtual circuit approach is best when different applications have very different quality-of-service demands (eg voice versus data)

VCC (Virtual Circuit Connection) and VPC (Virtual Path: bundle of VCCs)
Goal: allow support for QOS (eg cell loss ratio)

ATM cell format: Fig 11.6 (9th ed.), 11.4 (7th & 8th eds)
3-bit PT (Payload Type) field: user bit, congestion bit, SDU bit
CLP (Cell Loss Priority) bit: if we have congestion, is this in the first group to go or the second?

48-byte compromise [anecdote?]
why cells:

no-reordering rule, and its consequences