66
Voice over IP
Voice over IP
2
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
3
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
4
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
5
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, like speaking to man
on moon
HeaderPacket
PayloadInput Buffer
Output BufferHdr. Trans
Hdr. Trans
RoutingFabric
6
VoIP Network Architecture
Media gateways provide voice packetization Gatekeepers provides call control logic and permissions Gateway provides interworking with ISDN, SS7 and
signaling of PSTN (POTS)
IPnetwork
MediaGateway
Gatekeeper
MediaGateway
PSTNnetwork
Gateway
7
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 intelligencefrom switching fabric supplier
Self-provisioning networks
8
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 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
9
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 5.3 – 32 kb/s
Signaling DTMF, SS7 SIP, H.323, MGCP
Transport TDM in access, edge, core
ATM, FR, native IP in access; ATM native
IP in core, WiFi
How reliability achieved
Redundancy within each network
element
Redundant routes through network
10
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
11
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
12
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 $19 billion by 2004. IP Telephony Services: Market Review and Forecast, 1998-2004.
13
Growth in VoIP
0.0
5.0
10.0
15.0
20.0
25.0
2000 2001 2002 2003 2004 2005 2006
Re
ve
nu
es
($
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)
14
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
15
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
Internet
Class 5 DLCClass 5DLCRAS RAS
modem modem
RADIUSserver
RADIUSserver
16
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
Internet
Class 5 DLCClass 5DLCRAS
RADIUSserver
VoIPGateway
GateKeeper
modem
17
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
18
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
Class 5DLC
GateKeeper
VoIPGateway
IPnetwork
Class 5 DLCRAS
modem
RADIUSserver
19
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
20
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
21
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
22
Class 5
PSTN
GateKeeper
VoIPGateway
IntegratedAccess Device
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
IntegratedAccess Device
IPnetwork
Softswitch
23
Value Added Services
Converged services Internet Call Waiting Click to Call Unified messaging …
Video telephony (3rd time right?)
Standards for VoIP
25
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
26
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
27
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
28
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
29
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
30
Softswitch Architecture
Softswitch separates function of Gateway from the media gateway
AccessGateway
TrunkGateway
Softswitch
IPNetwork
PSTNNetwork
MGCPOr
Megaco
SIP-TTo other
Softswitches
31
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
32
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
33
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
34
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
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
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
40
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
41
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
43
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
44
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
45
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
46
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
47
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
48
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
49
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
50
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
51
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)
52
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
53
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
54
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
55
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
57
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
58
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
59
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
RegulatoryConsiderations
61
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).
62
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.
63
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.
64
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.
65
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
66
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.
