Voice over IP

Objectives
At the end of the module, the participants will be able to:

Define VoIP terms

Explain Packet Voice protocols

Differentiate VoIP codecs

Describe VoIP architecture

Course Outline

VoIP scenarios

Packet voice protocols

Voice payload

VoIP connectivity to PSTN / GSM

Voice Quality

VoIP Regulation

Is it a service?

Voice over IP is ?

Is it a new technology?

Is it a Protocol?

VoIP Defined

What is VoIP?
VoIP is a category of hardware and software
that enables people to use the Internet as the
transmission medium for telephone calls by
sending voice data in packets using IP rather
than by traditional circuit transmissions of the
PSTN.

[Source: Webopedia]

Another Definition
Voice over IP (also called VoIP, IP
Telephony, and Internet telephony)
is a technology enabling routing of voice
conversations over the Internet or any
other IP network.
[Source: Wikipedia]

Benefits of VoIP
For the provider:

Cost savings

Distributed nature

Conserve bandwidth by expanding only

when needed

Source: Cisco

Benefits for the Provider

Open standards and

multi-vendor
interoperability

Reduce or eliminate
dependency on
proprietary solutions

Can use any vendor

product without
compromising
functionality

Source: Cisco

Benefits for the Provider (2)

Integrated voice and

data networks

Voice becomes just

another IP
application

Operators need a
single network to
cater for voice and
data

Source: Cisco

Benefits for the User

Cheaper calls

No toll charges

Increased Functionality
Multiple numbers
Incoming phone calls
automatically routed to
IP-phone wherever it is
plugged-in

Makes it easy for traveling

Greater control on
subscribers part

Source: Cisco

VoIP Scenarios

Voice over IP

Have you tried VoIP lately?

Netopia is a PLDT subsidiary
Over 200 franchises nationwide

Net2Phone call cards available at

Netopia

Available in $2, $5, $10

denominations

10 per minute (USA)

Packet Voice Scenarios

PC-to-PC

PC-to-Phone

Phone-to-PC

Phone-to-Phone

PC to PC VoIP Connection
Cebu, Philippines

Internet

New York, USA

PC-to-Phone / Phone-to-PC

What do I need on a PC?

Operating System

Windows

Linux

Sound Card

Headset

Microphone

Softphone

Broadband / DSL
Connection

What are Soft Phones?

A software that simulates a real phone

and runs on a general purpose computer,
rather than a dedicated device.
[Source: Wikipedia]
SIP or H.323 capable
Mostly freeware

It is a software implementation of SIP

user agent.
[Source : SIP Forum]

Softphones

Softphone: EyeBeam

EyeBeam versions are available under Windows and Linux

Softphones: Teleo & Damaka

Popular softphones
SKYPE
Can call regular
phones
Cheaper rates for
numerous international
destinations
P2P architecture
Proprietary signaling

Popular softphones (2)

Net2Phone
Can call regular
phones
Cheaper rates for
numerous international
destinations
Available at various
Internet cafes across
the country
Prepaid cards
available
SIP-enabled

Hard Phones use ATA

Analog Telephone Adapter

Also known as FXS

Gateway

Used to connect ordinary

phones to the IP network

Built-in SIP or H.323

capability

Both RJ-11 and RJ-45

interfaces

Cisco AT 186

Looking for ATA?

Analog Telephone Adapter (ATA)

Analog Telephone Adapter (ATA)

Mediatrix ATA-1104

ATA Connections

AblaEZ FXS Gateway

Connecting Ordinary Phone to IP Network

ATA
RJ-11

Ordinary Phone
RJ-45

IP Network
Router

What is an IP Phone?

LAN-ready phones
RJ-45 connection
to IP network
Built-in Ethernet
switch

Built-in codec

Support G.711 and

G.729a codecs
Cisco 7970 IP Phone

Looking for an IP Phone?

IP Phones

Altigen IP-600 model

Uniden 300 IP Phone

Connecting to IP Network

IP Network
Router
IP Phone

Phone-to-Phone
Cebu
ISP /
VoIP Provider
DSL Modem
3Com 3101

New York
Internet

Router

Mitel 5240

Packet Voice Protocols

Packet Voice Protocols

Signalling and Call Control
H.323
MGCP Media Gateway Control Protocol
SIP Session Initiation Protocol
H.248/Megaco

Media Transport
RTP Real Time Protocol

MGCP

Media Gateway Control

Protocol
Defined in RFC 2705
Defines a centralized
architecture for creating
multi-media applications,
including Voice over IP
(VoIP)

Megaco / H.248

Megaco defined in RFC 2885

H.248 is an ITU-T Recommendation

that defines Gateway Control
Protocol

Joint collaboration between ITU and

IETF

Defines a centralized architecture for

multi-media applications, including
VoIP

Extended many capabilities of MGCP

ITU-T H.323 Multimedia

Protocol

Packet Voice Protocols

H.323 (ITU-T)

Multimedia conferencing protocol

Voice, video and data conferencing over
packet-based networks

Accepted as standard October 1996

Uses both TCP and UDP for control

Utilizes RTP for transport

H.323 Stack

H.323 vs OSI

H.323 Components
H.323 Terminal
MCU Multi-

point Control Unit

Gatekeeper
Gateway

H.323 Components
Terminal

Endpoint device on a LAN

Computer or workstation

Supports real-time, full-duplex

communications with another
H.323 entity

Support audio codecs and

signaling (Q.931, H.245)

H.323 Components (2)

MCU or Multi-point Control Unit

Supports sessions between 3 or more

endpoints

Centralized conferencing: handles

signalling and stream processing

Decentralized conferencing : handles only

signalling; streams go directly to endpoints

Can be integrated into gateway,

gatekeeper or H.323 terminal

H.323 Components (3)

Gatekeeper

Provides connectivity to the LAN/H.323

terminal

Address translation

Admission control

Bandwidth control and management

Call authorization

Directory services

Implemented as software

Can be integrated to gateway

H.323 Components (4)

Gateway

Interworking device

LAN and the CS network

Controls call setup ad clearing

Performs compression and packetization of

voice

H.323 Call Flow

Session Initiation Protocol

(SIP)

Packet Voice Protocols

SIP Session Initiation Protocol
RFC 2543
Uses port 5050
Distributed architecture for creating
multimedia applications, including VoIP
Text-based lightweight protocol like SMTP

SIP message:
INVITE sip:esjabonete@smart.com.ph

SIP Key Functions

Name mapping and redirection

Translation of participants descriptive

naming information to SIP location
information

SIP Key Functions (2)

Capabilities negotiation

Various media capabilities of the participants are

determined by the SIP in order to assure
appropriate usage of media facilities during the
session

SIP Key Functions (3)

Management of session participants

Allows participants to control the

incorporation of new arrivals into a session or
the termination of existing participants during
a session

SIP Key Functions (4)

Capabilities management

Monitor the media capabilities during a session

and thus make the appropriate adjustments
when necessary

Source: SIP Forum

SIP Advantages

Session setup is out-of-band

Resource searching via locationindependent identifier

E.g. user@domain
Real-time : Faster than e-mail

Can reach multiple end-points

simultaneously or in sequence (forking)

SIP Entities
Logical SIP Entities:

User Agent

Proxy Server

Redirect Server

Registrar

Logical SIP Entities

1. User Agent

Endpoint entity
Initiate and terminate sessions

User Agent Client (UAC)

Application that initiates SIP

requests

User Agent Server (UAS)

Application that contacts the

user when a SIP request is

received

Workstations, IP-phones,
telephony gateways are
examples of SIP User Agent

Logical SIP Entities

2. Proxy Server

Intermediary entity
Acts both as server and client

Can make a request in behalf of

other clients

Requests are either processed

internally or forwarded
Stateful or Stateless proxy server
Stateful server remember

requests
Stateless proxy forgets about
the data after sending it

Logical SIP Entities

3. Redirect Proxy Server

Intermediary entity

Accepts and process SIP requests

Maps called party/address to 0 if unknown

Returns new address to client

Do not pass or forward requests to other

servers

Logical SIP Entities

4. Registrar
Makes it possible for users to alter the address
at which they are contactable
Accepts REGISTER messages
Updates location database of users

SIP in Action

SIP Flow

SIP Call Flow

IP Telephony Protocols
H.323

SIP

H.248 /
Megaco

MGCP

Standardization

ITU

IETF

ITU / IETF

IETF

Architectural Model

Peer-to-Peer

Peer-to-Peer

Master /
Slave

Master / Slave

Media Types

Voice, video,
data

Voice, video,
data

Voice, video

voice

Call Control

Gatekeeper

Proxy Server

MGC

MGC

Endpoints

Gateway,
terminal

User Agent

Media
Gateway

Media Gateway

Signaling Transport

TCP or UDP

TCP or UDP

TCP and UDP

UDP

Network Scope

Intra-, Extra-,
Internet

Intra-, Extra-,
Internet

Intranet

Intranet

Extensibility

Low

High

Medium

Medium

Scalability

Medium

High

Low

Low

Ease of Deployment

Low

High

Medium

Medium

SIP-T (SIP for Telephone)

Defined in RFC 3372

Encapsulation of ISUP in SIP

Translation of ISUP parameters to SIP headers

Designed to interface traditional telephone

signaling with SIP

Provide protocol translation and feature

transparency across points of PSTN-SIP
interconnection

Intended for use where a VoIP network interfaces

with PSTN

Media Bearer
(RTP)

RTP Real Time Protocol

RFC 3550

Uses port 5004 (UDP)

Provides end-to-end network transport functions

Suitable for applications transmitting real-time data

audio, video or simulation data, over multicast or unicast

network services

Two streams created for each conversation

Symmetric RTP uses same port for both streams

VoIP Packet

Real Time Control Protocol

RTCP

Port 5005

RFC 3550 / 3605

Based on the periodic transmission of

control packets to all participants in the
session

Underlying protocol must provide

multiplexing of the data and control packets

Other Protocols

Supporting Protocols

Supporting Protocols (2)

Session Description Protocol (SDP)
RFC 2327
Uses UDP
Session information

Bandwidth to be used by the session

Session invitation
Multicast/unicast transport port for media

Media information

Type of media: audio or video

Transport protocol: RTP/UDP/IP or H.320
Media format: H.261 video or MPEG video

Supporting Protocols (3)

Session Announcement Protocol (SAP)

Port 9895

RFC 2974

Used by session directory clients

Announcement is multicast with the same

scope as the session it is announcing

Other Protocols
Telephony Routing over IP (TRIP)
Defined in RFC 3219 / 2871
Internet protocol that supports the discovery and
exchange of IP telephony gateway routing tables
between providers
Policy driven dynamic routing protocol for
advertising the reachability of telephony destinations
Modeled after BGP-4 and enhanced with some linksate features of OSPF
TRIP uses BGP's
inter-domain

transport mechanism

peer communication

Other Protocols (2)

SIMPLE - Session Initiation Protocol for Instant
Messaging and Presence Leveraging Extensions
Application of the SIP protocol for server-toserver and client-to-server interoperability in
instant messaging
SIMPLE would best be suited for short,
disconnected text messages
analogous to a two-way pager; rather than for a
constant stream of communications

Other Protocols (3)

PINT- PSTN/Internet Interworking Service

Defines protocol for invoking certain telephone

services from an IP network

Services include placing basic calls, sending and

receiving faxes, and receiving content over the
telephone

PINT is specified as a set of enhancements and

additions to the SIP 2.0 and SDP protocols

CPL - Call Processing Logic

Defined in RFC 3050

Enables screening, forwarding, filtering, and

notification of types of services
Example:

CPL specifying calls from Joe are to be

forwarded to a messaging server after 5pm

All other calls are routed to a PDA

Voice Payload

Analog to Digital

Sampling

Quantization

Encapsulation

Packetization

Transport

Group or pack bits into small

containers
VoIP packet

Move packets to the destination

Re-assemble / Retrieve data

Voice payload

Common VoIP Codecs

ITU G.711
International standard for encoding telephone
audio on 64 kbps channel
8 kHz sampling rate
8 bits per sample
Mu-law / A-law companding

Common VoIP Codecs (2)

ITU G.722

Benchmark coder for wideband speech

coding

Uses Sub-Band Adaptive Pulse Code

Modulation

16 kHz sampling; 14 bits per sample

48 or 56 or 64 kbps BW

Used in speech storing,

multiplication and telephony

digital

circuit

Common VoIP Codecs (3)

ITU G.723.1

Designed for video conferencing / telephony

over standard phone lines

Optimized for real-time encode and decode

30 ms frames; 7.5 ms look-ahead

Part of H.323 suite

Dual coding rate 5.3 or 6.3 kbps

Highest compression ratio of any current ITU

standard

Common VoIP Codecs (4)

ITU G.726

ADPCM Adaptive Differential Pulse Code

Modulation
16 or 24 or 32 or 40 kbps BW

ITU G.728

Uses LD-CELP or Low Delay Code Excited

Linear Prediction
16 kbps BW

Common VoIP Codecs (5)

ITU G.729A

8 kbps BW

Conjugate

Structure

Algebraic-Code-Excited

Prediction (CS-ACELP)

10 ms frame; 2 frames per packet for VoIP

Low bandwidth demand

Popular VoIP codec

Linear

Common VoIP Codecs (6)

iLBC

Internet Low Bandwidth Codec

Defined in RFC 3951 and 3952

Royalty-free codec

Two modes

15.20 kbps or 13.33 kbps payload bit rate

303 bits (38 octets) per 20 ms frame

399 bits (50 octets) per 30ms frame

Basic quality higher than G.729a

High robustness to packet loss

Bandwidth Requirement

Bandwidth Calculation
Amount of bandwidth required to transport

voice over an IP network depends on:

Codec used
Sample period
Headers
Transmission medium
Silence suppression

Bandwidth Calculation (2)

Frame Size = BW / FPS

Where:

BW = bandwidth (bps)
FPS = frames per second

Example:

G.711 codec

64,000 bits per second

20 ms per frame (50 frames per second)
Frame Size = 64,000 bits per frame / 50 fps
= 1280 bits per frame
= 160 octets per frame

Packet Overhead

RTP/UDP/IP Overhead

RTP = 12 octets

UDP = 8 octets

Allows the samples to be reconstructed in the

correct order and provides a mechanism for
measuring delay and jitter

IP

Routes the data to the correct destination port

= 20 octets

Total = 40 octets

Layer 2 Overhead
Datalink Overhead

Usually Ethernet/802.3
Preamble 8 octets
Source & Destination MAC 12 octets
Length 2 octets
CRC 4 octets
Gap 12 octets

Total Ethernet overhead = 38 octets

Token Ring is not recommended for VoIP traffic!

Other Layer 2 Media for VoIP

PPP (Point-to-Point) 12 octets

Multilink PPP 12 octets

Frame Relay 4 octets

Voice Payload + Overhead

Bandwidth Requirement
Total Overhead = IP + Ethernet overhead

= 40 + 38
= 78 octets
For every 20 ms of speech:

1 Frame = 160 (voice payload) + 78 (overhead)

= 238 octets

BW Requirement = 50 FPS * 238 octets/frame * 8 bits/octet

= 95,200 bits per second or 95.2 kbps

Other Features

RTP/UDP/IP Header Compression

cRTP
Reduces 40 octets into ~2-4 octets

VAD Voice Activity Detection

Users do not necessarily talk at the same

time
Reduces ~35-50% of BW requirement

Voice Activity Detection

Exercise
1. Compute the bandwidth requirement for Voice
over IP traffic using a G.729A codec
a)
b)

Without VAD and cRTP

With VAD and cRTP

Compare results if codec used is iLBC (in

13.33 kbps mode) instead of G.729A

Codec Summary
Codec

BW
(kbps)

Sample
Period
(ms)

Frame
Size
(octets)

Frames
/Packet

Ethernet
Bandwidth
(kbps)

G.711

64

20

160

95.2

G.723.1

6.4

30

24

27.2

G.726

32

20

80

63.2

G.728

16

2.5

78.4

G.729a

10

10

iLBC

13.3

30

50

VoIP to GSM / PSTN

VoIP/PSTN Gateway
Used to connect a
VoIP network to PSTN
PSTN gateways
convert SIP or H.323
messages into ISUP
messages

VoIP to PSTN

VoIP to PSTN
SIP Phone

Proxy / Gateway

PSTN Switch

INVITE
IAM
100 Trying
ACM

18x

ANM
200 OK

CONVERSATION
BYE
REL
200 OK
RLC
200 OK

VoIP/GSM Gateway
ATEUS VoiceBlue from
2N Telekomnikace

Up to 4 GSM channels
SIP-based gateway
RJ-45 interface to IP
network

Codecs supported:
G.711
G.723.1
G.729A

VoIP/GSM Gateway (2)

Electronic Number Mapping

(ENUM)

What is ENUM?

IETF protocol that will assist in the

convergence of PSTN and the IP network

RFC 3761 (obsoletes RFC 2916)

Telephone number IN, URL OUT

What is ENUM?

(2)

Utilizes DNS

Using NAPTR (Naming Authority Pointer)

e164.arpa domain

Currently administered by RIPE

Future of ENUM rests on worldwide adoption of

DNS NAPTR

NeuSTAR is currently conducting a public

ENUM trial

How does ENUM work?

PSTN/VoIP Gateway

PSTN user phone dials +63322628784 (SIP-enabled phone)

At The Gateway
ENUM-capable gateway creates a DNS
query

Put dots in between numbers

6.3.3.2.2.6.2.8.7.8.4

Reverse the order of the string

4.8.7.8.2.6.2.2.3.3.6

Add the string e164.arpa to the end

4.8.7.8.2.6.2.2.3.3.6.e164.arpa
Looks like a domain !

Gateway Utilizes DNS

4.8.7.8.2.6.2.2.3.3.6.E164.arpa

DNS
The gateway sends a query for the domain

DNS and SIP

DNS receives the request

Sample Record
$ORIGIN 4.8.7.8.2.6.2.2.3.3.6.e164.arpa.

NAPTR 10 100 "u "E2U+sip "!^.*$!sip:jane_doe@smart.com.ph!" .

DNS Response

enum
SIP:jane_doe@smart.com.ph

DNS

Connecting . . .

Gateway will now be able to connect PSTN to

IP-phone

Using the response from the DNS, the gateway will

be able to locate SIP proxies

Proxy server invites the phone

sip: jane_doe@smart.com.ph

INVITE sip:jane_doe@smart.com.ph

If the called party accepts the call

Conversation begins

How does ENUM work?

Tokyo

enum

DNS
SIP Proxy

IP Network

INVITE sip:jane_doe@smart.com.ph

Cebu City

Voice Quality

Measuring Voice Quality

Subjective testing is the

most "authentic" method
of measuring voice quality

specialized and
costly process

Used by CODEC designers

and equipment
manufacturers to validate
VoIP technology before
deployment

Mean Opinion Score

Listeners rate voice sample

Rating used: (Scale of 1 to 5)
1 = unacceptable call
5 = excellent call

Typical range = 3.5 to 4.3

MOS of 4.0 or greater is generally thought to

be of PSTN Quality

E Model

Standardized by ITU G.107

Originally developed by ETSI (ETR 250)

Objective: to determine a quality rating that

incorporated the "mouth to ear" characteristics of a
speech path
Range of R factor = 0 to 100 with < 50
unacceptable

R = Ro - Is - Id - Ie + A + W

E Model (2)
R = Ro - Is - Id - Ie + A + W
Ro

Base factor determined from noise,

loudness, etc

Is

Impairments occuring simultaneously with

speech

Id

Impairments are that are delayed with

respect to speech

Ie

Equipment impairment factor

Advantage factor (convenience of user to

make the phone call)

Wideband connection factor

VoIP Quality
Main factors that affect voice quality

Delay

Echo

Packet Loss

Jitter

Delay

ITU-T G.114 Recommendation

300 ms round-trip

Can be extended to 200ms without

significant loss in quality (Lab Tests)

End-to-End Delay

Types of Delay

Propagation delay

Caused by the characteristics of the speed of light

traveling through fiber-optic materials or copper
media

Handling delay

Also called processing delay

Includes compression delay, packet-switching delay,

and packetization delay

Serialization delay

The time it takes to actually place the bits onto the

interface

Delay vs MOS

Echo
Echo is a problem in voice-over-packet networks
because delay is always > 25 ms
Networks use echo-cancellation techniques to
minimize echo

Sound Files

Sample #1: Network ~200 ms delay

Sample #2: Network ~400 ms delay

Packet Loss
Networks should have <1% packet loss
Packet loss results in degraded voice

quality

Various models used in teletraffic

engineering:

Bernoulli model
Gilbert-Elliot model
Markov model

Packet Loss (2)

Packet loss can be compensated using

PLC (Packet Loss Concealment)

technique
Replaying last packet received

Applying a sophisticated algorithm to

generate speech based on previous data

Can compensate up to 20% packet

loss

Sound File

Network has ~ 5% packet loss

Packet Loss vs MOS

Packet Loss vs MOS (2)

Jitter

Jitter is a variation in packet transit delay

caused by queuing, contention and
serialization effects on the path through the
network
(Source: VoIP Troubleshooter)

Higher levels of jitter are more likely to occur

on either slow or heavily congested links

Types of Jitter

Type A constant jitter

Constant level of packet to packet
delay variation

Type B transient jitter

Substantial incremental delay

Type C short term delay variation

Commonly associated with congestion
and route changes

Common Causes of Jitter

Sending system packet scheduling

In softphones, application has to

contend with CPU time

LAN Congestion

Access link congestion

Happens on serialization

Routing table update

Route flapping

Jitter Effect

Jitter Buffers

Can be offset using Jitter Buffers

Fixed buffer
Constant size
Adaptive buffer

Adaptive jitter buffer able to maintain constant delay variation.

VoIP Regulation

VoIP and the NTC

Should NTC regulate the use of VoIP?
Telecommunications any process which
enables a telecommunications entity to
relay and receive voice, data, electronic
messages, written or printed matter, fixed
or moving objects, words, music or visible
or audible signals or any control signals of
any design and for any purpose by wire,
radio , or other electromagnetic, spectral, or
optical or technological means.

VoIP is what?
NTC: VoIP is Value-Added Service
RA 7925 has no explicit definition what Value Added
Service means.
Value Added Service Provider an entity
which relying on the transmission,switching
and local distribution facilities of the local
exchange and inter-exchange operators, and
overseas carriers, offers enhanced services
beyond those ordinarily provided by such
carriers.

NTC Draft Rules

Section 1
Voice over IP (VOIP) shall be classified as a
Value Added Service within the contemplation
of RA 7925 otherwise known as the Public
Telecommunications Policy Act.

NTC Draft Rules (2)

Section 3
Any person or entity seeking to provide
VoIP for use by the public for
compensation shall register themselves
with the Commission prior to operation
as a VoIP provider.

[ Source: http://www.ntc.gov.ph ]

VoIP Challenges

Quality of Service (QoS)

Traditional circuit-switched networks

reserves end-to-end resource before
connection

MPLS Multi-Protocol Label Switching

could change that

VoIP Challenges

Emergency Services

Big issue in the US

911 and E911 services

Vonage.com and Pulver.com affected

by initial FCC ruling regarding E911