12 vo ip-t-hutchinson-11feb2011

1

Carleton University

February 11th, 2011

Voice over IP

Presenter: Tony Hutchinson

System Engineering Manager

February 11th, 2011 Slide 2

Voice over IPCarleton University

Biographical Information

Tony Hutchison

Expertise:

VoIP and network design

PBX Design, TDM, ISDN, Ethernet, PSTN (PRI, BRI and Analogue), EMC, Safety

Telephony and data, TDM and PDH design

Current Position - 1998 to Present

System Engineer Manager – Mitel Networks (Canada)

VoIP design, PBX, Hosted Services, Network Design

Technical interface with RnD and customer facing Sales/System Engineers

Previous Positions (UK)

Telecom Sciences – SME PBX System Engineer

Philips – SME ISDN PBX System Designer (for the global market)

GEC – Transmission and Multiplex system (analogue and digital design)

Education

Birmingham University (UK): Electronic and Computer Engineering (Hons.)



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



Executive Summary

Business Case

VoIP

Toll Quality

?

Security?

Not In

tern

et

Convergence

of cabling

Moves, Adds,

Changes

Business Expansion

Enterprise/Branch office

Distributed BusinessHot DeskingResilie

ncy

Toll Bypass

Cut acrossgeographies

WorldCom

TDM Backup

Two towers NY

IP-Trunks

Cost ofdedicated

links Cost ofinfrastructure

Traffic rates

Line cardsVs L2

switches

Structurdwiring

Convergence

Voice/Data

Video

Conference

CTI

InteractivePresentation/

Passive

Services

Dial Tone

Hook Switch

Soft Phone

Teleworker

Road warrior

VPN

Voice Mail

UnifiedMessaging

PDA integration

BusinessApplications

Hotel

Security

Video/Cable

Infrastructure

SIP

Megaco

Signalling

ProprietaryFeatures

ManagedNetwork

SLA

VPN

ISP Access

MPLS

MAN

WAN

Fixed link

FrameRelayHistory

Lots Learnt

Lots

Forgotten

Signalling

Protocols

2W/4W

Echo

Sidetone

Acoustics

Ergonomics TransmissionLevels

LocalInternational

Challenges

Internet

Local

ISP

ManagedNetwork

OSI 7 Laye

rVoice

Quality

Toll

MOS

PESQ,PSQM

Maintaining

Guarantee

Bandw

idth

CODEC

G.711

G.729

G.726

WidebandVAD

VLAN

PriorityTOS

Diffserv

E911

Hybrid

Echo

Delay

Security

DOS NAT

Firewall

Encryption

PacketSize

Overhead Traffic

Delay

Jitter BufferConnectionless

PowerNetwork

impairments

Packet Loss

Culture

DataWorld Telecom

World

Training

NewStandards

Demarcation

Clock Sync

FAXIn-BandData

MODEM

POS

H.323

24/7

24/7

Agents

Presence

Real Internet

CableTV

Telephone

Numbers

GeographicIndependence

IPTDM



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



History

There has been much experience learnt in 100 years

Some is so common place, it has been forgotten

With IP some of these lessons need to be re-learnt

Echo was previously just louder side-tone

Added delays now affect conversation quality

Network Clocks were previously well defined

Data path wasn’t lossy, with potential gaps in speech



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



Business Case

So why all this interest in IP? Isn’t it just another transport medium?

Yes

Connectionless

Not constrained to a physical location

Path between two user points is not pre-defined, can change dynamically

Bandwidth is only consumed when needed

Cost Alternative

The long haul carriers (e.g. AT&T) are already carrying data traffic in their large networks (at a lower cost)

So, send voice as data and pay less!

Why Now?

Moore’s Law

Cheaper Processing

More readily available



Business Case

So why deploy IP rather than TDM?

Easier and cheaper maintenance: Integration of data and voice onto one network

Consolidation of trunk access to a central SIP gateway (IP) across the business

Lower operating costs: Integration of remote offices over a common corporate data network, rather than through PSTN. Single Dial Plan.

Access from anywhere: Power users such as Teleworker and sales ‘Road Warrior’. Global Access

Lower product costs: Integration of a voice application onto a central server, e.g. voice mail, means reduced number of devices. The remote sites no longer need their own local VM.

Security, Resiliency and Availability: In NY (September 11th) the IP infrastructure kept running; the PSTN didn’t

Future applications will be data centric, e.g. “Presence”

Displacement of current TDM systems and businesses



Business Case

There are still reasons for both IP and TDM to live together

Legacy devices are still going to be around (for some time) and people will still use these, e.g. FAX, remote MODEM

TDM is still likely to be the connection to the PSTN

Most businesses have a directory number via the PSTN. Not all have a fixed IP address.

By 2017, expected that most new PSTN core installations will be IP only

Mobile and 4G will increase VoIP uptake in the next 7-10 years.

By 2014 expected ratio is 5 mobile to 1 land-line connection

Existing landlines are being bundled with IP access for “last mile” access

~50% of mobile connections by 2020 expected via IP = $345Billion!

~30% of mobile IP connections expected through Google, Facebook, Yahoo

Largest growth area (mobile/smart phones) expected to be Asia Pacific



Business Case

In the Business PBX space three main tiers are emerging:

Managed Hosted Centrex

Globally the uptake is increasing, with predictions beyond $16billion

That’s a big market, and competition is fierce!

Hosting offers opportunity for VoIP without local “boxes”. High growth sector, but still early adopter cycle

Wireless connections and new data modes allow IP connections to be provisioned much easier in countries where it has traditionally been difficult to provide standard telephone cables and wires.

Global Service Revenues

0

2,000

4,000

6,000

8,000

10,000

12,000

14,000

16,000

18,000

2007 2008 2009 2010 2011 2012

Year

Rev

enue

s ($

mill

ion)

IP centrex service rental

Hosted IP PBX service rental

Managed IP PBX service rental

Total service revenues



Business Case

Largest revenue split today (Business Phones)

Americas Europe

Largest growth sectors:

Latin America Eastern Europe MEA Many smaller countries just

adding IP infrastructure

Slowest growth sectors:

North America Western Europe But it’s still growth!

Revenue Split 2011

NA

LA

W.Euro

E.Euro

MEA

Asia-Pac

ROW



Agenda

Executive Summary

History

Business Case

Services/Content

Convergence

Infrastructure

Challenges



Services/Content

What services are people looking for?

Basic hook-switch and dial tone

Call handling features

Advance features such as call centres, agents

Remote location, e.g. Teleworker, Remote Agent

Networking between sites

Virtual Private Networks

New features such as voice recognition

Integration with current applications such as customer accounts, hotel registration, etc.

Business Process Improvements

Unified Communications and mobility, including Fixed Mobile Convergence



Services/Content

Today the industry is comfortable at the level of V1 applications

Biggest features are Toll Bypass and Networking

Early adopters are now taking V2 and V3 applications

Remote workers and Applications that don’t require access to the office

Remote ACD, help desks, etc

“Road Warriors” - Sales

Service Personnel

Mobility integration

Common access number for all connections

By 2011 it is expected that in the US, 41% of companies will be IP only, using SIP trunks in favour of TDM

Aff

ect

on

b

usin

ess



Services/Content

Centralized Unified Messaging

Globally Accessible

Integrates with E-mail and mobile services

Presence and call routing

Redirection of calls based on time, availability and caller to different end points

Integration with Microsoft, e.g. Lync™ 2010, or LCS and existing user databases, e.g. Active Directory

Fixed Mobile Convergence

One number - able to pick up calls at desk and mobile, or alternative number

Switchover between mobile carrier and in-house Wireless LAN

ACD and call routing

Service is handled by same agent to give more personalized service

Agents located globally - full language support

Speech Recognition

Redirection of calls based on user spoken words

E-Business

Workforce is distributed, and mobile.

Inventory tracking, e.g. RFID tagging

On phone Advertising, e.g. hotel

Business Process ImprovementBusiness Process Improvement



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



Convergence

What do we mean by convergence?

Combining of different worlds

Different mindsets and cultures

Different set of standards

And why now?

Processing power is cheaper - Moore’s law!

Phones have more power today than early PCs

PCs and phones are standard desktop tools

Voice and data networks can be combined to ONE

Phones can now interact directly with data devices



PSTN, Mobile/CellCircuit Switched

$455B

Internet, IP DataConnectionless

$18B

FR, ATM, PrivateLine

ConnectionOriented

$18B

Cable TV$75B

VoIP VoDSL

Convergence

Four main business areas are converging

Voice, TV, VPN and Data

Triple Play

Broadcast TV - 100% users

Telephony - 100% users

Internet - 40% users and up

Voice is still the biggest revenue earner

Incumbents need to grow and expand

Many Cable TV providers now offer IP connectivity, many also voice.

New IP providers: Hosted VoIP, SIP Trunks

Courtesy: ATM Forum



Convergence

Business A Business B

Replacement of Local Loop for Voice

LANLAN

Long Distance PSTNe.g. AT&T

CO,E.g. Verizon CO,

E.g. Bell

IP Network 1

SIP Trunk Gateway

SIP Trunk Gateway

SIP Trunk Gateway

Existing TDM

IP Network 2

SoftSwitch

Peer2Peer BGP Router

Existing IP

Usage

Migration



Convergence

Convergence in the network is unseen by the user.

What does the user see at the access point?

Two line jacks into ONE?

In reality, once installed, building wiring isn’t removed

On new installations, it’s cheaper to pull too many wires, than not enough

Integration of ServicesIntegration of Services



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



Infrastructure

What are the building blocks of the system and how are these connected?

Common Architectures and voice media paths

Signalling Protocols

Network Interconnections



Infrastructure

The voice media paths and switching define the type of system. Three main types are defined:

IP Enabled PBX Here a line card is simply replaced by an Ethernet card. Voice switching

is done in TDM. This is not scalable and adds unnecessary delay.

Hybrid PBX TDM and IP are handled equally, only traversing a gateway when IP and

TDM devices need to connect.

Typical in an SME/Enterprise environment

IP-PBX (Hosted) All switching is done in IP. TDM connections are generally only to the

PSTN via external gateway, which may be off-site.

Model used for Hosted services

G/W

IPPhone

IPPhone

TDMPhone

TDMPhone

Hybrid

G/W

IPPhone

IPPhone

TDMPhone

TDMPhone

IP-Enabled

G/W

IPPhone

IPPhone

IP-PBX



Infrastructure

Basic VoIP system building blocks

Gateway between IP and TDM

Media Gateway Controller

Call Control

Features and Services

End users

Different protocols use different names, but functions are essentially the same

Peer to Peer or Central Control?

Central is good at resolving resource conflicts

Peer to peer is resilient to network failure

SIP can handle both aspects

MediaGateway

MediaGatewayController

FeatureServer, e.g.Voice Mail

CallControl/MediaServer

IPPhone

IP

PSTN

IPPhone

MediaStreaming

Signalling



Infrastructure

Signalling Protocols are numerous and include:

H.323

MGCP/Megaco

SIP

Proprietary

Why so many Signalling protocols?

Different starting perspectives of the requirements

They all offer some advantage for different users

Most are evolving as new features start to roll out



Infrastructure

H.323

Overview specification and includes: H.225 - Signalling H.245 - Media streaming TCP/IP and RTP/UDP/IP

One of the early protocols Standards based, uses current ISDN technology, works well for

interoperability between vendors Features are basic, but well proven Well proven ground rules about interoperability Centralised call control, based on known proven techniques, call state aware Slow to evolve Difficult to scale to millions of users Central call control = single point of failure Telephone routing biased rather than at application level



Infrastructure

MGCP/MEGACO

MGCP was initially a proposal to IETF for a stateless gateway protocol, it has similarities to H.323, and has the ability to evolve

Combined forces with ITU to create MEdia GAteway COntrol

Similar to H.323 in content, but reduced messaging

New standard and evolving

Allows central and distributed call control access to a gateway

Was thought to be the front runner with Enterprise business but little is heard

Difficulties again in scaling from a global view. Different gateways need different controllers which need to intercommunicate.



SIP (Session Initiation Protocol), RFC2543

More Client Server based and allowing Peer to Peer interaction. Call control can be distributed End devices need to be more intelligent than simple phones Has the ability to evolve quickly, and scale to large numbers Simple protocol, but lacks certain PBX capabilities Vendor specific options provide features Inter-vendor working is usually determined through “bake-off” but improving

as more vendors implement agreed solutions Networking features low, but improving Open Standards through IETF, agreed by many established industry leaders Continual proposal of new features and extensions SIP Extensions to include “proprietary” features to make them more

mainstream

SIP is the Internet Phone signalling protocol of choice

Infrastructure



Infrastructure

Business 1

Business 1

Service Provider 1

Internet

Service Provider 2

Business 2

Local Network Global Network

Local Network Management , one point of contact

Global Network Management, many points of contact

Common single private address space

Mixture of local private and public address spaces with overlapped addresses

Local QoS control No Guarantee of Qos or Service Level

Limited protocols Many protocols



NAT

ALG

Private IP Address Space

Public IP Address Space

Infrastructure

Firewalls

Used to keep out unwanted access

Restricts flow of data both ways, including voice

Network Address Translation (NAT)

Maps many internal private addresses to limited number of public IP addresses

NAT is typically not application aware

VoIP media and signalling may include private IP addresses in messages which will be confusing externally in public IP space

Application Level Gateway (ALG)

Stateful and knowledgeable of protocol, e.g. SIP

Can translate private/public addresses within messages

NAT and IPv6

NAT and ALG will not be needed

Any device can access any other device in both public and private address space

Truly global access- one large address space



VoIP

Infrastructure

Carrier/SP

PSTN

LAN

LAN

SIP Trunk Gateway

Internet

SIP ALGLAN

Carrier2

Border Gateway

Architecture of SIP in a large carrier deployment

• SIP ALG provides IPv4 NAT and firewall functions for SIP (a.k.a. Session Border Controller (SBC))

Hosted

SIP ALG

SoftSwitch

Public IP

Private IPPrivate IP

Private IP

Public IP

Public IP



Industry Trends

SIP Trunks

SIP User

Network SP provides phones

Network SP provides end-end IP

IPv6 provides everyone with a global address

SPs compete on a global scale

Infrastructure

With IPv6 all devices can be addressed globally

Removes need for NAT and SIP Proxies (ALG), making global connections possible

For example: call control in NA, gateway in Asia, IP phone in Europe!

SIP is becoming an accepted global standard for IP media device signalling

SIP and IPv6 have the potential to become disruptive technologies in displacing the current (TDM) telephone network systems

Today



Agenda

Executive Summary

History

Business Case

Services

Convergence

Infrastructure

Challenges



The ChallengesMany!

There are many…

Voice Quality

Delay, lost data, jitter, echo

Network issues, non deterministic, connectionless

Bandwidth, packet overhead, queue delays

Clock synchronisation

NAT and ALG for “off-net” connections

Security

Emergency Location E911

IP address space AND translation

End points need to use the same media format, or CODEC



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95

The ChallengesVoice Quality - Metrics

To a User - It’s a Phone!

Voice Quality Metrics

Toll Quality

Mean Opinion Score (MOS) of 4.0 or better

E-Model with R=80 or better

Output based on many inputs:

Delay

Levels

Echo

Background noise

CODECR=88

Continued Voice Quality is expected Continued Voice Quality is expected



The ChallengesVoice Quality- Delay and Loss

Voice Quality

With good echo cancellation techniques

End to end delays of ~150ms are tolerable

1% packet loss with good Packet Loss Concealment is also tolerable

Jitter only becomes significant when it results in packet loss

Jitter buffer balance between adding delay and introducing packet loss

G.711 - QoS Versus Delay and % Packet Loss

40

50

60

70

80

90

0 100 200 300 400 500

Total One Way Delay (ms)

QoSR Value

0% PL

1% PL

2% PL

3% PL

1% PLC

2% PLC

3% PLC

Satisfied

Some User Dissatisfied

Many Users Dissatisfied

Nearly All Users Dissatisfied

Not Recommended

55 dB

Far EndEcho Loss

Note: Above 200ms an additional 20ms delay is worse than 1% packet loss with PLC.

Some Delay is tolerable Some Delay is tolerable



The ChallengesVoice Quality - Echo

De-packetisation

Packetisation

SpeechDecode

SpeechEncode

Jitter Bufferand Packet

LossConcealment

Echo Canceller

NLP

EchoPrediction

D / A

A / D

Electrical Coupling

IP Gateway End Point

De-packetisation

Packetisation

SpeechDecode

SpeechEncode

Jitter Bufferand Packet

LossConcealment

Echo Canceller

NLP

EchoPrediction

D / A

A / D

Acoustic Coupling

IP-Phone End Point

Echo is always present, even in TDM

Delays in IP make this more noticeable

IP

IP

Control of Echo is importantControl of Echo is important



The ChallengesVoice Quality - Delay

Let’s look at where delay occurs

Fixed Delays in CODECs and filters

Packet size delays to build a packet

Jitter Buffer

Network (which also introduces jitter)

End to End Delay = 79ms, but with 10ms jitter (router)

3ms

CODECFilters

2ms

20msPacket Creation

40msJitter Buffer

2ms

L2Switch

2ms

1-10ms

RouterQueue

1-10ms

2ms

L2Switch

2ms

2ms

CODECFilters

3ms40ms

Jitter Buffer20ms

Packet Creation

Network

Control of Delay is importantControl of Delay is important



The ChallengesNetwork Jitter

Where does jitter come from?

Serialization delay: Waiting for larger packets to transfer

Lack of Priority means all data is treated equally - First in First out

Apply priority queues for voice and set MTU to cut large packetsVoice 1 Voice 2 Voice 3 Voice 1 Voice 2 Voice 3

Data

Input

O/Pw/o

MTU

Voice 1 Voice 2 Voice 3 Data Voice 1

Delay x ms

O/PwithMTU

Voice 1 Voice 2 Voice 3 Data1 Voice 1 Voice 2 Voice 3Data2 Data3

MTU Breaks up large packets

Priority mechanism to get voice into gap first

Use QoS settings to prioritize voice and minimize jitterUse QoS settings to prioritize voice and minimize jitter



The ChallengesNetwork Jitter

Removal of jitter

Voice CODECs run at a constant rate

Too much or too little will result in a gap

Small gaps in voice are not discernable <60ms

Small gaps in tones are discernable

Jitter Buffer needed = Leaky Bucket

Packet Loss Concealment hides loss

Fill gaps with noise, silence

Remove data in fixed size, during silence

Packet Arrival

Buffer Fill

Jitter Range

Jitter Buffer = ‘Leaky Bucket’

PLC Hides lost packets

Jitter Buffer = ‘Leaky Bucket’

PLC Hides lost packets



The ChallengesClock Slip

Clock Slip

The CODEC at each end may run at 64kbits/s, but they have a tolerance

No clock synchronization, therefore need to add or drop data

Example of packet drop due to slip Suppose two device, each at 50ppm (TDM tolerance)

That’s 100 bits drift in 1 million bits, or

8 bits in 80,000 bits which = 1 bit every 1.25 seconds @ 64kbits/s, or

1 packet (160 bytes) every 3 minutes, 20 seconds

Clock slip buffer needs to consider this drift up and down

Often, slip correction is included with jitter buffer control to minimize media delays and complexity of multiple buffers

Clock Slip

Fast Clock

Slow Clock

Clock Slip needs to be consideredClock Slip needs to be considered



The ChallengesTransmitting Tones

Transferring tones is problematic if jitter buffer discards

A DTMF tone need only be 75ms long. Losing 20ms from this is significant, results in

No digit being detected, or double digits

Big deal? ‘91’ gets you out of a PBX, double digits get you ‘9911’, I.e. emergency services!

DTMF information can be sent ‘in-band’ as an RTP datagram using RFC4733

Call Progression tones can also be sent as descriptions using RFC4733

New standards RFC4733 and RFC4734 supercede RFC2833

RFC4733 ensures DTMF tones are transferred correctlyRFC4733 ensures DTMF tones are transferred correctly



The ChallengesFAX and Modem

In band tone transmission

Other devices use in band tones, such as:

FAX and MODEM

FAX will work, but only under very controlled network conditions, such as packet loss

MODEMs will work, but again under controlled conditions such as echo cancellation

Alternative CODEC for FAX is T.38 (and less often T.37)

Alternative CODEC for MODEM (V.150) is under investigation

Proposals have been made, but due to complexity there is currently little enthusiasm to include this in gateways.

Limited (proprietary) solutions are available.

FAX and MODEM need alternative CODECsFAX and MODEM need alternative CODECs



The ChallengesPacket Size

How big a packet should be used?

Consider bandwidth use with different payload size and overhead

10 to 50ms is a good size to use for voice

Below 10ms: more bandwidth for payload is needed

Above 50ms: voice delays cause quality issues

Good compromise is 20-30ms, many people fixing on 20ms.

Some administrations using 10ms to decrease user-user delay

Other issues also appear:

Smaller packets mean more Packets Per Second (PPS)

Wireless connections, especially WiFi, have difficulty with high PPS rates

Preference is for larger packets, but this adds more voice delay

20ms Packets - Good Compromise20ms Packets - Good Compromise



The ChallengesCODEC

So many CODECs, which one to choose?

G.711

This is the base level, so must be in (defined in H.323, and defacto)

G.726

Good voice quality, but limited bandwidth reduction

G.729, G.729a, G.729ab

Good reduction in bandwidth, with good voice quality

729a is reduced MIPS in conversion

729ab only sends with voice activity, so even less bandwidth, but voice may be clipped

Wideband (G.722 and others)

Works especially well for conferences, offering 8kHz voice Bandwidth

Balance of Voice Quality and Bandwidth usageBalance of Voice Quality and Bandwidth usage



The ChallengesBandwidth

How much bandwidth needed?

Payload

G.711: 160 Bytes (64kbps)

G.729: 20 Bytes (8kbps)

8:1 compression in payload

Overhead: RTP, UDP, IP, MAC and Ethernet + interpacket gaps (dead space that can’t be used)

With overhead only 5:2 ratio

CODEC Type Bytes per Packet BandwidthG.711 242 96.8kbits/sG.726/32k 162 64.8kbits/sG.729 102 40.8kbits/s

G.711 ~ 100kbits/s

G.729 ~ 40kbits/s

G.711 ~ 100kbits/s

G.729 ~ 40kbits/s



WAN/InternetLAN

The ChallengesNAT and ALG (Off network connections)

Private IP Address Space Public IP

Address Space

NAT (Translates header addresses between Private and Public networks)

ALG (Protocol Aware and translates messages as well)

10.10.1.1

2.3.4.55.6.7.8

SA DA Message10.10.1.1 5.6.7.8 Send Voice to 10.10.1.1



NAT Only

NAT and ALG



The ChallengesSecurity

Security:

How accessible is the equipment

Put a lock on the door!

How robust is the system to attack, DOS?

Harden system to cater for fault conditions as well as normal operation.

Authentication (Who is this?)

Authorization (Is this action allowed?)

Encryption (You can’t see this, well not easily)

Integrity (Did someone tamper with this?)

Phreakers gaining access for free calls, or charging others

Provide separate access, e.g. separate physical connection

Remove ‘backdoors’

Ring-back on MODEM

Lock the Door! Lock the Door!



The ChallengesSecurity

Security

Monitoring and substitution of voice

UDP has no ACK/NACK, can be substituted, redirected

Encryption, use of public and private keys

DES, DES-3, RC-4, AES, SSH, SSL, IP-SEC, etc.

Legal issues and Intellectual Property in distribution and use of encryption

Access through firewalls

Open up ports, but this makes it ‘look like a pin cushion’

Use a Session Border Controller, or Application Level Gateway, dynamically opens ports as needed.

VPN between sites, but not to Internet direct

Understand where data may be public and safeguard access and read rights

Understand where data may be public and safeguard access and read rights



The ChallengesRules and Regulations

Emergency Location (E911)

Emergency Location (E911) requires that a person making an emergency call can be physically located within a pre-defined area

IP phones can move and be located globally

These requirements are potentially in conflict

New global standards and regulations are evolving to maintain this capability

IETF-ECRIT : “Framework for Emergency Calling using Internet Multimedia”

CALEA

Call Tracing, Malicious call handling

Wire-tapping

Charging for services

Who pays? The Internet is ‘free’ But, is it?

Local and Global rules need to be

applied

Local and Global rules need to be

applied



The ChallengesIPv6

IPv4 Public Address

The current public address range is running out

Main users are NA and Europe

Insufficient for ROW

Exhaustion Prediction:

IANA allocation: Feb 2011

Regional Internet Regions: Nov 2011

IPv6 Public Address

Driver: 3G/4G wireless, internet connected appliances

Already being deployed in a number of countries IPv6 is here! IPv4 is

running out

IPv6 is here! IPv4 is running out



Finale

VoIP is mainstream Business Process Improvement, rather than networking and toll bypass Mobility and Unified Communications

Technical challenges for voice quality are being overcome

The large Telecos are changing to embrace the IP changes, e.g. BT 21CN

SIP is becoming a common communication method and feature interaction between vendors is improving

IPv6 is being implemented to provide truly global communications

SIP and IPv6 are disruptive communication technologies Many business and global changes expected because of these Many carriers providing voice, data and now IP Voice services

Thank You

12 vo ip-t-hutchinson-11feb2011

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