Introduction to Telephony and Voice over IP (VoIP)

The Analog Circuit

Typical Analog Circuit

The twisted pair wires from the central switch office to a subscriber's home is

called a subscriber loop The subscriber loop handles two types of information: signals and voice on the same twisted pair

Loop Start Signaling

On-Hook

In on-hook stage the switch is open and there is no current flow

Off-Hook

When the handset is picked up (going off-hook) a switch on the phone closes

the connection between the two wires and a -48 VDC current is drawn from the central office switch
The switch determines that current is being drawn and provides dial tone so

the person on the phone knows it is time to dial a number

Dialing

Upon hearing the dial tone, the user pushes the number buttons, which are

connected to a tone generator inside the dial, which generates DTMF tones The Telephone Switch collects the DTMF digits and maps them to a physical subscriber

DTMF - Dual Tone Multi-Frequency

DTMF is the common method of sending dialing information (replaced pulse

dialing) Each number is represented by two tones which are transmitted simultaneously on the voice path Each row representing a low frequency and each column representing a high frequency
1209 697 1336 1477 1633

1 4 7 *

2 5 8 0

3 6 9 #

A B C D

770 852

941

Ringing

The Telephone Switch applies an AC ringing voltage which causes the sound

mechanism of the Called Telephone to ring

The Telephone Switch also plays a Ringback tone to assure the calling party

that a ringing signal is being sent on the called party's line

Conversation

The transmitter (handsets microphone) puts out an electric current which

varies in response to the acoustic pressure waves produced by the voice The resulting variations in electric current are transmitted along the telephone line to the other phone

Call Tear Down

When a party "hangs up" (puts the handset on the cradle), DC current ceases

to flow in that line, thus signaling to the telephone switch to disconnect the call
The switch plays a fast busy tone to the remote party

Call Progress Tones

In Telephony, call progress tones are audible tones sent from the PSTN or a PBX

to calling / called parties to indicate the status of phone calls.

Call Progress Tone

Description
Indicates that the telephone exchange is working, has recognized an off-hook, and is ready to accept digits
This tone assures the calling party that a ringing signal is being sent on the called party's line Indicates to the calling party that the remote phone is occupied Indicate that a person has dialed an invalid code, or that all trunks are busy and/or their call is unroutable

Dial Tone

Ringback Tone

Busy Tone

Reorder Tone (Fast Busy)

Telephony Network (1)

415-577-3800

415-577-3801

415-577-3700

Central Office)

415-577-3701

415-577-3722

415-577-3733

415-577-3760

415-577-3785

Telephony Network (2)

415-577-3800

415-577-3801

415-577-3700

415-577-3701

415-577-3702

415-577-3703

415-577-37xx

415-577-3704

415-577-3705

Digital Communication

Digital Communication

A digital trunk is a single communication path between two switches that is used

to carry many simultaneous voice conversations

Pulse Code Modulation (PCM)

A method of encoding an audio signal in digital format A standard audio signal is encoded as 8000 analog samples per second, of 8 bits

each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either -law (North America and Japan) or A-law (Europe and most of the rest of the world)

Time Division Multiplexing (TDM)

Uses time-division multiplexing

1 64 Kbps

2
64 Kbps

3
3 64 Kbps

1
3

. . . 32 64 Kbps

. . . 32

E1

Data rate of 2.048 Mbit/s (full duplex) Split into 32 time slots Each time slot sends and receives an 8-bit sample 8000 times per second (8 x 8000 x 32 = 2,048 Mbit/s) Ideal for voice telephone calls where the voice is sampled into an 8 bit number (PCM) One timeslot (TS0) is reserved for framing purposes One timeslot (TS16) is often reserved for signaling purposes

T1

Data rate of 1.544 Mbit/s Split into 24 time slots each encoded in 64 kbit/s streams 8 kbit/s of framing information for synchronization 64,000 x 24 + 8 = 1544 Mbit/s Timeslot (TS24) is often reserved for signaling purposes

Signaling Methods

In-band signaling is the exchange of signaling (call control) information on the same B-

channel that the telephone call itself is using CAS (Channel Associated Signaling)

Out-of-band signaling is the exchange of signaling that is done on a channel that is dedicated for the purpose and separate from the channels used for the telephone call Common Channel Signaling (CCS) such as ISDN and SS7

ISDN

Integrated Services Digital Network is an ITU-T term for integrated transmission

of voice, video and data on the digital public telecommunications network

Two interfaces are available:

PRI (Primary Rate Interface) primarily used to link PBXs and to connect a

PBX to the PSTN. Composed of 23 or 30 B-channels and one D-channel, all at 64 Kbps
BRI (Basic Rate Interface) an ISDN interface typically used by smaller sites

and customers. Consists of a single 16 Kbps D-channel plus 2 B-channels for voice and/or data

ISDN (Q.931) Call Flow

Voice Channel

BRI

Point to Point

U-Interface

S/T Interface NT1

PBX

ISDN Switch

Point to Multi-Point

U-Interface

NT1

S/T Interface

TE
ISDN Switch

TE

TE

TE

BRI (cont.)

The ISDN Basic Rate Interface (BRI) service offers two B-channels and one D-channel (2B+D)

B-channel service operates at 64 kbps and is meant to carry user data D-channel service operates at 16 kbps and is meant to carry control and

signaling information

Signaling

Clock Synchronization
Master Clock

Timing
Toll Center

Timing

Timing

End Office

End Office

Timing
PBX PBX

Timing

Voice over IP (VoIP)

What is VoIP

VoIP is a set of technologies that enable the transmission of

voice traffic over IP-based networks instead of the Plain Old Telephone System (POTS)

Circuit vs. Packet Switching

Circuit Switching - Traditional voice calls, running over the PSTN, are

made using circuit switching, where a dedicated circuit or channel is set up between two points before the users talk to one another
Packet Switching data transmission technique in which data is

separated into small 'packets', each with its own routing information and then sent through a shared, often public, network. At the other end the packets are reassembled into the original data format. In this method bandwidth is only used when something is actually being transmitted

VoIP Protocol Stack

VoIP is composed of two key components: The bearer (the actual voice being sent over the network) using the

RTP / RTCP protocols

The signaling (which are additional messaging necessary to control,

establish and tear-down the voice calls). The most common signaling protocols are: SIP, H.323, MGCP and MEGACO

RTP

RTP (Real-Time Transport Protocol) is used to encapsulate VoIP data packets

inside UDP packets. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data

RTP Header 12 octets

Voice Codecs

A codec (Coder/Decoder) converts analog signals to a digital bitstream, and back

into an analog signal for transmission across IP networks. Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted

Codec G.711 PCM (A-Law / Mu-Law) G.726 ADPCM G.729 CS-ACLEP G.723.1 CELP

Bit Rate (kbps) 64 16, 24, 32 and 40 8 6.3 and 5.3

VoIP Challenges

Delay - Each component in the path adds delay (sender, network,

receiver). ITU-T G.114 recommends 150 msec as maximum desired delay to achieve high voice quality.
Jitter - Variation in delay. The effects of jitter can be mitigated by storing

voice packets in a jitter buffer upon arrival and before producing audio
Packet loss - Occurs either in bursts or due to congested network.

Periodic loss in excess of 5-10% of all VoIP packets can degrade voice quality significantly

Delay

Start Talk

Sender Network
Packet X Transmitted Packet X Arrive

Receiver

Start Hear

Processing Delay

Network Transit Delay

End-to-End Delay

Processing Delay

Jitter
Jitter (delay variation) caused when voice packets suffer different transit

delays, causing variation in arrival times at the receiver The jitter buffer collects voice packets, stores them and sends them to the voice processor in evenly spaced intervals

Sender

t A D1 B D2 = D1 C
Receives

D3 = D2

VoIP Gateways

Enterprise PSTN & Data Network

Headquarters

Branch PSTN

IP

Telecommuter

FXS Gateways

FXS (Foreign Exchange Station) Emulates a PSTN/PBX.

Provides battery power, sends dial tone and generates ringing voltage. A standard telephone / fax machine plugs into such an interface to receive telephone services. FXS gateways convert (in real time) loop start signaling to SIP and variable electric current to RTP

FXO Gateways
FXO (Foreign Exchange Office) Generates the on-hook and off-hook

indicators used to signal a loop closure at the FXS's end of the circuit. Analog telephone handsets, fax machines and (analogue) modems are FXO devices FXO gateways convert (in real time) loop start signaling to SIP and variable electric current to RTP

FXS Call Flow

IP
Dial Tone Dialing

INVITE
100 Trying

Ringback Tone

180 Ringing

200 OK
ACK

Voice
BYE 200 OK

Digital Gateway

Digital gateways convert

(in real time) ISDN or CAS signaling to SIP and

PCM to RTP
PBX

E1 / T1

Mediant 1000

Mediant 2000

IP

E1 / T1
PCM

PSTN

ISDN Call Flow

IP
Setup INVITE Call Proceeding

100 Trying
180 Ringing

Alert

Connect

200 OK ACK

Voice
Disconnect BYE 200 OK

Release
Release Complete

T1 CAS (Wink Start) Call Flow

IP
Line Seizure Wink ANI Wink Optional Prefix/Authorization Code/ Called Number

INVITE 100 Trying

Ringback Tone Answer Supervision

180 Ringing 200 OK

ACK

Voice

Media Processing

Digital Audio Source (PCM)

Analog Audio Source