Voice over IP
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
2
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, xDSL, Ethernet, WiFi
3
Circuit switching served voice well
for 100 years!
Signal System 7
Data link
Signal Transfer
Point
Trunk
Group
Loop
User - A
Class 5
Switching System
User - B
Central
Office - A
Transit
Office
Connection Through
Switching Fabric
Central
Office - B
Class 4
Switching System
 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
4
Packet Switching
Well-matched for data transmission
Packet
Payload Header
Input Buffer
Hdr. Trans
Routing
Fabric
Output Buffer
Hdr. Trans





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, like speaking to man on moon
5
VoIP Network Architecture
Gateway
Gatekeeper
PSTN
network
IP
network
Media
Gateway
Media
Gateway
 Media gateways provide voice packetization
 Gatekeepers provides call control logic and permissions
 Gateway provides interworking with ISDN, SS7 and signaling of
PSTN (POTS)
6
Advantages of VoIP
 Lack of access charges, flat rate or volume based IP
 Cheap setup costs competition with POTS
 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, conferencing calls
 Intelligent terminals (e.g., PC)
 Better (graphical) user interface
 Clean slate design:
 Separation of feature intelligence
from switching fabric supplier
 Self-provisioning networks
7
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 overprovisioning bandwidth to ensure quality
 TDM Switch costs in traditional PSTN replaced by cost of
Router plus cost of Gateway and new billing systems
No network cost savings, and very likely a cost penalty, in the initial
years, in going from PSTN voice to VoIP for public networks
8
PSTN versus VoIP
Today’s PSTN
VoIP
TDM circuit switching
Packet switching
QoS guarantees
Yes
No
Network resource
reserved at call setup
Yes
No
Class 4, Class 5
switching systems
Mostly integrated in
switching system
Gateways, gateway
controllers, routers
In separate gateway
controllers
64 kb/s
Variable 5.3 – 32 kb/s
DTMF, SS7
SIP, H.323, MGCP
Underlying
Technology
Network elements
Call processing
intelligence
Bandwidth per call
Signaling
Transport
How reliability
achieved
9
ATM, FR, native IP in
TDM in access, edge,
access; ATM native IP
core
in core, WiFi
Redundancy within
Redundant routes
each network
through network
element
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
10
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
11
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.
IP TELEPHONY OVER LAN MARKET FORECASTED TO GROW
138% AVERAGE ANNUALLY OVER NEXT 5 YEARS
September 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
1999
Business Use Will Accelerate in 2001
September 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 $19 billion by 2004.
IP Telephony Services: Market Review and Forecast, 1998-2004.
12
Growth in VoIP
25.0

Early growth from expense savings

Later growth from revenue
20.0
Revenues ($ billion)
generation from new services

15.0
and CLECs

10.0
Later deployment by incumbent
carriers
5.0
0.0
2000
13
Early deployment by enterprises
2001
2002
2003
(source: Frost & Sullivan)
2004
2005
2006
VoIP Applications
 Some trends can be discerned:
 First wave:
Bypassing the PSTN
 Second wave:
Replacing the PSTN
 Third wave:
Value-added services
PSTN
DLC
14
Class 5
Class 5
DLC
PSTN bypass –
IP Telephony (PC to PC)
 Microsoft NetMeeting or similar
 through dial-up/adsl/cable connection to ISP
 All VoIP processing in the PC
 no special infrastructure required
 Issues:
 software compatibility
 QoS / latency over public Internet
 Strange dialing
RADIUS
server
RADIUS
server
Internet
modem
DLC
15
Class 5
RAS
RAS
modem
Class 5
DLC
PSTN bypass –
IP Telephony (PC to PHONE)
 From Multimedia PC to any PHONE
 First applications 1993
 Required:
 VoIP gateway on the phone side
 gateway manager
 billing system (unless free)
 Issues:
 software compatibility
 QoS / latency over public Internet
RADIUS
server
Gate
Keeper
Internet
modem
DLC
16
Class 5
RAS
VoIP
Gateway
Class 5
DLC
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…
Gate
Keeper
IP
network
DLC
17
Class 5
VoIP
Gateway
VoIP
Gateway
Class 5
DLC
PSTN bypass –
IP Telephony (phone to pc)
 From any PHONE to any PC
 First VoIP application – 2004
 Try to replace PSTN
 Required:
 VoIP gateway on PSTN side
 MSN numbers
 gateway manager
 billing system (unless free)
 Issues:
 QoS / latency over public Internet
Gate
Keeper
RADIUS
server
IP
network
DLC
18
Class 5
VoIP
Gateway
RAS
modem
Class 5
DLC
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)
Soft
switch
IP
network
DLC
19
Class 5
Trunking
Gateway
Access
Gateway
DLC
PSTN replacement –
Integrated access network
 Integrating Access Gateway into DLC
 Required:
 “Next Gen” DLC, with integrated IP gateway
 Issues:
 immaturity of standards
Soft
switch
IP
network
NexGen
DLC
20
NexGen
DLC
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)
A
B
IP
network
IP-PABX
IP-PABX
PSTN
21
IP-phone
PSTN replacement –
Integrated Access Devices
 Target: single voice/data access network
 for example wireless access network
 Home networks
 companies
 Required:
 Integrated Access Device (IAD)
 gateway to PSTN somewhere
 Issues:
 immaturity of standards
Integrated
Access Device
Gate Soft
Keeper
switch
VoIP
Gateway
22
IP
PSTN
network
Class 5
Value Added Services
 Converged services
 Internet Call Waiting
 Click to Call
 Unified messaging
 …
 Video telephony
(3rd time right?)
23
Standards for VoIP
The H.323 Protocol Stack
System control user interface
H.225
RAS
channel
Q.931
call
setup
H.245
control
Mic
Audio
And
Video
Control
Camera
Data
applications
Video
Codec
H.261
H.263
T.120
Audio
codec
G.711
G.723
G.729
RTCP
RTP
Transport Layer (TCP or UTP)
IP
25
H.225 RAS Control
Gatekeeper
Endpoint
H.225
Multiport
Control Unit
Gateway
 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
26
Call Signaling in H.232
Q.931
H.245
 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
27
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
28
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
29
Softswitch Architecture
To other
Softswitches
SIP-T
MGCP
Or
Megaco
Softswitch
IP
Network
PSTN
Network
Trunk
Gateway
Access
Gateway
 Softswitch separates function of Gateway from the media
gateway
30
ATM QoS Parameters
 Peak-to-peak cell delay variation
 Maximum cell transfer delay
Negotiated at start of call
 Cell loss ratio
 Cell error ratio
 Severely errored cell block ratio
 Cell misinsertion rate
31
Controlled via
Network design
Real-Time Multimedia over ATM (RMOA)
PSTN
Switch
H.323
Gateway
VoIP
Gateway
IP Network
ATM
network
H.323
Gateway
PSTN
Switch
VoIP
Gateway
 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
32
Resource Reservation Protocol (RSVP)
Host
Router
RSVP
Process
Application
Control
Policy
Control
RSVP
Process
Routing
Process
Control
Admission
Control
Packet
Classifier
Packet
Scheduler
Policy
Control
Admission
Control
Packet
Classifier
Packet
Scheduler
 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
33
Adding QoS to IP Networks: Diffserv
Meter
Classifier
Marker
Shaper /
Dropper
 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
34
VoIP access via DSL
and Cable Modems
Cable Telephony
Video
Content
Internet
Service
Head
end
Fiber
Node
PSTN
Gateway

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
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
North America Cable Telephony
Market Size
1 6 ,0 0 0
1 4 ,0 0 0
1 2 ,0 0 0
1 0 ,0 0 0
8 ,0 0 0
6 ,0 0 0
4 ,0 0 0
2 ,0 0 0
0
C irc uit
S w itc he d
V o IP
05
20
04
20
03
20
02
20
01
20
20
19
00
T o ta l
99
Million H ouseholds
N o rth A m e ri c a C a b l e T e l e p h o n y
 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
Voice over DSL
CO
PSTN
CO / CEV
GR303 Voice
Gate
Way
Data
Network
HOME/BUSINESS
ADSL
DS3 / OC-3
Class 5
Switch
4-16
ATM
Switch
1 VC for Voice
1 VC for Data
DSLAM
LAN
Integrated Access Device
 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
40
Alternatives for VoDSL
Voice over ADSL Alternatives
• Voice over IP
IP
Layer 3
• Voice over ATM
ATM
Layer 2
• Voice over TDM
DMT
Layer 1
• Voice in separate spectrum
(e.g., ADSL over DAML)
Analog
Spectrum
 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
41
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)
(Concatenation of)
Packet-based
Network(s)
Encoding and
packetization stage
Packet transport stage
Dejittering and
decoding stage
Echo control performed
close to destination
43
Components of the M2E delay
Packetization
delay
Total
minimal
delay
Total
queuing
delay
Dejittering
delay
M2E delay
 Packetization delay is chosen by the source terminal or ingress
GW
 Minimal delay and queuing delay depend on QoS provided by
traversed network(s)
 Each network component has its specific contribution
 Dejittering delay is chosen by the destination terminal or egress
GW
44
Trade-off M2E delay vs. packet loss
in destination or egress GW
Pdf(delay)
Dejittering delay
Packet loss
Minimal Delay of
delay first packet
M2E 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
45
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
46
Trade-offs
Network (transport) parameters
 minimal delay
 delay jitter
 packet loss
Echo
control
Dejittering
delay
Voice quality
47
Packet
size
Codec
Header
compression
Efficiency of transport
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
48
Major Parameters of Standard Codecs
Origin
Standard
Type
Codec
Bit rate
G.711
PCM
64
Voice
Frame (ms)
Look
ahead
(ms)
Algor.
delay (ms)
le
Intrinsic
quality
0
94.3
50
44.3
24
25
69.3
32
7
87.3
2
92.3
20
74.3
7
87.3
10
84.3
19
75.3
15
79.3
16
0.125
G.726
G.727
0
0.125
ADPCM
40
0.125
0
0.125
ITU-T
12.8
G.728
LD-CELP
0.625
0
0.625
16
G.729(a)
CS-ACELP
8
ACELP
5.3
G.723.1
ETSI
49
10
5
15
30
7.5
37.5
MP-MLQ
6.3
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
Influence of packet loss on distortion
100
90
Intrinsic rating R int
80
70
60
50
40
G.729(A) + VAD
G.723.1@6.3 kb/ s + VAD
GSM-EFR
G.711 with PLC
G.711 without PLC
30
20
10
0
0
2
4
6
8
10
packet loss ratio (%)
50
12
14
16
Transcoding matrix
 Transcoding is the translation of one codec format into another
(via the linearly quantized 8 kHz sampled voice format)
CODEC
G.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)
51
G.711
G.726
G.726
G.726
G.726
G.728 GSM-FR
G.728
GSM-EFR G.729 G.723.1 GSM-HR G.723.1
(64kb/s) (40kb/s) (32kb/s) (24kb/s) (16kb/s) (16kb/s) (13kb/s) (12.8kb/s) (12.2kb/s) (8kb/s) (6.3kb/s) (5.6kb/s) (5.3kb/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
75.3
92.3
90.3
85.3
67.3
42.3
85.3
72.3
72.3
87.3
82.3
77.3
69.3
71.3
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
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
44.3
42.3
37.3
19.3
0
37.3
24.3
24.3
39.3
34.3
29.3
21.3
25.3
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
74.3
72.3
67.3
49.3
24.3
67.3
54.3
54.3
69.3
64.3
59.3
51.3
55.3
74.3
72.3
67.3
49.3
24.3
67.3
54.3
54.3
69.3
64.3
59.3
51.3
55.3
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
84.3
82.3
77.3
59.3
34.3
77.3
64.3
64.3
79.3
74.3
69.3
61.3
65.3
79.3
77.3
72.3
54.3
29.3
72.3
59.3
59.3
74.3
69.3
64.3
56.3
60.3
71.3
69.3
64.3
46.3
21.3
64.3
51.3
51.3
66.3
61.3
56.3
48.3
52.3
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
M2E delay and packet loss bounds
Bounds under perfect echo control

If there is no packet loss, the
M2E delay can exceed 150
ms
Origin
codec bit M2E delay
Standard
rate (kb/s) bound (ms)
G.711
G.726
G.727
ITU-T
G.728
G.729(A)
G.723.1
ETSI
52
GSM-FR
GSM-HR
GSM-EFR
64
16
24
32
40
12.8
16
8
5.3
6.3
13
5.6
12.2
400
NA
NA
324
379
212
324
296
221
253
212
180
345

If the M2E delay is below
150 ms some packet loss can be
tolerated
Origin
ITU-T
ETSI
codec bit PL bound
rate (kb/s)
(%)
G.711 without PLC
64
1
G.711 with PLC
64
10
G.729(A) + VAD
8
3.4
6.3
2.1
G.723.1@6.3 kb/s + VAD
GSM-EFR
12.2
2.7
Standard
Quality of a telephone conversation (using
the E-model of ITU-T Rec. G.107 and G.109)
45
Perfect
echo control
40
(Very) Bad
35
Poor
distortion
30
25
Medium
20
15
High
10
5
Best
0
0
50
100
150
200
M2E delay (ms)
53
250
300
350
400
Conclusions
Quality 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
54
Conclusions
Setting 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
55
Addressability in
VoIP Networks
Addressibility in VoIP
 Question: How do you dial a VoIP user if all you have is their
telephone number?
alcatel.com
ibm.com
ge.com
fcc.gov
 Users resistant to change services if they have to change
phone numbers
57
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
58
How does ENUM work?
DNS-A
(9.1.9.1.e164.arpa)
DNS-B
(0.5.8.9.1.9.1.e164.arpa)
proxy.com
INVITE
“(919) 850-5500"
59
INVITE
Regulatory
Considerations
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).
61
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.
62
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.
63
The EU view
 Advocates the principle of technological neutrality.
 EU has a strict definition of voice telephony in terms of the following
four principles:




64
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
65
Further Reading …
 David J Write, Voice over Packet Networks, J. Wiley.
 Jonathan Davidson and James Peters, Voice over IP
Fundamentals, Cisco Press.
 Daniel Minoli and Emma Minoli, Delivering Voice of IP
Networks, Wiley Computer Publishing.
 David Collins, Carrier Grade Voice over IP, McGraw-Hill.
66