# Presentation: 609434

## Voice over IP

***for Carriers***

## 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 wellfor 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**

**Loop**

**Trunk**
**Group**

**Central**
**Office - A**

**Central**
**Office - B**

**Signal System 7**
**Data link**

**Signal Transfer**
**Point**

**Transit**
**Office**

**Class 5**
**Switching System**

**Connection Through****Switching Fabric**

**Class 4**
**Switching System**

## Packet SwitchingWell-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

## VoIP Network Architecture

- Media gateways provide voice packetization
- Media gateway controller provides call control logic
- Signaling gateway provides interworking with SS7 signaling of PSTN

**Media**
**Gateway**

**Media**
**Gateway**
**Controller**

**Media**
**Gateway**

**Signaling**
**Gateway**

## 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 oflow-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.

**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*.

## Growth in VoIP

- 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)**

## VoIP  Applications

- Class 5

- DLC

- Class 5

- DLC

***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)

***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

- DLC

- Class 5

- DLC

- RAS

- RAS

- modem

- modem

- RADIUS
- server

- RADIUS
- server

## PSTN bypass – IP Telephony (PC to phone)

***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

- DLC

- Class 5

- DLC

- RAS

- RADIUS
- server

- VoIP
- Gateway

- Gate
- Keeper

- modem

## PSTN bypass – IP Telephony (phone to phone)

***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

- DLC

- Class 5

- DLC

- VoIP
- Gateway

- Gate
- Keeper

- VoIP
- Gateway

**IP**
**network**

## PSTN bypass – Enterprise gateway

***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!

**IP**
**network**

- PABX

- PABX

- PSTN

- VoIP
- Gateway

- VoIP
- Gateway

## PSTN bypass – IP PABX

***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)

**IP**
**network**

- IP-PABX

- IP-PABX

- IP-phone

- PSTN

- A

- B

## PSTN replacement – Softswitch

***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)

- DLC

- Class 5

- DLC

- Access
- Gateway

- Trunking
- Gateway

- Soft
- switch

**IP**
**network**

## PSTN replacement – Integrated access network

***PSTN replacement – ******Integrated access network***

- Integrating Access Gateway into DLC
- Required:
- “Next Gen” DLC, with integrated IP gateway
- Issues:
- immaturity of standards

- NexGen
- DLC

- NexGen
- DLC

- Soft
- switch

**IP**
**network**

## PSTN replacement – Integrated Access Devices

- Class 5

- PSTN

- Gate
- Keeper

- VoIP
- Gateway

- Integrated
- Access 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

## 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.225**
**RAS**
**channel**

**Q.931**
**call**
**setup**

**H.245**
**control**

**Audio**
**And**
**Video**
**Control**
**RTCP**

**T.120**

**Audio**
**codec**
**G.711**
**G.723**
**G.729**

**Video**
**Codec**
**H.261**
**H.263**

**RTP**

**Transport Layer (TCP or UTP)**

**IP**

**System control user interface**

**Mic**

**Camera**

**Data**
**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 – Padding**
**X – Extension**
**CC – 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

**Gatekeeper**

**Endpoint**

**Gateway**

**Multiport**
**Control 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

## 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 via**
**Network 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

**ATM**
**network**

**PSTN**
**Switch**

**PSTN**
**Switch**

**IP Network**

**VoIP**
**Gateway**

**VoIP**
**Gateway**

**H.323**
**Gateway**

**H.323**
**Gateway**

## 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

**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

## 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

**Head**
**end**

**Video**
**Content**

**Fiber**
**Node**

**Gateway**

## 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 ModemsMore suppliers, lower prices

**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

## 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Market Size

**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 Telephony******Market Size***

## Voice over DSL

- Class 5
- Switch

- ATM
- Switch

- Voice
- Gate
- Way

**Integrated Access Device**

- DSLAM

- LAN

**1 VC for Voice**

**1 VC for Data**

**ADSL**

**DS3 / OC-3**

- GR303

**HOME/BUSINESS**

**CO / CEV**

**CO**

- 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**

**Data**
**Network**

## Voice over ADSL Alternatives

**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**

***Alternatives for VoDSL***

## 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_[ ](file:///echo_control.ppt)__performed __close to destination_

**(Concatenation of)**
** ****Packet-based **
**Network(s)**

- Dejittering and
- decoding stage

## Components of the M2E delay

- Packetization
- delay

- Total
- queuing
- delay

- Dejittering
- delay

- Total
- minimal
- delay

- M2E delay

***Components of the 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

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

- 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
- size

- Network (transport) parameters
-  minimal delay
-  delay jitter
-  packet loss

- Codec

- Efficiency of transport

- Voice quality

- Dejittering
- delay

- Echo
- control

- Header
- compression

## The E-modelITU-T Rec. G.107 

*R*  =  *R*0* *- *I**s** *- *I**d**  *- *I**e**  *+ *A*
- Rating Factor in [0,100]

**Objective**** ****network**** ****parameters**

- Basic signal-to-noise ratio

- Distortion impairment

- Advantagefactor

**M**ean **O**pinion **S**core (MOS)

- Predictions of user reactions:

- E-model

**Subjective quality measures**

- Impairments which occur simultaneously with voice signal

- Impairments caused by delay

## Subjective quality of a voice communication Defined quality categories in ITU-T Rec. G.109

## 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 *I**d* and *I**e*)
*	**R* = *f* ( M2E delay, level of echo ; codec(s), packet loss)
- We do not consider influence of *R*o , *I**s*  and *A *as their influence is not fundamentally different in packet-based and circuit switched transport

## Influence of distortion

*R* = *f* ( M2E delay, level of echo ; codec(s), packet loss)
- Intrinsic rating *R**int* = 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 *I**e *associated with the use of “special equipment” or distortion
- Coding
- Packet loss
- Transcoding
- The impairment *I**d** *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 | Type | Codec
Bit rate | Voice
Frame (ms) | Look ahead (ms) | Algor.
delay (ms) | le | Intrinsic
quality |
| --- | --- | --- | --- | --- | --- | --- | --- | --- |
| ITU-T | G.711 | PCM | 64 | 0.125 | 0 | 0.125 | 0 | 94.3 |
|  | G.726
G.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-CELP | 12.8 | 0.625 | 0 | 0.625 | 20 | 74.3 |
|  |  |  | 16 |  |  |  | 7 | 87.3 |
|  | G.729(a) | CS-ACELP | 8 | 10 | 5 | 15 | 10 | 84.3 |
|  | G.723.1 | ACELP | 5.3 | 30 | 7.5 | 37.5 | 19 | 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

***Intrinsic quality of codecs ******as function of their bit rate***

## Influence of packet loss on distortion

***Influence of packet loss on distortion***

## Transcoding matrix

***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

*R* = *f* ( M2E delay, level of echo ; codec(s), packet loss)
- In the context of VoIP, there are two impairments that mainly hamper the quality of a voice call
- The impairment *I**e *associated with the use of “special equipment” or distortion
- The impairment *I**d** *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)

## 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%

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

## M2E delay and packet loss bounds

***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

- Bounds under perfect echo control

## Quality of a telephone conversation (using the E-model of ITU-T Rec. G.107 and G.109) 

- Perfect echo control

## 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

## 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

## Addressability inVoIP Networks

##  Addressibility in VoIP

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

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

## What is ENUM?

- T_e_lephone _num_ber 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

## How does ENUM work?

- (0.5.8.9.1.9.1.e164.arpa)

- DNS-A
- (9.1.9.1.e164.arpa)

***How does ENUM work?***

- proxy.com

- INVITE

- INVITE

**Answer = sip:niel@proxy.com**

- “(919) 850-5500"

**Query 0.0.5.5.0.5.8.9.1.9.1.e164.arpa**

**Authority = DNS-B**

**Query 0.0.5.5.0.5.8.9.1.9.1.e164.arpa**

## What is TRIP?

- _T_elephony _R_outing 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?

***How does TRIP work?***

**GW-A**

**GW-B**

**LS-Y**

**LS-Z**

**NHSx**

**LS-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

## 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).

## 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.

## Slide 88