Telecommunications Tech Blog January 2014
Analog Telephony... How We Got Here, and What's
Happening To It Now?
By Mike Sandman •
In order to understand telephony
today and make it work right it's handy to know how telephony has evolved. The
key component of all this is our ears and mouth.
What is analog telephony? It's just a
pair of wires from one phone to another that transmits vibrations from our mouth
to someone else's ear (and vice versa).
Since it's unlikely that we'll evolve
to have a USB connector for a digital input to our heads anytime soon, every
kind of telephone will have an analog speaker and microphone of some kind for
the foreseeable future.
Without that USB connection in our
head, email and text will also require us to have eyes (analog light
transmission from a screen or piece of paper).
In the old days electrical signals
from phone to phone were switched between the two phones by Central Offices (by
human operators at cord boards, and later by mechanical or electronic Central
Offices). That's just connecting (temporarily splicing) two wires to a different
pair of two wires, maybe many times over a long distance, to get from one phone to another.
Phones weren't "electronic" until
fairly recently. The signal from the microphone was sent on the two wires
to the CO from the handset using a transformer in the telephone's internal
network, as can be seen in this picture of a standard 500 or 2500 set network
used since the 1940s
with no active electronics:
networks used screws instead of push-in spade lug sockets.
The telephone's internal network has a transformer that changes the four wires
going to the handset (transmit and receive) to two wires (going to the phone
Rotary dials contained no electronics, but touch tone dials did contain
electronics to make the tones (using transistors developed by Bell Labs),
powered by the DC current on a standard analog telephone line (which was
originally used to power the carbon
Really old phones each contained
"local" batteries to power the carbon microphone. "Common Battery" phones became
the norm where the electricity to power the transmitter (and dial) were supplied
from the Central Office end, instead of the subscriber end.
The amount of talk battery (DC current and voltage) on
a phone line was originally what was needed to make the carbon transmitter in
the handset work (over 20ma DC). The AC ring voltage was chosen at about 90VAC
at 20 to 30 cycles (instead of 60 cycles per second). The 20 cycle frequency was
slow enough to let the clapper in a mechanical bell swing back and forth between
the gongs slow enough to make a nice ringing sound.
The engineers designing later phone line
powered stuff with electronics just assumed that every line would have that minimum amount of
20ma DC loop current. The phone lines sometimes didn't. And engineers still
don't consider that today the loop current can be as high as 110ma DC, which
often burns up the electronics or makes it do strange stuff (so you're a test
pilot if you have high loop current because the engineers never tested it at
One of the problems the original
long-lines engineers dealt with was that phone lines were only two wires, with
transmit and receive on the same pair of wires. That means that as
they tried to amplify the line to make it easier to hear they created feedback
like you'd get from a mic on a PA system - usually called "singing" or "ringing"
during parts of the conversation (squealing).
Well before long lines were digital,
engineers developed ways to separate transmit from receive using transformers,
and amplified the voice with tube type amplifiers. Amplifying the voice
separately in each direction on a four wire line works great. Most of the time
four wire lines were used between Central Offices and to PBXs, and two wire out to the
subscriber with standard phones. The analog signals on analog pair gain systems used
tube type equipment to put many calls on a single coax cable, and AT&T
long-lines strung multi-core coax cables all over the country:
An early AT&T
long-lines coax core that would carry over 30,000 telephone conversations.
AT&T also used lots of microwave
radio towers for the analog signals that can still be seen here and there.
longer used by AT&T long-lines, but cellular stuff is on a lot of them. The
microwave also carried analog TV signals for networks before satellites.
Horns on an AT&T tower. 107 were spaced across the country about
apart, receiving and re-transmitting radio signals to the next tower.
When you make a 3-way conference call today from your home or office using a POTS line (2
wires) the amplification needed to allow each person to hear the other callers
clearly is added on the four wire side of the CO, not to the 2 wire side going
to your home or office. So the conference call sounds perfect.
In the early 1960s T1 pair-gain became
popular (made possible by transistors), where two pairs of wires between two
points could carry 24 conversations. The analog conversations were turned into TDM (Time-Division Multiplexing) electronically by equipment at each end of the
two pairs of wires, with repeaters along the way.
With TDM each conversation uses 64K
of bandwidth, with the entire T1 handling 24 of those conversations. That's on
pair of wires, that used to carry just two analog conversations. 1544Kbit/s of
Once engineers developed ways to
digitize the analog conversation, the pairs of wires were no longer the limiting
factor. Most of the long line communications was digitized, put on copper pairs
as well as coax and microwave (including submarine cables between continents).
They sent the data for phone calls
through satellites for a while but it took too long for the data to get up and
down from the satellites. The delay (latency) make it too hard to carry on a
conversation, and even the best echo cancellers of the day had a hard time
Then phone calls were finally put on
fiber optic strands.
The real breakthrough was when
engineers figured out how to split up the rainbow of colors in a single fiber
optic strand to send many times the amount of data. Instead of dividing the two
pairs of copper wires into 24 time slots, a single fiber is divided into many colors, each of
which can handle a lot of data (all using sophisticated electronics).
Our current telephone network is
still like the original circuit-switched network where the two wires for one
phone are connected to two wires on the other phone. But what happens
in-between the two Central Offices is invisible to us... where conversations
are usually converted to data.
Luckily for us the phone network has
always had very "lo-fi" voice quality (about 300 to 3500 hertz). Unusable
for music, and not real good for the tones sent from end to end by a modem.
A lot of "lo-fi" conversations can be
stuck on one data pipe and they arrive at the other end in a reasonable time
(latency), and in the right order. It takes 64K of bandwidth to send one "lo-fi"
conversation by TDM pair gain.
1 Netflix HD Movie = 78,000 VoIP
A single HD Netflix movie uses maybe
5 million bits of bandwidth, compared to 64 thousand bits for a "lo-fi" telephone conversation.
If I'm doing the math right that pipe with the HD Netflix movie can carry about
78,000 telephone conversations at the same time, or the one movie.
VoIP calls have a lot of competition out there!
The competition will probably become
overwhelming if super high quality 4K movies start filling the pipes. At 15
million bits per second sharing the same pipes with movies becomes absurd:
1 Netflix 4K Movie = 234,000 VoIP
It turns out that the process of
coding analog voice into data for TDM pair gain, and decoding it back to analog
is very fast. And it doesn't take long to send those bits of data from one point
to the other on dedicated copper or fiber. With TDM, what goes in first on one end comes
out on the other end first. Just what we need for voice calls.
This is the
best book I've found describing how telephones and the legacy
telephone network works, and it's really cheap bought used from Amazon!
Today phone companies are getting rid
of TDM as much as possible, replacing the infrastructure between the two central
offices with equipment that "packetizes" the data (Internet Protocol,
or IP). It takes small chunks of our voice and turns it into a packet of digital
data, and sends lots of those packets on to the destination Central Office.
The problem with packetizing our
voice is that the packets are often mixed with other packets of non-voice data.
Netflix movies, Pandora radio and even porn on the
For now, the "packets" sent between
the phone company Central Offices are using a "private" IP network owned by the
phone companies (mostly using fiber rather than copper). Since the packets don't
have to compete with Netflix or Porn they arrive pretty quickly and in the correct
order. That's likely to change soon since AT&T wants to close all the telephone
Central Offices and just put your voice calls on the public Internet, mixed with
all the other traffic.
VoIP packets were travelling from Point A to Point B with other voice packets, without
other types of data, it would work pretty well.
But those packets are competing for
bandwidth with movies and email so they are slowed down somewhat. Most of the
current Internet has QOS (Quality of Service) implemented to give priority to
voice traffic, which is unlike almost any other traffic in that it's "real time"
and can't be buffered. VoIP packets must arrive on-time and in the
When packets arrive out of order (too
late) they're thrown out by VoIP equipment, which is when we hear burbles and
cut-outs in the conversation. We might hear all the packets in the right order
on some calls,
but the latency (delay) can make it frustrating to carry on a conversation.
Sometimes the conversation
is converted from analog to digital (TDM or IP) and back several times on a
call to get from Point A to Point B. Bits are lost in the process, which is OK. We usually can't
hear that there's lost data until it "piles up," with too much of the original
conversation being lost by the various conversions.
It's just a fact of life that
packetizing digits for IP takes longer than converting the data for TDM. If
there isn't much latency (delay), we might only notice it if we put a handset from two
phones up to our ears and talk to ourselves. For conversations between two
people it's fine,
except when there is echo.
Echo essentially never
occurs on a VoIP to VoIP phone call where it's never changed to analog between
the two parties, and both users are using a regular handset. VoIP always has separate
transmit and receive (to and from your ear and mouth).
Echo problems occur any time a four
wire VoIP call (with separate transmit and receive) is put onto legacy telephone
equipment - which has two wires.
A four wire call with separate
transmit and receive is changed into a two wire analog call by a transformer
(a special kind called a hybrid transformer). Basically there are two wires
on one side of the transformer, and four on the other side. Through
induction the transformer changes the four wire analog signal from VoIP
equipment to a two wire analog signal used by POTS type telephone equipment (a
phone, modem, trunk card in a phone system, analog station port in a phone
system, ATA, etc.). The hybrid transformers carry the voice in both directions
during the call - converting two wire to four, and four wire to two.
The echo problem comes from two facts
that aren't easily overcome:
1. The hybrid transformer is never
100% efficient. Some of the transmit bleeds over to the receive, called "sidetone".
2. It takes a while for the
electronics to packetize (or un-packetize) the voice, done by "codecs."
Hybrid transformers have always been
used in phone equipment. That's why you can hear a little bit of your own voice
in your ear while you're talking on a standard POTS phone. It's nice because it gives us a warm feeling
that you haven't been disconnected. Most people hate phones without sidetone
because they're never sure they haven't been cut-off when talking.
With standard analog telephony
there's no delay when we're talking. The hybrid is inefficient and we hear the
sidetone, but since there's no delay it's not heard as echo. Just a little bit
of our own voice coming back to us. There's so little delay involved with
digitizing voice by TDM that we generally don't hear echo, and the echo
cancellers in the telephone network don't have to work very hard.
With the delay in VoIP caused
by packetizing the analog voice, we hear our own voice or the other party's voice
coming back after a short delay. Echo.
It's unlikely that we'll be able to
afford to replace every piece of analog equipment in the near future, so we have
to make it work with packetized voice. We're stuck with work-arounds.
The #1 work around is echo cancellers
at all the points where the packetized voice jumps from four wire to two wire.
Using DSPs (Digital Signal Processors) most of the echo can be eliminated if
it's not too loud, or isn't delayed too much.
The echo cancellers are located at
the phone company, in our VoIP phone systems with two wire trunk and station cards,
in VoIP ATAs, and in our VoIP phones, etc.
One specific echo problem comes when we're using a speakerphone on a VoIP phone, analog
a VoIP phone system, or even a cell phone's speakerphone. Some of that analog audio from the
caller on the speaker is picked up by the mic in the speakerphone. It's then
packetized again which has that inherent delay, and is heard as echo.
Getting rid of echo on a speakerphone
is fairly easy with an echo canceller because the engineers are working with an
environment that's always the same. The speakerphone mic is always the same
distance from the speaker, and the volume on the speaker can never be louder than what they
allowed the user to raise it to.
For phone calls on phone systems the
echo canceller has a lot more to do. It's made to adapt to the conditions as
quickly as possible, which is why you sometimes hear echo for the first few
seconds of a phone call. It's learning.
Some VoIP phone systems and VoIP equipment
allows a technician to adjust the settings on the echo canceller. When the settings
are mistakenly set to be too aggressive you end up hearing a crackling sound on the call as
the echo canceller starts removing some of the real voice as well as the echo
The #2 way to eliminate echo is to
reduce the volume of the call. That doesn't work so well since it's hard to
understand what the other person is saying if we can't hear what they're saying.
The #3 way to eliminate echo is to
make the hybrid transformer in the phone equipment more efficient either through
better design, adding some electronics, or adjusting the impedance of the phone
line to match the hybrid transformer in the equipment (which is what our
So if we eliminate all two wire
telephone equipment and don't use speakerphones we've made problems with echo
caused by packetization go away.
But it's still extremely unlikely we'll ever
have all of the packets for phone calls go over a private network
dedicated to VoIP, instead of over the public Interent, so we'll always be
dealing with voice packets sharing the pipes with Netflix and porn causing some
of the packets to be lost.
Greed has a lot to do with the poor
quality of VoIP calls. Putting 24 phone calls in a 1544Kbit/s pipe seemed like a
waste of bandwidth, so codec and compression algorithms were designed to
compress a pretty good sounding 64K phone call down to 8K. That's usually done
to either fit more (bad sounding) calls on a given pipe, or share more Internet
(downloads, email, etc.) with the phone calls on that single pipe.
You can still hear OK on a phone call
compressed down to 8K (with the g.729 codec) as opposed to 64K (the g.711
codec), but modem tones and faxes are clobbered to death. Lose some of those
g.711 packets along the way to delay and routing and you ain't going to get
reliable fax or modem traffic even with the best error correction (but faxes and
modem calls usually work after a couple of tries). Even DTMF touch tones have a
hard time being transmitted end to end by VoIP because it's optimized for voice,
not steady tones. Oh, and most alarm systems use DTMF or modems to communicate.
Since touch tones are important to
Voice Mail and IVRs (like telephone banking), VoIP has a work-around that lets
you turn on a mode where when the near end VoIP device hears the analog DTMF
tone but doesn't send the audio to the other end. The touch tone is stripped out
of the audio being sent to the far end, and a packet of data telling the far end
VoIP device what digit to reproduce on the far end is sent (the setting is
usually called RFC 2833). Of course there are
occasional problems where the whole DTMF digit audio doesn't get
stripped out and the far end equipment hears what's left of the "stripped out"
digit as well as the one created by the far end equipment, creating a double
digit (and a headache to troubleshoot without a DTMF decoder like our
Sometimes it just makes sense to keep
some of those POTS line from the phone company, at least until they won't give
them to us anymore. That's already happening in brand new neighborhoods and
business parks all over the country where the local phone company will no longer
run copper for any reason. The media converters that change the fiber to copper
have all kinds of strange problems working with real phone equipment - and you
never know what the problems will be until you try it.
What can we do to improve VoIP
The biggest thing we can do to
improve VoIP phone calls is to eliminate sharing the local Ethernet network,
and sharing the local pipe to the Internet.
When we share the local network with
computers we're limiting the bandwidth available to VoIP at the same time someone is uploading or
downloading large files. Worse, if the router is mis-configured the voice
packets may not have QOS - and even if QOS is set correctly there's only so much
the router can do with the size of the data and the pipe it's attached to.
Diagnosing VoIP isn't as easy as
connecting a butt-set to a line or station port. Sometimes you need to use a packet
sniffer to see why network stuff doesn't work right with computers (and is
effecting VoIP calls), or with VoIP why there is
delay or a conversation sounds bad.
Using a full duplex Ethernet tap (like our
Ethershark™) and a free packet sniffing program like Wireshark is really the
only way to know without guessing (guessing is very frustrating for all parties
involved!). With Wireshark you can even
reassemble the packets you capture from a VoIP call (by IP address) and play it back.
You can download the free Wireshark
sniffer program and a VoIP Screen Phone to your laptop without any other stuff
to see how it works on your network.
Terrell Boyer is a Wireshark VoIP expert and in his YouTube
video "The Ultimate Wireshark Tutorial" he gives you all the information
you need to start using Wireshark. This is the best 50 minutes you're ever going
to spend if you have to fix VoIP problems:
You can do a lot of troubleshooting
using an Ethernet switch that allows you to mirror a port to your laptop with
Wireshark. What you don't get on the mirrored port is errors caused by a bad
piece of equipment (NIC card, or whatever). Those errors aren't mirrored, and
Murphy's Law says that will be the problem!
The pipe to the Internet is even more
important than the local network since it's pretty small by comparison. Where the bandwidth
on your network may be 100 million bits per second, the pipe
from the router to the Internet is probably closer to 1.5 million, or maybe 6
million bits per second.
The controlling factor for an
Internet connection is the upload speed if the pipe isn't a symmetrical
line (same up and down speeds). ADSL is a really bad choice for VoIP since it's
asymmetrical, with upload bandwidth topping out at maybe 750 thousand bits and
download speeds at maybe 6 million bits per second. Even if ADSL2 is available
it usually doesn't offer high upload speeds.
If each VoIP phone call takes maybe
100 thousand bits per second of bandwidth (a reasonable number to figure on
average), you can have maybe 7 simultaneous conversations on an ADSL line
dedicated to VoIP, with 750 up. If you're sharing that ADSL line with email,
music, files and movies, you're down to maybe a couple of VoIP calls before
things start sounding pretty bad.
SDSL or T1s are a better choice for
voice, but you need a good Internet provider (not that easy to find!).
Generally speaking Internet pipes
from the cable company, U-verse or FIOS aren't a good choice to use with VoIP
because all of the subscribers in a neighborhood are sharing the same pipe to
the Internet from the "terminal" in the neighborhood. If a lot of subscribers
are downloading movies at the same time as your VoIP voice calls your calls are
going to sound pretty bad at times (but fine at other times). You have
absolutely no control over what the other subscribers are doing, so you have
absolutely no control over the quality of the sound of the business phone calls
with these types of Internet connections.
Business class DSL or T1s are
absolutely needed for a successful implementation of VoIP (assuming your
Internet provider has a big enough pipe to the Internet to service all the
subscribers from that Central Office). And a
dedicated pipe for VoIP makes a lot more sense than sharing that pipe with the
rest of the Internet traffic from the office.
Unfortunately it's difficult to
impossible to convince a business owner to spend more money when they were told
they would "save money" by going with VoIP, using their existing network and
Internet bandwidth - and getting rid of those expensive evil phone company POTS
and T1 lines.
On the good side we're all being
conditioned to accept pretty horrible sounding phone calls compared to the old
TDM networks / phone systems. In 1970 there would have been a revolt against the
phone company if calls sounded like they do today. Cell phones began that
change. We were just happy to not be tied to our landline phone all day long.
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engineers find a way to get a USB connector into our heads so we don't need
those inefficient analog mouths or ears anymore.