voice over ip for carriers. 2all rights reserved © 2001, alcatel agenda advantages of packet...

Voice over IPfor Carriers

2 All rights reserved © 2001, Alcatel

Agenda

Advantages of packet switching for voice communications

VoIP applications

VoIP technology overview

VoIP standards

Quality-of-Service in VoIP networks

Addressability in VoIP networks

VoIP regulatory considerations


What is VoIP?

Technical answer:

“the ability to make phone calls over IP-based data network”

Commercial answer:

”the Multi-Billion Revenue Opportunity for the 21st Century”

VoIP > IP Telephony typically “IP Telephony” indicates using IP terminals most VoIP is between normal telephones

VoIP < “Voice over Packet” includes Voice over Frame Relay, ATM


Circuit switching served voice wellfor 100 years!

Transmission circuits and switch path assigned during call setup for the duration of the call

Call blocks if not enough network resources available Essentially one class of service: 3.5 kHz, 64 kb/s Poorly matched for bursty data transmission

User - A User - B

LoopTrunkGroup

CentralOffice - A

CentralOffice - B

Signal System 7Data link

Signal TransferPoint

TransitOfficeClass 5

Switching System Connection ThroughSwitching Fabric

Class 4Switching System


Packet SwitchingWell-matched for data transmission

Great fit for bursty data transmission! Packets sent at full rate of transmission facility Supports variable information transfer rates Resources not consumed when nothing to send Potential to eliminate call setup phase

But … Transmission capacity used for header Buffering introduces varying delays

HeaderPacket

PayloadInput Buffer

Output BufferHdr. Trans

Hdr. Trans

RoutingFabric


VoIP Network Architecture

Media gateways provide voice packetization Media gateway controller provides call control logic Signaling gateway provides interworking with SS7

signaling of PSTN

IPnetwork

MediaGateway

MediaGateway

Controller

MediaGateway

PSTNnetwork

SignalingGateway


Advantages of VoIP

Regulatory arbitrage (e.g., lack of access charges) Low entry barrier competition with incumbent carriers Cheaper switching systems

Per Gb/s, IP routers cheaper than TDM Class 5 switching systems

Ability to operate one network for voice and data Cost savings through use of

low-bit-rate voice Ability to offer more complex services

E.g., Multimedia, multiport calls Intelligent terminals (e.g., PC)

Better (graphical) user interface Clean slate design:

Separation of feature intelligencefrom switching fabric supplier

Self-provisioning networks


PSTN Vs VoIP Network Costs

Network costs (transmission and switching costs) contribute only 10-15 % of overall cost of a voice call terminated by an ILEC or a PTT, and 20-30% of overall costs for calls not terminated by a ILEC or a PTT

Of the network costs, switching costs range between 50 % of network costs for domestic calls to 15 % of network costs for international calls, transmission costs contributing the rest

Negligible savings in transmission costs through the use of VoIP: lower bandwidth for VoIP offset by need for over-provisioning bandwidth to ensure quality

TDM Switch costs in traditional PSTN replaced by cost of Router plus cost of Gateway [GW] plus cost of Gateway Controller [GWC] plus cost of new OSS/NMS/Billing /Provisioning

No network cost savings, and very likely a cost penalty, in the initial years, in going from PSTN voice to VoIP for public networks


PSTN versus VoIP

Today’s PSTN VoIP

Underlying Technology

TDM circuit switching Packet switching

QoS guarantees Yes No

Network resource reserved at call setup

Yes No

Network elements Class 4, Class 5 switching systems

Gateways, gateway controllers, routers

Call processing intelligence

Mostly integrated in switching system

In separate gateway controllers

Bandwidth per call 64 kb/s Variable 8 – 32 kb/s

Signaling DTMF, SS7 SIP, H.323

Transport TDM in access, edge, core

ATM, FR, native IP in access; ATM native

IP in core

How reliability achieved

Redundancy within each network

element

Redundant routes through network


VoIP versus Voice-over-the-Internet

Voice-over-the-Internet

No bandwidth guarantees

No prioritization of traffic within network

All traffic receives “best effort” service

Each Internet user is at the mercy of all other users

Voice quality ranges from acceptable to atrocious

However

Internet technology continues to evolve (e.g., IPv6)

Development of Next Generation Internet


What does “Carrier Grade” really mean?

“Five 9’s” reliability (down time of 5 minutes a year) Full redundancy of electronics, power supplies, fans, etc. No down time for upgrades or maintenance

Accounting and billing capabilities Interoperability with legacy telecommunications

equipment Feature parity with equipment it replaces Service quality measurements Support for CALEA, unbundling, and other governmental

mandates NEBS compliance for operation in central offices

Both safety and performance requirements Scalability to millions of subscribers Integration into the myriad of Operations Support

Systems


VoIP market

Voice over Internet Protocol (VoIP) gateway sales will increase 280 percent during the next five years, reaching $3.8 billion in 2003, according to research by Cahners In-Stat Group.

Voice over Internet Protocol (VoIP) gateway sales will increase 280 percent during the next five years, reaching $3.8 billion in 2003, according to research by Cahners In-Stat Group. IP TELEPHONY OVER LAN MARKET FORECASTED TO GROW

138% AVERAGE ANNUALLY OVER NEXT 5 YEARSSeptember 22, 1999 - IP Telephony

[IP PABXes], according to a study from The Phillips Group-InfoTech, will spawn a $1.9 billion industry by the year 2004 with an average annual industry growth of 138 percent over the next 5 years.

IP TELEPHONY OVER LAN MARKET FORECASTED TO GROW 138% AVERAGE ANNUALLY OVER NEXT 5 YEARSSeptember 22, 1999 - IP Telephony

[IP PABXes], according to a study from The Phillips Group-InfoTech, will spawn a $1.9 billion industry by the year 2004 with an average annual industry growth of 138 percent over the next 5 years.

IDC Forecasts IP Telephony Market Will Soar to 2.7 Billion Minutes of Use and $480 Million in Revenues by Year end

1999Business Use Will Accelerate in 2001September 1, 1999 - The worldwide Internet protocol (IP) telephony will explode from 310 million minutes of use in 1998 to 2.7 billion by year end 1999. By 2004, IP telephony minutes will reach 135 billion. Revenues for this service will skyrocket from $480 million in 1999 to $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004.

IDC Forecasts IP Telephony Market Will Soar to 2.7 Billion Minutes of Use and $480 Million in Revenues by Year end

1999Business Use Will Accelerate in 2001September 1, 1999 - The worldwide Internet protocol (IP) telephony will explode from 310 million minutes of use in 1998 to 2.7 billion by year end 1999. By 2004, IP telephony minutes will reach 135 billion. Revenues for this service will skyrocket from $480 million in 1999 to $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004.


Growth in VoIP

0.0

5.0

10.0

15.0

20.0

25.0

2000 2001 2002 2003 2004 2005 2006

Re

ve

nu

es

($

bill

ion

)

Early growth from expense

savings

Later growth from revenue

generation from new services

Early deployment by

enterprises and CLECs

Later deployment by

incumbent carriers

(source: Frost & Sullivan)


Class 5 DLCClass 5DLC

VoIP Applications

Some trends can be discerned:

First wave: Bypassing the PSTN

Second wave: Replacing the PSTN

Third wave: Value-added services

PSTN


PSTN bypass – IP Telephony (PC to PC)

Microsoft NetMeeting or similar through dial-up connection to ISP

All VoIP processing in the PC no special infrastructure required

Issues: software compatibility QoS / latency over public Internet

Internet

Class 5 DLCClass 5DLCRAS RAS

modem modem

RADIUSserver

RADIUSserver


PSTN bypass – IP Telephony (PC to phone)

From Multimedia PC to any phone

Required: VoIP gateway on the phone side gateway manager billing system (unless free)

Issues: software compatibility QoS / latency over public Internet

Internet

Class 5 DLCClass 5DLCRAS

RADIUSserver

VoIPGateway

GateKeeper

modem


PSTN bypass – IP Telephony (phone to phone)

From any phone to any phone First VoIP application – 1995 Caused by high international tariffs

Required: VoIP gateway on both sides gateway manager billing system (unless free)

Issues: QoS / latency over public Internet sometimes it takes 24 digits to reach

a subscriber…

Class 5 DLCClass 5DLCVoIP

Gateway

GateKeeper

VoIPGateway

IPnetwork


PSTN bypass – Enterprise gateway

Link PABXes on company sites through data network first Voice over Frame Relay, next Voice over IP

Required: VoIP gateway at each site

sometimes integrated in WAN router

Issues: dial plan configuration not easy!

IPnetwork

PABX PABX

PSTN

VoIPGateway

VoIPGateway


PSTN bypass – IP PABX

Two steps:

A. PABX with integrated IP gateway B. Fully integrated enterprise LAN

Required: IP PABX IP phones (step 2)

Issues: dial plan configuration not easy! how to quarantee QoS on LAN? (step 2)

IPnetwork

IP-PABX IP-PABXIP-phone

PSTN

A B


PSTN replacement – Softswitch

Replace complete Class 4 / Class 5 switch very ambitious undertaking! different introduction strategies

Required Softswitch - contains Call Control & Mgmt software Trunking Gateway – interfaces to “legacy” PSTN Access Gateway – interfaces to DLCs

Issues: immaturity of standards (MGCP vs Megaco debate)

DLCClass 5DLCAccess

GatewayTrunkingGateway

Softswitch

IPnetwork


PSTN replacement – Integrated access network

Integrating Access Gateway into DLC

Required: “Next Gen” DLC, with integrated IP gateway

Issues: immaturity of standards

NexGenDLC

NexGenDLC

Softswitch

IPnetwork


Class 5

PSTN

GateKeeper

VoIPGateway

IntegratedAccess Device

PSTN replacement – Integrated Access Devices

Target: single voice/data access network for example wireless access network (LMDS, WLL)

Required: Integrated Access Device (IAD) gateway to PSTN

Issues: immaturity of standards

IntegratedAccess Device

IPnetwork

Softswitch


Value Added Services

Converged services Internet Call Waiting Click to Call Unified messaging …

Video telephony (3rd time right?)


Application Scenario

Mike Brown uses his Palm XIXv for voice calls and Internet access. He subscribes to “Mobile Web” from Verizon Wireless.

Mike is wallpapering, but has some trouble. He calls Home Depot for help. Mike reaches Home Depot’s IVR and is greeted with the

following: “Hi Mike, welcome back to Home Depot, the only choice you need in home improvement supplies and help. Press 0 at any time to speak with an operator. To help with you home improvement needs, we have created a number of short videos which are available free of charge to Preferred Customers such as yourself. To access the video library, say ‘Video’. Otherwise, please say ‘Sales’ for sales; ‘Delivery’ to schedule a delivery; “’Service’ for Customer Service, or ‘Message’ to leave a message. Have a great day.”

Mike mumbles ‘Video’.


Application Scenario (Cont.)

Announcement “Welcome to Home Depot’s video library. Our database indicates you have a PalmPhone so we will play the streamed video. Please say ‘email’ if you prefer the video as email attachment. [short silence]Please make your selection from the following list of home improvement videos by pressing the corresponding number at any time. 1 for hardwood floors, 2 for interior painting, 3 for wallpapering, 4 for outside lighting”

Mike says ‘3’. Announcement: “Thank You. We hope the video on

wallpapering is helpful. Please press ‘video’ on your screen, and your video should start within a few seconds. Have a great day.”

Mike presses “play video” on the PalmPhone screen. Video starts (advertising opportunity/branding

opportunity, with hotlinks back to Home Depot site and context-sensitive buttons to contact product specialists).

Standards for VoIP


The H.323 Protocol Stack

H.225RAS

channel

H.225RAS

channel

Q.931call

setup

Q.931call

setup

H.245control

H.245control

AudioAnd

VideoControl

RTCP

AudioAnd

VideoControl

RTCP

T.120T.120

AudiocodecG.711G.723G.729

AudiocodecG.711G.723G.729

VideoCodecH.261H.263

VideoCodecH.261H.263

RTPRTP

Transport Layer (TCP or UTP)Transport Layer (TCP or UTP)

IPIP

System control user interface Mic CameraData

applications


Real-Time Transport Protocol

Developed by IETF Provides end-to-end delivery services for data with real-

time characteristics such as interactive audio and video. Supports a wide variety of fixed and variable-speed audio

and video signals Applications typically run RTP on top of UTP Supports multicast distribution

V=2

P X CC M PT Sequence Number

Timestamp

Synchronization Source Identifier

Contributing Source Identifiers (0 to 15 entries)

P – PaddingX – ExtensionCC – Contribution source count

M – Marker (for silence suppression)PT – Payload type


RTP Control Protocol (RTCP)

Sent from receiver back to source Used to monitor the quality of service Feedback allows modifications to be made at source 5 percent of session bandwidth allocated to RTCP

V=2 P PT=SR=200 Length

SSRC of sender

NTP Timestamp (Network time)

NTP Timestamp

RTP Timestamp (Network time in RTP units)

Sender’s packet count

Sender’s octet count

SSRC of first cource

Fraction lost Cumulative number of packets lost

Extended highest sequence number received

Interarrival jitter

Last SR

Delay since last SR


H.225 RAS Control

Gatekeeper Optional network entity Offers bandwidth control services Offers address translation to enable use of aliases

H.225 Operates between a Gatekeeper and the endpoints it

controls Provides functions of discovery, registration, admission,

bandwidth change, disengage

GatekeeperEndpoint

GatewayMultiportControl Unit

H.225


Call Signaling in H.232

Q.931 Establishes and tears down calls between endpoints (Q.931 is the signaling protocol for the ISDN user-network

interface) H.245

Negotiates and establishes media streams between call participants

Takes care of multiplexing multiple media streams for functions such as lip synchronization between audio and video

Q.931

H.245


Session Initiation Protocol (SIP)

User to user protocol Developed by IETF (RFC 2543) Establishes and maintains session level information

Creating and tearing down of sessions, session parameters, and media type

Supports personal mobility Heavily influenced by http protocol A light weight protocol compared to H.323

Fewer messages required on a typical call Allows for faster call setup

Flexible in enabling other information to be included messages Allows user devices to exchange specialized information

to enable new services E.g., indicate when a busy terminal will become free

Example SIP addressing; sip:9729965000@gateway


Internet call processing

Decentralized (independent, self-reliant, user to user):

ITU H.323

IETF Session Initiation Protocol (SIP)

Centralized (intelligence in Softswitch):

IETF MEGACO

ITU H.248


Softswitch Architecture

Softswitch separates function of Gateway from the media gateway

AccessGateway

TrunkGateway

Softswitch

IPNetwork

PSTNNetwork

MGCPOr

Megaco

SIP-TTo other

Softswitches


ATM QoS Parameters

Peak-to-peak cell delay variation

Maximum cell transfer delay

Cell loss ratio

Cell error ratio

Severely errored cell block ratio

Cell misinsertion rate

Negotiated at start of call

Controlled viaNetwork design


Real-Time Multimedia over ATM (RMOA)

Developed by ATM Forum More efficient and scalable than H.323 VoIP over ATM New type of gateway: the H.323 to H.323 gateway

Placed at the edges of an ATM network Intercepts H.323 signaling messages to set up virtual circuits in

the ATM network Efficient: IP and UDP headers not carried on the ATM network Takes advantage of QoS capabilities of the ATM network

ATMnetwork

PSTNSwitch

PSTNSwitch

IP Network

VoIPGateway

VoIPGateway

H.323Gateway

H.323Gateway


Resource Reservation Protocol (RSVP)

Specified in RFC 2215 Reserves resources along path from received back to sender Implements various services

Guaranteed service – no packet loss and minimal delay Controlled load service – service like a lightly loaded network Number of parameters associated with each service

Comprehensive, close to circuit emulation, but at significant cost

Application RSVPProcess

PolicyControl

AdmissionControl

PacketScheduler

PacketClassifier

Control

RoutingProcess

RSVPProcess

PolicyControl

AdmissionControl

PacketScheduler

PacketClassifier

Control

Host Router


Adding QoS to IP Networks: Diffserv

Relatively simple means for prioritization traffic (RFC 2475) Makes use of the IPv4 Type of Service (TOS) field Defines two types of packet forwarding:

Expedited Forwarding – assigns a minimum departure rates greater than the per-agreed maximum arrival rate

Assured Forwarding – packets are forwarded with high probability if arrive no faster that per-agreed maximum

Keeps core relatively simple Pushes processing to the edge

Meter

Classifier MarkerShaper /Dropper

VoIP access via DSL and Cable

Modems

All rights reserved © 2001, Alcatel

Cable Telephony

Where to put the RJ-11 telephone jack? On cable modem On set-top box On separate telephony modem On interface on side of house

Local powering or network powering options

Headend

Headend

VideoContent

FiberNode

InternetService

GatewayPSTN


What is DOCSIS?(Data Over Cable System Interface

Specifications)

Started 12/95 by MCNS consortium (Multimedia Cable

Network System)

Goal: Interoperable cable modems and Cable Modem

Termination Systems (CMTS)

Steamed rolled slower (ATM-based) IEEE 802.14

standardization process

Gaining momentum in Europe as EuroDOCSIS

(8 MHz channelization)

Testing and certification by Cable Labs


Who are the DOCSIS Cable Modem Suppliers?

3Com Ambit Arris Interactive Askey Computer Corp. Best Data Castlenet Cisco Systems Com21 Dassault DeltaKable DX Antenna ELSA E-Tech Future Networks GadLine Toshiba

Turbocom General

Instrument GVC Joohong Motorola Net N Sys Nortel Philips Powercom Samsung Sohoware Sony Tarayon Thomson Zoom ZyXel


Cable ModemsMore suppliers, lower prices

0

1000

2000

3000

4000

5000

6000

1990 1992 1994 1996 1998 2000

Cab

le M

odem

Pri

ce

5 Suppliers

32 Suppliers


What is Packet Cable?

Network specification by Cable Labs

Based on DOCSIS 1.1

Support for voice services

Complete network architecture specification

First using GR 303 gateways (to legacy telephone networks)

Evolving to Softswitch (next generation IP networks)

Trials underway


CableLabs Packet Cable Architecture

Managed IPNetwork

HFC AccessNetwork(DOCSIS)

MTACable

Modem CMTS

CallManagement

Server(CMS)

AnnouncementController

(ANC)

AnnouncementPlayer(ANP)

AnnouncementServer

PSTN

OSSBack OfficeServers andApplications

HFC AccessNetwork(DOCSIS)

MTACable

Modem CMTS

Media GatewayController

(MGC)

Media Gateway(MG)

Signaling Gateway(ANP)


Status of VoIP over Cable

Cable executive panel at Cable 2001 (June 11, 2001):

Most cable operators will not roll out significant VoIP until at least the end of 2002

Concerned about the operational issues that come with providing a voice service

General agreement that VoIP economics work in their favor if combined with existing cable services

Some view VoIP as primary line replacement; others (e.g., Time Warner) as second line service


North America Cable Telephony

02,0004,0006,0008,000

10,00012,00014,00016,000

Mill

ion

Ho

useh

old

s

CircuitSwitched

VoIP

Total

Cable projected to capture 15 % telephony market share by 2005

Shift from proprietary TDM solutions towards VoIP DOCSIS

Residential VoIP happening first in the Cable Access Market

North America Cable TelephonyMarket Size


Class 5Switch

ATMSwitch

VoiceGate Way

Integrated Access Device

DSLAM LAN1 VC for Voice1 VC for Data

ADSLDS3 / OC-3

GR303

HOME/BUSINESS

CO / CEVCO

4-16

Voice over DSL

Integrated Access Device (IAD) provides LAN interface and provides multiple telephone interfaces

IAD could be integrated into NID at side of the home Voice Gateway provides same switch interface as though lines

were concentrated on a Digital Loop Carrier system GR303 allows for number portability, billing and additional

voice features

PSTN

DataNetwork


IP

ATM

DMT

AnalogSpectrum

• Voice over IP

• Voice over ATM

• Voice over TDM

• Voice in separate spectrum(e.g., ADSL over DAML)

Voice over ADSL Alternatives

Choice of Voice over ATM in initial implementations– AAL-2– Low-delay, clear 64 kb/s PCM and 32 kb/s ADPCM– QoS support within ATM– Full PSTN quality– V.90 modem support

Support for Voice over IP gaining momentum Maturing of QoS capabilities Potential of IAD becoming a SIP terminal

Layer 3

Layer 1

Layer 2

Alt

ern

ativ

es f

or V

oDS

L

Quality issues for the transport of

voice over packet-based networks


The three essential stages of packet-based voice transport

one-way Mouth-to-Ear (M2E) delay

overall distortion (codec & packet loss)

Encoding and packetization stage Packet transport stage

Echo control performed close to destination

(Concatenation of)(Concatenation of) Packet-based Packet-based

Network(s)Network(s)

Dejittering and decoding stage


PacketizationPacketizationdelaydelay

Total Total queuingqueuing

delaydelay

DejitteringDejitteringdelaydelay

TotalTotalminimalminimal

delaydelay

M2E delay

Components of the M2E delay

Packetization delay is chosen by the source terminal or the source terminal or ingress GWingress GW

Minimal delay and queuing delay depend on QoS QoS provided by traversed network(s)provided by traversed network(s) Each network component has its specific contribution

Dejittering delay is chosen by the destination terminal the destination terminal or egress GWor egress GW


Trade-off M2E delay vs. packet loss in destination or egress GW

Packet lossPacket loss

DejitteringDejittering delaydelay

Delay of first packet

Minimal delay

M2E delay

Pdf(delay)Pdf(delay)

Static dejittering mechanism = delay first packet over dejittering delay and then read dejittering buffer periodically

Choose dejittering delay on save side: for the case when first packet is the fastest possible Adaptive dejittering


Contributions to distortion

Voice compression encoding/decoding

voice activity detection

transcoding

Packet loss in network

in dejittering buffer

Remarks packet loss concealment techniques

trade-off packet loss vs. delay when choosing the dejittering delay


Trade-offs

Packet Packet size size

Network (transport) parametersNetwork (transport) parameters minimal delayminimal delay delay jitterdelay jitter packet losspacket loss

Codec Codec

Efficiency of transport Efficiency of transport Voice quality Voice quality

Dejittering Dejittering delay delay

Echo Echo control control

Header Header compression compression


The E-modelITU-T Rec. G.107

RR = = RR00 - - IIss - - IIdd - - IIee + + AA

Rating Factor in [0,100]Rating Factor in [0,100]

Objective network parameters

Basic signal-to-Basic signal-to-noise rationoise ratio

Distortion Distortion impairmentimpairment

AdvantageAdvantagefactor factor

Mean Opinion Score (MOS)

Predictions of user reactions:

E-model

Subjective quality measures

Impairments Impairments which occur which occur

simultaneously simultaneously with voice with voice

signalsignal

ImpairmentImpairments caused by s caused by

delaydelay


Subjective quality of a voice communication

Defined quality categories in ITU-T Rec. G.109

R-value range 100 - 90 90 - 80 80 - 70 70 - 60 60 - 0

Speech transmissionquality category

best high medium low (very) poor

PSTN quality

GSM quality

(very) poor(very) poor

bestbest

highhigh

mediummedium

lowlow

Traditional PSTN qualityTraditional PSTN quality

1

1.5

2

2.5

3

3.5

4

4.5

5

0 10 20 30 40 50 60 70 80 90 100

Rating R

MO

S


The E-model in the context of VoP

In the transport of Voice over Packet-based (VoP) Networks M2E delay distortion

is likely to be higher, acoustic echo may be more important and 4-to-2-wire

hybrid echo might be different

than in circuit-switched voice transport With the E-model we determine (the influence of Id and

Ie)

RR = = ff ( ( M2E delay, level of echo M2E delay, level of echo ;; codec(s), packet losscodec(s), packet loss))

We do not consider influence of Ro , Is and A as their influence is not fundamentally different in packet-based and circuit switched transport


Influence of distortion

RR = = ff ( ( M2E delay, level of echo M2E delay, level of echo ;; codec(s), packet losscodec(s), packet loss)) Intrinsic rating RRintint = rating for a M2E delay of 0 ms In the context of VoIP, there are two impairments that

mainly hamper the quality of a voice call The impairment Ie associated with the use of “special

equipment” or distortion Coding Packet loss Transcoding

The impairment Id associated with the M2E delay


Speech Coding Techniques

Waveform coding – Tries to preserve the time-domain picture of the signal Sampling – 2 X highest frequency preserved Quantizing – the accuracy of each sample

Linear – simple digital / analog conversion Logarithmic – more accuracy for weak signals Adaptive – match measurement to size of signal

Sounds great at high bit rates but degrades quickly at lower bit rates

Vocoding – Tries to represent the characteristics of the human voice Prametric Vocoders

Dozen coefficients to define vocal tract Indication of voiced or unvoiced Excitation energy Pitch

Synthetic sounding at all bit rates but works OK at low bit rates Vector Quanitization – Matches information signal with entries

in a code book. Uses lots of processing power but provides the best quality at lower

bit rates


Major Parameters of Standard Codecs

Origin Standard TypeCodecBit rate

VoiceFrame (ms)

Look ahead (ms)

Algor.delay (ms)

leIntrinsicquality

ITU-T

G.711 PCM 64

0.125 0 0.125

0 94.3

G.726G.727

ADPCM

16 50 44.3

24 25 69.3

32 7 87.3

40 0.125 0 0.125 2 92.3

G.728 LD-CELP12.8

0.625 0 0.62520 74.3

16 7 87.3

G.729(a) CS-ACELP 8 10 5 15 10 84.3

G.723.1ACELP 5.3

30 7.5 37.519 75.3

MP-MLQ 6.3 15 79.3

ETSI

GSM-FR RPE-LTP 13 20 0 20 20 74.3

GSM-SR VSEPL 5.6 20 0 20 23 71.3

GSM-ESR ACEPL 12.2 20 0 20 5 89.3


Intrinsic quality of codecs as function of their bit rate

0

10

20

30

40

50

60

70

80

90

100

0 8 16 24 32 40 48 56 64

codec bit rate (kb/s)

Intr

insi

c ra

ting

Rin

t

G.711

G.726

G.728

G.729

G.723.1

GSM-EFR

GSM-FR

GSM-HR


Influence of packet loss on distortion

0

10

20

30

40

50

60

70

80

90

100

0 2 4 6 8 10 12 14 16packet loss ratio (%)

Intrin

sic

ratin

g R

int

G.729(A) + [email protected] kb/s + VADGSM-EFRG.711 with PLCG.711 without PLC


CODECG.711

(64kb/s)G.726

(40kb/s)G.726

(32kb/s)G.726

(24kb/s)G.726

(16kb/s)G.728

(16kb/s)GSM-FR(13kb/s)

G.728(12.8kb/s)

GSM-EFR(12.2kb/s)

G.729(8kb/s)

G.723.1(6.3kb/s)

GSM-HR(5.6kb/s)

G.723.1(5.3kb/s)

G.711(64kb/s) 94.3 92.3 87.3 69.3 44.3 87.3 74.3 74.3 89.3 84.3 79.3 71.3 75.3G.726

(40kb/s) 92.3 90.3 85.3 67.3 42.3 85.3 72.3 72.3 87.3 82.3 77.3 69.3 71.3G.726

(32kb/s) 87.3 85.3 80.3 62.3 37.3 80.3 67.3 67.3 82.3 77.3 72.3 64.3 68.3

G.726(24kb/s) 69.3 67.3 62.3 44.3 19.3 62.3 49.3 49.3 64.3 59.3 54.3 46.3 50.3

G.726(16kb/s) 44.3 42.3 37.3 19.3 0 37.3 24.3 24.3 39.3 34.3 29.3 21.3 25.3G.728

(16kb/s) 87.3 85.3 80.3 62.3 37.3 80.3 67.3 67.3 82.3 77.3 72.3 64.3 68.3GSM-FR(13kb/s) 74.3 72.3 67.3 49.3 24.3 67.3 54.3 54.3 69.3 64.3 59.3 51.3 55.3G.728

(12.8kb/s) 74.3 72.3 67.3 49.3 24.3 67.3 54.3 54.3 69.3 64.3 59.3 51.3 55.3GSM-EFR(12.2kb/s) 89.3 87.3 82.3 64.3 39.3 82.3 69.3 69.3 84.3 79.3 74.3 66.3 70.3

G.729(8kb/s) 84.3 82.3 77.3 59.3 34.3 77.3 64.3 64.3 79.3 74.3 69.3 61.3 65.3G.723.1(6.3kb/s) 79.3 77.3 72.3 54.3 29.3 72.3 59.3 59.3 74.3 69.3 64.3 56.3 60.3GSM-HR(5.6kb/s) 71.3 69.3 64.3 46.3 21.3 64.3 51.3 51.3 66.3 61.3 56.3 48.3 52.3G.723.1(5.3kb/s) 75.3 73.3 68.3 50.3 25.3 68.3 55.3 55.3 70.3 65.3 60.3 52.3 56.3

Transcoding matrix

Transcoding is the translation of one codec format into another (via the linearly quantized 8 kHz sampled voice format)


Influence of M2E delay

RR = = ff ( ( M2E delay, level of echo M2E delay, level of echo ;; codec(s), packet losscodec(s), packet loss)) In the context of VoIP, there are two impairments that

mainly hamper the quality of a voice call The impairment Ie associated with the use of “special

equipment” or distortion The impairment Id associated with the M2E delay

Loss of interactivity Talker echo Listener echo


Rating as function of M2E delay for traditional G.711 codec (no

packet loss)

0

10

20

30

40

50

60

70

80

90

100

0 100 200 300 400mouth-to-ear delay (ms)

ratin

g R

EL=11 dBEL=21 dBEL=31 dBEL=41 dBEL=51 dBEL=infinity

packet loss = 0.0%

Interactivity Interactivity boundbound

(150 ms)(150 ms)

Echo Echo controlcontrolboundbound

(25 ms)(25 ms)

Maximum Maximum boundbound

(400 ms)(400 ms)


Influence of echo control

Echo control can increase the echo loss value EL Echo Controllers compliant with ITU-T Rec. G.168

increase EL by at least 30 dB Even perfect echo control is possible (at the expense of

some distortion) Electrical echo (introduced in 4-to-2-wire hybrid) has

typically EL = 21 dB Acoustic echo is dominant in non-optimized terminals

and is harder to control than electrical echo Echo controller is simpler, if it is close to source of echo Echo control is required in an VoP transport

environment


Rating for G.711 codec (without PLC)Packet loss: 0%, 0.5% and 1%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 0.5%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 1.0%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 0.0%


Rating for G.729 codec with VAD Packet loss: 0%, 1%, 2%, 3% and 4%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 0.0%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 1.0%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 2.0%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 3.0%

0

10

20

30

40

50

60

70

80

90

100


ratin

g R


packet loss = 4.0%


M2E delay and packet loss bounds

If there is no packet loss, the M2E delay can exceed 150 ms If the M2E delay is below 150 ms some packet loss can be tolerated

Origin Standardcodec bit rate (kb/s)

PL bound(%)

G.711 without PLC 64 1G.711 with PLC 64 10G.729(A) + VAD 8 3.4

[email protected] kb/s + VAD 6.3 2.1ETSI GSM-EFR 12.2 2.7

ITU-T

Origin Standardcodec bit rate (kb/s)

M2E delaybound (ms)

G.711 64 40016 NA24 NA32 32440 379

12.8 21216 324

G.729(A) 8 2965.3 2216.3 253

GSM-FR 13 212GSM-HR 5.6 180

GSM-EFR 12.2 345ETSI

ITU-T

G.726G.727

G.728

G.723.1

Bounds under perfect echo control


Quality of a telephone conversation (using the E-model of ITU-T Rec.

G.107 and G.109)

0

5

10

15

20

25

30

35

40

45

0 50 100 150 200 250 300 350 400

M2E delay (ms)

dis

tort

ion

(Very) Bad(Very) Bad

PoorPoor

MediumMedium

HighHigh

BestBest

Perfect Perfect echo controlecho control


ConclusionsQuality of a telephone call

(Perfect) echo control is strongly recommended Under perfect echo control the intrinsic quality

remains constant if M2E delay < 150 ms Choose codec to have an intrinsic quality that is

good enough e.g. G.711, G.729, ...

Avoid transcoding from one low bit rate codec into another

Keep M2E delay and packet loss under control bounds are codec-dependent

There is a trade-off between M2E delay and distortion


ConclusionsSetting the parameters

The quality with which the voice flows are transported influence the overall quality, but …

… the choice of the codec, packet size and dejittering delay is also primordial In the choice of the codec there is a trade-off between

efficiency and quality In the choice of the packet size there is a trade-off

between efficiency and quality Tuning the dejittering mechanism correctly is important

to attain high quality

Addressability inVoIP Networks


Addressibility in VoIP

Question: How do you dial a VoIP user if all you have is their telephone number?

alcatel.com

ge.com

fcc.gov

ibm.com

Users resistant to change services if they have to change phone numbers


What is ENUM?

Telephone number mapping (RFC 2916, RFC 2915)

Allows a phone number to enable a caller to reach all

kinds of devices (fax, IP telephone, email, etc.) by

knowing a single contact number

Originally proposed by Patrik Falstrom of Cisco

Uses DNS structure to map an E.164 phone number

into a series of Internet addresses:

SIP, H323, SMTP, VPIM, IPP, etc.

Enables Local Number Portability, 800 services


DNS-B(0.5.8.9.1.9.1.e164.arpa)

DNS-A(9.1.9.1.e164.arpa)

How does ENUM work?

proxy.comINVITE

INVITE

Answer = sip:n

iel@pro

xy.com

“(919) 850-5500"

Qu

ery

0.0.

5.5.

0.5.

8.9.

1.9.

1.e1

64.a

rpa

Au

tho

rity

= D

NS

-B

Query 0.0.5.5.0.5.8.9.1.9.1.e164.arp

a


What is TRIP?

Telephony Routing over IP

Identifies gateway for PSTN destinations

Being developed in IETF; currently only in IETF draft

Distributes routes to E.164 prefixes (gateways)

Enables policy based routing

Based on BGP

Real-time recalculation of routes whenever paths

becomes unavailable


How does TRIP work?

GW-AGW-B

LS-Y

LS-Z

16132-16135,GW-A16136-16139,GW-B

1613,NHSx 1613,NHSw

1613,NHSx

NHSxLS-X

LS-W


ENUM Implementation Issues

Should ENUM databases be public or commercial? Public ENUM (NeuStar) Commercial ENUM: (NetNumber) (VeriSign keeping its options open on both)

Which agency within the government will be responsible for ENUM implementation within the US

Selection of a Tier 1 ENUM operator for the United States portion of the North American Numbering Plan

State Dept Study Group A ENUM Ad Hoc Formed to develop a recommendation to the USG on its

participation in global and national ENUM implementation Ad Hoc participants intend to form an industry forum to

address the ENUM implementation in the United States

RegulatoryConsiderations


Context

The third ITU-T World Telecommunication Policy Forum (Geneva, March 7-9 2001) discussed issues related to “Internet Protocol (IP) Telephony”.

The WTPF discussed the impact of IP telephony on regulation and policies of ITU member states and ways for offering technical assistance to developing countries.

A report of the secretary-general and draft opinions for the forum are finalized and available on the ITU website (http://www.itu.int/wtpf).


What is at stake ?

Beyond the technological hype surrounding IP telephony, the real issue is the structure of the 21st century world-wide telecom network and the nature - and mere existence ! - of the settlement system governing the interconnection between operators.

Many developing countries are fearing that widespread deployment of unregulated IP telephony traffic will dramatically lower the revenue stream drawn from the settlement system and, by way of consequence, the eventual insolvency of their local PTO(s).

The secretary-general’s report on IP telephony is quite objective and factual but the WTPF draft opinion recommendations reflect conflicting interests.


The “Netheads” view

Driven by CISCO, VON coalition, global operators (Worldcom, AT&T) .

Objective: convince reluctant (mainly developing) countries to allow free competition of IP telephony with their local PTO.

Mantra: IP is “the new” technology for telecommunications; IP is much more efficient (cost) than legacy TDM; IP networks open the way for new services and help reduce the

“digital divide”; IP telephony should not fall under the telecom regulation regime

(or this regime should evolve) because it uses a new technology.


The EU view

Advocates the principle of technological neutrality. EU has a strict definition of voice telephony in terms of the

following four principles: it is offered commercially as such; it is provided to the public; it is provided to and from PSTN termination points; it involves speech transport and switching of voice in real-time

with the same level of reliability and quality as existing PSTN networks.


Other Regulatory Implications

Regulatory parity (regulating services vs. technologies) Should a telephone call be

regulated differently if it is TDM, VoIP, FTTH, DOCSIS?

Protocol conversion Is gateway functionality protocol

conversion in a CI-II / CI-III context? Unbundling

What are the UNE’s of a VoIP network?

How should competitive access provided in a VoDSL and FTTH environment?

CPE Deregulation With gateway functionality moving

to the end user


Further Reading …

David J Write, Voice over Packet Networks, J. Wiley, 2001.

Jonathan Davidson and James Peters, Voice over IP Fundamentals, Cisco Press, 2000.

Daniel Minoli and Emma Minoli, Delivering Voice of IP Networks, Wiley Computer Publishing, 1998.

David Collins, Carrier Grade Voice over IP, McGraw-Hill, 2001.


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