Part 2. Converged networks and services  4. Convergence of fixed networks

4.1. Network characteristics # PSTN/ISDN

# Data networks

4.2. PSTN/Internet convergence for data services # Internet access

4.3. PSTN/Internet convergence for voice services # VoIP and IP Telephony

4.4. QoS issues and Reliability

4.5. Estimation of Call Quality

4.1. Network characteristics• PSTN – more then 100 years history• Basic principals – circuit switching, connection-oriented• Three phases on the session• Reservation of network resources:

# analog voice channel – 4 kHz

# digital voice channel – 64 kbps• Guaranteed level of QoS (delay/loss)• Very high availability – outage is less then 5 min/year

(Bellcore – 3 min/year)

PSTN

PBXLELE

PBX PBX

PSTN

Branch office HQ office

PSTN Call Processing and Protocol Flows

4.1. Network characteristics (Cntd)

• Data networks – 60s, ARPA• Basic principals – packet switching, connectionless-

oriented (IP)• No resource reservation for the transmission• No guarantee for delay and loss – it’s not critical for

data, but critical for other possible apps

Data network

ServerRouter Router

App server

Branch office HQ officeRes. house

Modem/router

Public/private network

Ethernet, ATM, FR, PPP

Physical layer

TCP UDP

HTTP, FTPH.323, SIP, RTP, RSVP, MGCP,

MEGACO/H.248

Web Browser, MS Outlook, LOTUS

IP

4.1. Network characteristics (Cntd)

Characteristics of PSTN and IP networks 

  PSTN IP Network

Bandwidth Fixed Variable

Technology Circuit-switched Packet-switched

Call handling Connection-oriented Connectionless-oriented

Quality Guaranteed limit No guarantee on delay, jitter and loss on transmission quality

4.2. PSTN/Internet convergence for data services: Narrowband Internet access

(local area)

LEXLEX

LEXLEX

(local area)

(local area)LEXLEX

LEXLEX

ISP

LEXLEX

AccessPoP

AccessPoP

LEXLEXLEXLEX

LEXLEX

AccessPoP

AccessPoP

AccessPoP

AccessPoP

CentralPoP

CentralPoP

PSTN

trunk(ISDN PRI)

trunk(SS7)

LEXLEX LEX - Local ExchangePoP – Point-of-PresenceISP – Internet Service Provider

Internet access methods

Home Network Intermediate Network

modem bank/access server/router

Access Devices

ISP

X

ATM/FR/LLATM/FR/LL

ISP PoPCorporate PoP

accessserver /router

POTSISDN

xDSLcablemodem

X

CATVCATV

FR - Frame RelayLL – Leased line

Virtual PoP(VPOP)

FR/ATM/LL

POTS/ISDN

Narrowbanddial-in accessNarrowbanddial-in access

Narrowbanddial-in access

withvirtual POP

Narrowbanddial-in access

withvirtual POP

Broadbandaccess

Broadbandaccess

Corporate leased line

access

Corporate leased line

access

ISP

bac

kbon

eIS

P b

ackb

one

PSTN

PSTNX

BroadbandaccessBroadbandaccess

4.3. PSTN/Internet convergence for voice servicesA. Converged network

GatewayGateway

LAN PC LANLAN LAN

Modem/Router

RouterServer

Router

App server

Res. house

PBX

Branch officeHQ office

IP-based public/private network

VoIP Call Processing and Protocol Flows

PS

TN

/IS

DN

B. Network scenarios for VoIP

GatekeeperCall ProcessingName’s ServerOAM Server

RAS

POP

PS

TN

/IS

DN

Internet

64 kbit/s speechVoice over IPMessage interface to central server

PC to Phone

Phone to PhonePhone to PC

PC to PC

VoiceVoice

Voice IWU(Gateway)

RAS

POP

Voice IWU(Gateway)

S0urce

Destination

Registration, Admission, and Status Protocol (RAS)

VoIP components and their functionsMedia Gateway• Packetizes voice • Supports telephone signaling• Applies audio compression• Provides connection control (mapping signaling protocols and addresses: E.164 IP address)• Tags voice packets using QoS mechanisms (DiffServ, Priority,…)Router• Recognizes voice packet and tags it accordingly• Prioritizes packets as needed• Manages bandwidth allocation• Provides queuing of traffic overflow

Gatekeeper - media gateway controller• MGC acts as the master controller of a media gateway• Supervises terminals attached to a network• Provides a registration of new terminals• Manages E.164 addresses among terminals

VoIP components

Intranet/Internet

(IP Network)

Intranet/Internet

(IP Network)Router

Gatekeeper

VoIPTerminals

Router

Gateway(Voice IWU)

PSTN/ISDN

PSTN/ISDN ATMATM

PBX

VoIPTerminals

Gatekeeper

Gateway(Voice IWU)

C. VoIP signaling protocols

• VoIP signaling protocols are the enablers of the VoIP network

• Centralized and distributed VoIP architectures• Call control is implemented by call-control software

running on servers (gatekeepers, proxy/RS, MGC) • Gatekeepers communicate with voice gateways,

end-user handsets or PCs using call-control protocols.

VoIP signaling protocols: 1. H.323, ITU-T

• H.323 - first call control standard for multimedia networks. Was adopted for VoIP by the ITU in 1996• H.323 is an ITU Recommendation that defines “packet-based

multimedia communications systems.” In other words, H.323 defines a distributed architecture for creating multimedia applications, including VoIP.

• H.323 is actually a set of recommendations that define how voice, data and video are transmitted over IP-based networks • The H.323 recommendation is made up of multiple call control protocols. The audio streams are transacted using the RTP/RTCP • In general, H.323 was too broad standard without sufficient efficiency. It also does not guarantee business voice quality

H.323 call setup process

VoIP signaling protocols: 2. SIP - Session Initiation Protocol, IETF (Internet

Engineering Task Force)

• SIP - standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality. Protocol claims to deliver faster call-establishment times.

• SIP works in the Session layer of IETF/OSI model. SIP can establish multimedia sessions or Internet telephony calls. SIP can also invite participants to unicast or multicast sessions.

• SIP supports name mapping and redirection services. It makes it possible for users to initiate and receive communications and services from any location, and for networks to identify the users wherever they are.

2. SIP - Session Initiation Protocol, IETF (Internet Engineering Task Force) (Cntd)

•SIP – client-server protocol, Rq from clients, Rs from servers. Participants are identified by SIP URLs. Requests can be sent through any transport protocol, such as UDP, or TCP.

•SIP defines the end system to be used for the session, the communication media and media parameters, and the called party's desire to participate in the communication.

•Once these are assured, SIP establishes call parameters at either end of the communication, and handles call transfer and termination.

SIP Proxy operation

SIP Redirect Server

VoIP signaling protocols : 3. MGCP/Megaco/H.248

• MGCP - Media Gateway Control Protocol, IETF [Telcordia (formerly Bellcore)/Level 3/Cisco] also known as IETF RFC 2705, defines a centralized architecture for creating multimedia applications, including VoIP.

• MGCP – control protocol that specifically addresses the control of media gateways

How MGCP coordinates the Media Gateways

Megaco/H.248

•Megaco/H.248 (IETF, ITU) Megaco, also known as IETF RFC 2885 and ITU Recommendation H.248, defines a centralized architecture for creating multimedia applications, including VoIP which combines elements of the MGCP and the H.323, ITU (H.248)

•The main features of Megaco - scaling (H.323) and multimedia conferencing (MGCP)

Real-time Transport Protocol (RTP)

• Real-Time Transport Protocol (RTP), also known as IETF RFC 1889, defines a transport protocol for real-time applications. Specifically, RTP provides the transport to carry the audio portion of VoIP communication

• RTP is used by all the VoIP signaling protocols

• RTP provides end-to-end delivery services for data with real-time characteristics • RTP is an application service built on UDP, so it is connectionless, with best-effort delivery.

Real-time Transport Control Protocol (RTCP)

•RTCP is the optional companion protocol to RTP •The primary function of RTCP is to provide feedback on the quality of the data distribution being accomplished by RTP. •RTCP enables administrators to monitor the quality of a call session by tracking packet loss, latency (delay), jitter•Bandwidth calculations for the protocol. Administrators need to limit the control traffic of RTCP to a small and known fraction of the session•RFC specifications recommend that the fraction of the session bandwidth allocated to RTCP be fixed at five percent of RTP traffic.

Which Standard?

1. H.323 H.323, with its roots in ISDN-based video-conferencing,has served its purpose of helping to transitionthe industry to IP telephony. Today, however, itscircuit switched heritage makes H.323 complex toimplement, resource intensive, and difficult to scale.Vendors and service providers are now de-emphasizingH.323’s role in their IP voice communicationsstrategies.

Which Standard?(Cntd.)

2. SIPSIP is ideal for IP voice and will play an importantrole for next generation service providers and distributedenterprise architectures. SIP suffers from someof the limitations of H.323 in that it has become acollection of IETF specifications, some of which arestill under definition. The other similarity withH.323 is that SIP defines intelligent end points andvendors have found this approach to be more costlyand less reliable.

Which Standard?(Cntd.)

3. MGCP/MEGACO/H.248In contrast to SIP, the MGCP/MEGACO standards

both centralize the control of simple telephones.

This is popular in environments where both cost and

control are important issues, which is certainly the

case in the enterprise environment where the PC an

be used to augment features and functionality.

Details of signaling protocols

D. VoIP scenarios: Phone-to-Phone

PS

TN

/IS

DN

VoIP Server(Gatekeeper)

RAS

POP

PS

TN

/IS

DN

InternetVoice

Voice

Voice IWU(Gateway A)

RAS

POP

Voice IWU(Gateway B)

Basic Call "Phone-to-Phone" A-Subscriber dials IWU E.164 number Normal Call Setup (a) between A-Subscriber and A-IWU Announcement from A-IWU to user Input of A-Subscriber E.164 Number, PIN and B-Subscriber E.164 Number (via multi-

frequency code) (SP) Call setup (b) within the Internet between A-IWU and B-IWU (routing functions

are in gatekeeper) Normal Call Setup (a) between B-IWU and B-Subscriber.

A B

A B

MGCP

(a) (b) (a)

VoIP scenarios: PC-to-Phone

PS

TN

/IS

DN

VoIP Server(Gatekeeper)

RAS

POP

PS

TN

/IS

DN

InternetVoice

Voice

Voice IWU(Gateway)

RAS

POP

Voice IWU(Gateway)

Basic Call "PC-to-Phone" PC needs VoIP software (support on of Signaling Protocols) Normal Internet login (a) of A-Subscriber Access to VoIP Server Input PIN and B-Subscriber E.164 Number (SP) Call setup (b) within the Internet between A-subscriber and B-IWU (routing

functions are in gatekeeper) Normal Call Setup (a) between B-IWU and B-Subscriber.

(b)

(b)

(a)

(a)

AB

AB

VoIP scenarios: Phone-to-PC

PS

TN

/IS

DN

VoIP Server(Gatekeeper)

RAS

POP

PS

TN

/IS

DN

InternetVoice

Voice

Voice IWU(Gateway)

RAS

POP

Voice IWU(Gateway)

Basic Call "Phone to PC" PC needs VoIP software (support on of Signaling Protocols) Normal Internet login (a) of B-Subscriber and registration at gatekeeper (E.164 to IP

address mapping) A-Subscriber dials IWU E.164 number Normal Call Setup (a) between A-Subscriber and A-IWU Input of A-Subscriber E.164 Number, PIN and B-Subscriber E.164 Number (SP) call setup call setup (b) within the Internet between A-IWU and B-subscriber PC (routing

functions and address mapping are in gatekeeper)

MGCP

(a)

(a)

(b)

(b)A

B

A B

E. Difference between VoIP and IP-T

• Voice over IP (VoIP) indicates that an analog voice signal has been digitized and

converted into the packet format used by IP. This is done in order to allow telephony and

other audio signals to be transported over the same network as regular data traffic.

Thus, VoIP refers to a conversion and transportation process.

• IP-Telephony is a service and it refers to VoIP over the public Internet. Although

technically feasible, the call quality is considered to be too variable for serious use by

business professionals. This comes from the fact that voice traffic has to be given

priority over data. However, VoIP is employed over managed IP infrastructures, e.g.

corporate intranets and the backbone networks of carriers.

• Unfortunately, the terms VoIP and IP-Telephony are often used interchangeably.

• Business VoIP service is defined as a high quality, reliable service capable of

sustaining mission-critical communications. High quality is defined as clear audio with

the absence of echo. A reliable service connection provides an error free transmission

with no service interruptions.

• IP-Telephony uses IP as the transport mechanism but it uses the public data

network (i.e., the Internet) to transmit voice packets. Because the Internet is an

unmanaged, non-voice engineered conglomerate of many networks, it cannot

guarantee bandwidth and timely delivery of voice packets, resulting in unacceptable

voice quality for business communications.

• By transmitting voice over a private managed IP data network, you can control all of the network characteristics required to ensure high-quality, reliable voice communications over a data network.

Business VoIP and IP-T

TeleGeography VoIP market predictions for 2005

In 2005 the international VoIP traffic will exceed 40 billion minutes with more than 30% annual growth.

Roadblocks to ConvergenceQuality of Service (QoS): The converged network must deliver the same QoS as the traditional Public Switched Telephone Network (PSTN); without it, video- and voice-over-IP are simply not viable. In an IP-based network, this requires handling data packets - to reduce loss, latency and jitter - with a QoS significantly higher than most data transmission networks are designed to support.

Reliability and Availability: The converged network must provide redundancy and fault-tolerance with "five nines" (99.999%) availability. While this is the standard level for most voice systems, many data networks lack the infrastructure to deliver such high availability across the entire system.

Bandwidth: The converged network must provide the necessary bandwidth to accommodate voice and video applications, which can demand considerably more than most data applications. While some efficiency schemes have proved useful in lowering the required bandwidth, most have been unable to effectively balance transmission speeds with voice and video quality.

Security: In traditional IP networks, packets are transmitted across shared segments, where the possibility exists that someone could decode packets and access secure information. A converged network must provide a new measure of encryption and security for voice traffic.

4.4. QoS issues and Reliability

• The number one issue operators have is:guarantee of Quality of Service

How to support voice traffic on backbone ?Actually, this is the number two issue

• The number one issue is:

Reliability of the data network

• Why? QoS makes only sense if the network is up and running all the time, hence reliable

A. Reliability

• Reliability in PSTN networks is already for 10s of years equal to the famous 99.999%, also called the 5 nines

• Operators are so used to this reliability that they take it for granted

• Why is it so important?– 99% means downtime of 3.7 days per year– 99.9% means downtime of 9 hours per year– 99.99% means downtime of 53 minutes per year– 99.999% means downtime of 5.5 minutes per year

• Traditional IP data equipment does not offer 5 nines reliability

Nines of availability and corresponding downtime

Reliability is a fundamental philosophy

Source: Infonetics Research, November 2001 The Tier 1 Service Provider Opportunity, US/Canada 2001

0% 25% 50% 75% 100%

Product Reliability

Best Price-to-Performance Ratio

Financial Stability

Leading-Edge Technology

Manufacturer’s ProductsAlready Installed

Pre-and post-salesservice and support

Manufacturer reputation

Manufacturer’s futureproduct offering

Leasing and Financing Options

Lowest Price

Sales and Marketing Services

Network Integration andDesign Services

100

82

73

73

64

64

45

45

27

27

18

9

Percent of Respondents Rating 6 to 7

Manufacturer Selection Criteria (Q61, n-11)

Source: Contingency Planning Research, a division of Eagle Rock Alliance Ltd

Reliability moved up the value scale

andnow rates

highest for Tier_1Service Providers

Reasons for system unavailability

Source: Gartner Group

• User Errors and Process: Change management, process inconsistency• Technology: Hardware, network links, environmental issues, natural disasters• Software Application: Software issues, performance and load, scalingOn average, computer system reliability is estimated at around 98.5%. This number includes not only the data networks and their components, but all the core business applications, servers, and mainframes.

Why are traditional IP Routers Unreliable? 7% Customer Premises Equipment

36% Router Operations Software/hardware

updates Configuration errors

21% Router Failures Hardware fault

intolerance Software quality

Physical Links 27%

Congestion 5% Network Engineering

Malicious 2%Unknown 2%

Source: University of Michigan

MPLS traffic engineering

Diversity of paths Fast Restoration

Software process isolation and redundancy 99.999 percent available hardware

Software upgrades Hardware upgrades

Common causes of downtime in IP networks Source: University of Michigan and Sprint study, October 2004

More than half of the problems causing downtime in IP networks- 59% - pertain to routing management issues.

More deeply, 36% of these problems are attributable to router misconfigurations, and 23% come from a category broadly described as "IP routing failures."  By contrast, of the remaining 41% of problems, link failures of some form account for 32%, and "other causes" comprise the remaining 9%.

Benefits of network reliability and losses due to failures

– Reductions in capital expenditure • eliminates requirement for duplicate hardware

configurations to support redundancy

– Reductions in ongoing operational costs• lower maintenance due to reduced number of

network elements• true non-service-interrupting upgrades• reduced floor space, cooling and power

requirements

– Revenue opportunities• no data session interruption during control

plane switchover will allow customers to achieve 99.999 percent availability

• increased customer retention

– Ability to offer low-risk SLAs• Five nines SLA

Business

Brokerage operationsCredit card/sales authorisationPay-per-viewHome shopping (TV)Airline reservationsTele-ticket salesPackage shippingAutomated teller machines

Source FCA

Cost per minute of downtime ($)

107,333 46,333

2500 1883 1500 1150 467 242

Commonly used techniques to “solve” reliability

• Instead of one reliable router, provide a reservation for each router

• Not quite the solution, isn’t it ?– double the price– need for extra interfaces for interconnection– but more importantly in case of failure, it takes time to

reroute the traffic from one to the other, in the meantime the ongoing calls are affected

• outage time can be quite long

B. QoS parameters - system performance metrics

QoS Applications

Interactive TV

Streaming media

E-mail, file transfer

Voice

Web browsing

• Bandwidth (Network Throughput)• Network/Devices Availability• Packet Delay• Packet Delay Variation - Jitter• Packet Loss

• There are no agreed quantifiable measures that define unambiguously QoS, as perceived by a user. Terms, such as “better”, “worse”, “high”, “medium”, “low”, “good”, “fair”, “poor”, are typically used, but these are subjective and cannot therefore be translated precisely into network level parameters that can subsequently be designed for by network planners.

• The end effect at the terminal is also heavily dependent upon issues such as compression algorithms, coding schemes, the presence of protocols for security, data recovery, re-transmission, etc., and the ability of applications to adapt to network congestion.

• However, network providers need performance metrics that they can agree with their peers (when exchanging traffic), and with service providers buying resources from them with certain performance guarantees.

• The following five system performance metrics are considered the most important in terms of their impact on the end-to-end QoS, as perceived by a user:

Bandwidth

• This is the effective data transfer rate measured in bps. It is not the same as the maximum capacity of the network, often erroneously called the network's bandwidth. A minimum rate of throughput is usually guaranteed by a service provider (who needs to have a similar guarantee from the network provider).

Availability (Reliability )

Ideally, a network should be available 100% of the time. Even a high-sounding figure as 99.5% translates into about an 44 hours of down time per month, which may be unacceptable to a large enterprise. Serious carriers strive for 99.9999% availability, which they refer to as "Six nines," and which translates into a downtime of 2.6 seconds per month

Delay

• The time taken by data to travel from the source to the destination is known as delay. The average time varies according to the amount of traffic being transmitted and the bandwidth available at that given moment. If traffic is greater than bandwidth available, packet delivery will be delayed.

Voice is a delay-sensitive application while most data applications are not. When voice packets are lost or arrive late they are discarded; the results are reduced voice quality.

• Components of delay - PrD, TD, PcD, JBD

Delays• Propagation delay: the time to travel across the network from end to end. It’s based

on the speed of light and the distance the signal must travel. For example, the propagation delay between Singapore and Boston is much longer than the propagation delay between New York and Boston.

• Transport delay: the time to get through the network devices along the path. Networks with many firewalls, many routers, congestion, or slow WANs introduce more delay than an overprovisioned LAN on one floor of a building.

• Packetization delay: the time for the codec to digitize the analog signal and build

frames – and undo it at the other end. The G.729 codec has a higher packetization delay than the G.711 codecs because it takes longer to compress and decompress the signal.

• Unless satellites are involved, the latency of a 5000 km voice call carried by a circuit-switched telephone network is about 25 ms. For the public Internet, a voice call may easily exceed 150 ms of delay because of: signal processing (digitizing and compressing the analogue voice input) and congestion (queuing). The important factor regarding delay is the propagation time along the cable (approx. 15 ms to cross the US and 30 ms to cross Russia).

Jitter (delay variation - the variability in packet arrival times at the destination)

• In general - voice packets must compete with non real-time data traffic # bursts structure of data traffic inside of the network # congestion problemResults are in varied arrival times for voice packets.

• When consecutive voice packets arrive at irregular intervals, the result is a distortion in the sound, which, if severe, can make the speaker unintelligible.

• Jitter has many causes, including: # variations in queue length # variations in the processing time needed to reorder packets that arrived out of order because they traveled over different paths # variations in the processing time needed to reassemble packets that were segmented by the source before being transmitted.

Sources of delays within the VoIP network

Packet loss - the percentage of undelivered packets in the data network

• Network devices, such as switches and routers, sometimes have to hold data packets in buffered queues when a link gets congested.

•If the link remains congested for too long, the buffered queues will overflow and data will be lost.

•The lost data packets must be retransmitted, adding, of course, to the total transmission time. In a well-managed network, packet loss will typically be less than 1% averaged over, say, a month.

• When data packets are lost, a receiving computer can simply request a retransmission. When voice packets are lost or arrive too late they are discarded of retransmitted. The result is in the form of gaps in the conversation (like a poor cell phone connection).

• Delay– E2E delay (Customer to Customer) < 250ms

(no echo canceling is required)– objective is < 150ms

• human ear starts to notice response delay above 150 ms

– 400 ms is unacceptable, except for satellite links • Delay variation or jitter

– E2E should be < 40ms – Delay variation: example of ETSI TIPHON

• <10 ms class 1 = gold• 10 ms to 20 ms class 2 = silver• 20 to 40 ms class 3 = bronze

QoS: Voice transport requirements

• Packet loss– E2E packet loss for voice should be < 2% – E2E 64k transparent should be more stringent < x %– ETSI TIPHON (voice)

• <0.5% class 1 = gold• 0.5% to 1% class 2 = silver• 1% to 2% class 3 = bronze

– Provided the E2E delay < 150 ms all above classes are acceptable

QoS: Voice transport requirements (Cntd)

Summary of network QoS requirements

Optimal network QoS parameters Limits of network QoS parameters

Delay – one way <= 100ms Delay – one way <= 150msJitter <= 40ms Jitter <= 75msPacket loss <= 1% Packet loss <= 3%

Internet performance measurements: RTT (from Belgium to a specific region)

1200

EuropeNorth-

AmericaSouth-

AmericaAsia Oceania Africa

Middle-East

1000

800

600

400

200

0

RTT (ms)

1998 353.3 417.3 882.6 841.3 738.8 808.4 1270.6

Sept-Oct 1998

2001 204.4 219.7 509.6 461.8 441.0 521.4 620.9

Mar-Apr 2001

Source: AlcatelRTT – round-trip time

Internet performance measurements

Source: NetIQ Corp.One-way delay = receiver timestamp – sender timestamp

Distance 10 100 1000 10.000 100.000 kmSTR – Stratosphere balloonLEO – Low-orbit satellite MEO – Middle-orbit satellite GEO – Geostationary-orbit satellite

Delays for different satellite communications systems

Internet performance measurements: Packet Loss (from Belgium to a specific region)

30%

EuropeNorth-

AmericaSouth-

AmericaAsia Oceania Africa

Middle-East

25%

20%

15%

10%

5%

0%

PacketLoss (%)

1998 11.2% 15.3% 17.0% 26.6% 12.6% 14.4% 23.4%

Sept-Oct 1998

2001 3.7% 2.4% 5.8% 12.1% 3.0% 10.1% 10.2%

Mar-Apr 2001

Source: Alcatel

C. State of IP networking today – from the QoS point of view

• IP FUD (fear, uncertainty and doubt)– IP is NOT just traditional backbone technology– Voice over IP today? Yes, but better - over ATM for

quality– Video distribution?

State of IP networking today (Cntd)

• To move to profitable IP-based services we need reliable, scalable, QoS aware, secure IP network– Online gaming/trading

• you’re about to win a game or complete a trade when a router reboots, and you lose your link.

The same problem, but with radically different consequences

– Streamed audio/video (Internet radio, TV)• a software upgrade during the season cliff-hanger

of your favorite show• a virus attack crashing a router in the last 20

seconds of the World Cup final

Key drivers affecting the Internet

• Today: not only voice matters:– Multimedia traffic explosion due to:

• the advent of real-time interactive multimedia applications (videoconference, 3-D animation/games/telemedicine…)

– Virtual Private Networks: Migration of business traffic from data to IP based networks to

• reduce expenses and operational complexity• provide improved connectivity to customers, business

partners and employees

• For all these applications, reliability and QoS are mandatory

D. QoS guaranteesPossible approaches to the problem

1. Over-provisioning the core network - simliciter

2. Congestion avoidance mechanisms by reservation

3. Service differentiation using IP QoS mechanisms

1. Over-provisioning the core network

# Assumption: physical bandwidth is available to scale and cheap bandwidth will be plentiful (based on FOC networks). The cost of bandwidth in the FOC backbones is decreasing, since:  

@ The supply of long-distance fiber in the ground currently exceeds

the demands for it

@ DWDM technology the specific cost of a capacity and the

specific cost of a transmission is almost zero

        # Provisioning can be planned

The capacity of the access tributaries is known, and the combined data rate cannot exceed the sum of the access links. As orders for faster access links are received, a decision can be made (taking also into account the current measured traffic load) whether or not it is necessary to upgrade the backbone capacity.

1. Over-provisioning the core network (Cntd)

– Ultimately, the main argument for the QoS decision via over-provisioning - the availability of fiber. So this does not apply to all networks, and, of course, not to the edges of the network

– Over-provisioning the core is a short-term solution. As access capacity progressively increases, backbone networks will become susceptible to congestion and overloading

Reservation and service differentiation - IP QoS mechanisms

• QoS on IP can be delivered on the base of mechanisms:

- IntServ (Integrated Services)

- DiffServ (Differentiated Services)

- MPLS

2. Reservation mechanisms•Integrated Services (IntServ)

# IETF Integrated Services (IntServ) Working Group developed a service model based on the principle of integrated resource reservation.

# The group of IntServ mechanisms (first of all, RSVP) refers to a group of methods providing a “hard” QoS.

# RSVP (Resource ReSerVation Protocol) mechanism is the most well known representative of the IntServ mechanisms (RFC 2205, 1997).

# RSVP is a signaling protocol according to which reservation and resource allocation is carried out to guarantee “hard” QoS. Reservation is accomplished for the certain IP packet flow before the main flow transmission start up.

# Hard requirements to network resources

Integrated Services (IntServ)

• Flow = stream of packets with common Source Address, Destination Address and port number

• Requires router to maintain state information on each flow; router determines what flows get what resources based on available capacity

IntServ components• Traffic classes

– best effort – controlled load (‘best-effort like’ w/o congestion) – guaranteed service (real-time with delay bounds)

• Traffic control – admission control – packet classifier – packet scheduler

IntServ components (cont.)

• Setup protocol: RSVP • “Path” msg from source to destination collects

information along the path; the destination gauges what the network can support, then generates a “Resv” msg

• If routers along the path have sufficient capacity, then resources back to the receiver are reserved for that flow; otherwise, RSVP error messages are generated and returned to the receiver

• Reservation state is maintained until the RSVP “Path” and “Resv” messages stop coming

IntServ/RSVP problems

• Scalability (processing of every individual flow on core Internet routers)

• Lack of policy control mechanisms

3. Service differentiation using IP QoS mechanisms

Differentiated Services (DiffServ)

• DiffServ concept and mechanisms # Necessity to develop more flexible mechanisms for providing QoS # The detailed specifications of DiffServ (RFC 2475) - in the middle 1999. # As against IntServ group the DiffServ methods provide a “relative” or “soft”

QoS. 

• The main idea of DiffServ mechanisms to provide differentiated services to a set of traffic classes characterized by various requirements to QoS parameters

• One of the central point of DiffServ model is the Service Level Agreement (SLA)

# SLA – the contract between the user and the service provider # SLA - basic features of users’ traffic and QoS parameters ensured by

providers # SLA - static or dynamic contract

Differentiated Services (DiffServ) - Cntd

• Main issues of QoS - priorities The support of a satisfactory QoS: - means for labeling flows with respect to their priorities - network mechanisms for recognizing the labels and acting on them

• According the IETF Differentiated Services model the network architecture includes two areas - edge segment and core segment

• In the edge routers a short tag is appended to each packet depending on its Class of Service (CoS)

• DS byte - ToS (IPv4) or TC (IPv6)

Differentiated Services (DiffServ) - Cntd

Network mechanisms • Edge routers

#Traffic Classification mechanism (to select the packets of one flow featured by common requirement to QoS) # Conditioning mechanism If necessary a part of packets can be discarded. # Shaping mechanism (if required)

• Backbone routers # Packets forwarding in compliance with the required QoS level # Two forwarding classes are specified - Expedited Forwarding (EF) and Assured

Forwarding (AF). # EF class provides the Premium Service (apps requiring forwarding with minimum

delay and jitter) # AF class maintains a lower QoS than EF class, but higher than BES # AF class identifies 4 classes of traffic and three levels of packet discarding – 12 types of traffic depending on the set of the required QoS

Differentiated Services (DiffServ) - Cntd

• Queuing mechanisms # Target - a control of a packet delay and jitter and elimination of possible losses # Based on priority level and type of traffic # Mechanisms • Priority Queuing• Weighted Fair Queuing • Class-Based Queuing

• In the past - QoS planners supported both IntServ and DiffServ. At present - DiffServ supplemented by RSVP at the edges. At the edges of the network, resources tend to be more limited, and there are not so many flows to maintain

Example - QoS in LANs

Ethernet’s QoS based on 802.1p/Q

• The IEEE 802.1Q standard adds four additional bytes to the standard 802.3 Ethernet frame

• Three-bit field provides Ethernet QoS

• Three priority bits create 8 Classes of Service (CoS) for packets traversing Ethernet networks

• For IP telephony, a binary value of 100 for 802.1p is recommended with both voice bearer and voice signalling

• Remaining part of four additional bytes is used for the virtual LAN (VLAN) ID

A. Data and Voice network performance requirements.

DATAFile transfer applications - big volumes, big resources, E-mails - small volumes, tolerance to delays and losses Using TCP

VoIP applicationsRelatively little bandwidth, but can’t tolerate large delays, variations, losses.

• Protocol units have different packet sizes• Packets are sent at different rates• TCP for data• RTP for voice• Packets are buffered and delivered to the destination differently

Delays caused by other applications, overloaded routers, or faulty switches may be inevitable for VoIP apps

4.5. Estimation of call quality

•Quality goal for a VoIP call the PSTN level of quality (“toll” quality)

•But what is in IP networks???

•We need to understand some of the different measurement standards for voice quality

•The leading subjective measurement of voice quality - Mean Opinion Score (MOS) – Recommendation ITU P.800 – but for telephone equipment!

•The Mean Opinion Score (MOS) described in ITU P.800 is a subjective measurement of call quality as perceived by the receiver. A MOS can range from 5 down to 1, using the following rating scale (see Table)

B. Standards for measuring call quality

The MOS is measured on a scale from 5 down to 1

This mapping between audio performance characteristics and a quality score makes the MOS (Mean Opinion Score) standard valuable for

network assessments, benchmarking, tuning, and monitoring

MOS in VoIP apps• MOS and other methods are based in older telephony approaches. These approaches are not very well suited to assessing call quality on a data network, as they can’t take into account the network issues of delay, jitter, and packet loss.

• Models don’t take into account E2E delay between the telephone speaker and listener. Excessive delay adversely affects MOS.

• Models show quality in one direction at a time.

• Models don’t scale to let you see the effect of multiple, simultaneous calls.

• Recommendation ITU G.107 introduced the E-model. The E-model is better suited for use in data network call quality assessment because it takes into account impairments specific to data networks.

• The output of an E-model calculation is a single scalar, called an “R-value” or R-factor derived from delays and equipment impairment factors. Once an R value is obtained, it can be mapped to an estimated MOS.

E-model

R-factor values from the E-model and corresponding MOS values

The R value, the output from the E-model, ranges from 100 down to 0, where 100 is excellent and 0 is poor. The calculation of an R value starts with the undistorted signal.

R-factor values from the E-model and corresponding MOS values (Cntd)

R-factor values from the E-model and corresponding MOS values (Cntd)

MOS•One-way delay•Percentage of packet loss•Packet loss burstiness•Jitter buffer delay•Data lost due to jitter buffer overruns•Behaviour of the codec.

R = R0

R = R0 – Is – Id – Ie + Awhere:• Is: channel’s noise impairments to the signal• Id: delays introduced from end to end• Ie: impairments introduced by the equipment, including packet loss• A: advantage factor (for example, mobile users may tolerate lower quality because of the convenience).

Calculating an R value

C. Codec’s selection• In audio processing - a codec is the hardware or software that samples the sound and defines the data rate of digital output. There are, each with different characteristics •Dozens of available codecs •Types of codecs correspond to the certain ITU standards

• First codecs - G.711a/G.711 - 64 kb/s (PCM) – ADC with no compression and high quality

• New generation of codecs based on new compression algorithms New codecs provide intelligible voice communications with reduced bandwidth consumption.

•The lower-speed codecs # G.726-32 (32 kb/s) # G.729 (8 kb/s) # G.723.1 family (6.3/5.3 kb/s)• New codecs consume less network bandwidth – bigger number of concurrent calls• BUT - bigger impairment on the quality of the audio signal than high-speed codecs, the compression reduces the clarity, introduces delay, and makes the voice quality very sensitive to a packet loss

Parameters of VoIP codecs

m

a

• MOS and R value include Pack delay and Jitter buffer delay• Common bandwidth – real bandwidth consumption: # Payload = 20 bytes/p (40 bytes/s) # Overhead includes 40 bytes of RTP header (20 IP + 8 UDP + 12 RTP)• G.723.1 – Quality is“Acceptable” only

a

m

1) Based on the specified bit-rate

2) Based on two voice frames per packet

Common voice codecs and corresponding audio quality

Codec R-factor MOS G.711 93.2 4.4G.729 82.2 4.1G.732.1m 78.2 3.9G.723.1a 74.2 3.75

-

Codecs’ comparison

m

a

Codec R-factor MOS G.711 93.2 4.4G.729 82.2 4.1G.732.1m 78.2 3.9G.723.1a 74.2 3.75

m a

m a

Codecs’ comparison (Cntd)• Any lost datagram impairs the quality of the audio signal. Data loss is thus a key call-quality impairment factor in calculating the MOS.

• Random loss –simplest loss model

# One datagram is lost or two datagrams are lost time by time

# Small effect inside of delay limit (<=150 ms)

• Bursts of loss

# Quality degrades most significantly

# More than two consecutive datagrams are lost

• 5% random packet loss (upper Figure)

# MOS starts at around 4 for the G.711 codec

• 5% bursty packet loss (Figure below)

# MOS starts at around 3.5 for the same codec

• The effect of bursty loss is even greater on the other codecs

Codec R-factor MOS G.711 93.2 4.4G.729 82.2 4.1G.732.1m 78.2 3.9G.723.1a 74.2 3.75

List of VoIP network design tips

Main factors QoS of VoIP - delay, jitter and packet loss. Following design tips could be useful during VoIP deployment process Use the G.711 codec on E2E if a capacity is enough # G.711 codec gives the best voice quality - no compression, minimum delay, less sensitive to packet loss # Other codecs - G.729 and G.723 use compression. Results – economy of a bandwidth, but delay is introduced and the voice quality very sensitive to lost packets Keep packet loss well below 1% and avoid bursts of consecutive lost packets # Sources of packet loss - channel noise, traffic congestion and jitter buffer size # Tools - Increased bandwidth and TE can often reduce network congestion, which, in turn, reduces jitter and packet loss Use a small speech frame size and reduce the number of speech frames per packet # Using small packets/datagrams (in ms) - impact of the packet loss is less than losing a big packets # One of standard size - 20ms of speech frame per datagram. Of course, using small packets increases an overhead conditions, because each packet requires its own fixed-size header Always use codecs with packet-loss concealment (PLC) # PLC masks the loss of a packet or two by using information from the last good packet # PLC helps with random packet loss

List of VoIP network design tips (Cntd)

Actively minimize one-way delay, keeping it below 150msE2E Delay = PrD + TD + PcD + JBD <= 150ms

• PrD – physical distance (3-5 mcs/km) # Routing – network path ADAP

• TD – all network devices (routers, gateways, TE tools, firewalls) # Factors – number of hopes, software/hardware processing

• PD - fixed time needed for the AD conversion # G.711 - adds 1ms # G.723 – adds 67.5ms # E2E – the same type of codecs

• JBD - to decrease variations in packet arrival rates # Larger jitter buffer than in a network where the delay is already high.

Avoid using slow speed links

Use RTP header compression for links with the lack of capacity # CRTP can reduce the 40-byte RTP headers to a tenth of their original size # Decreasing the bandwidth consumption # BUT - it adds latency.

Use call admission to protect against too many concurrent calls#Call Admission Control

Use priority scheduling for voice traffic

# DiffServ (EF)# Queuing mechanisms - giving VoIP higher priority

Get your data network ready for VoIP# In general, unsatisfactory data networks# Network should be fully upgraded and tuned, before starting a VoIP deployment

List of VoIP network design tips (Cntd)

QoS - Concluding remarks

• Real-time applications should be supported by manufacturers’ products due to reliability and Quality of Service capabilities

• QoS demanding applications come from:–introduction of multimedia–bypass of voice networks (e.g. Long-Distance Bypass)–growth in the voice networks–migration of voice to data networks

TeleGeography VoIP market predictions for 2005

In 2005 the international VoIP traffic will exceed 40 billion minutes with more than 30% annual growth.

Convergence of PSTN and data networks - concluding remarks# Debates are over • Q1 2004 - about 12% of all phone calls use VoIP # How legacy voice will migrate toward IP?   Many factors:• End-user (R&B) behavior to adopt VoIP• Availability of cost-efficient and friendly terminals• End of life of legacy PSTN equipment • Sharp increase of OPEX

# Early adaptors of VoIP - gamers and abroad communicator use VoIP already – technology reduces communications costs # Business VoIP – VPN. Available QoS# Main benefits come from real-time communications applications• CTI Apps# Unified messaging # Unified communications # Web contact centers

AppendixiLBC (internet Low Bitrate Codec)

VOCAL Technologies, Ltd.

• iLBC - speech codec suitable for robust voice communication over IP. •The codec is designed for narrow band speech and results in a payload bit rate of 13.33 kbit/s with an encoding frame length of 30 ms and 15.20 kbps with an encoding length of 20 ms.

Features• Bit rate 13.33 kbps (399 bits, packetized in 50 bytes) for the frame size of 30 ms • 15.2 kbps (303 bits, packetized in 38 bytes) for the frame size of 20 ms • Basic quality higher then G.729A, high robustness to packet loss • Computational complexity in a range of G.729A                                                                                                                           

Codec comparison