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

1
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

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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)

3
PSTN
LE
LE
PBX
PSTN
PBX
PBX
Branch office
HQ office
4
PSTN Call Processing and Protocol Flows
5
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 its not
  critical for data, but critical for other
  possible apps

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Data network
App server
Server
Router
Router
Public/private network
Modem/router
HQ office
Res. house
Branch office
7
Web Browser, MS Outlook, LOTUS
HTTP, FTP
H.323, SIP, RTP, RSVP, MGCP, MEGACO/H.248
TCP
UDP
IP
Ethernet, ATM, FR, PPP
Physical layer
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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

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4.2. PSTN/Internet convergence for data services
Narrowband Internet access
trunk (ISDN PRI)
LEX
Access PoP
LEX
(local area)
ISP
trunk (SS7)
LEX
Central PoP
LEX
Access PoP
(local area)
LEX
LEX
Access PoP
PSTN
LEX
(local area)
LEX
LEX - Local Exchange PoP Point-of-Presence ISP
Internet Service Provider
LEX
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Internet access methods
ISP
Narrowbanddial-in access
Access Devices
ISP PoP Corporate PoP
POTS/ISDN
ISP backbone
X
POTS ISDN
PSTN
X
X
Broadbandaccess
PSTN
modem bank/ access server/ router
xDSL
cable modem
ATM/FR/LL
access server / router
CATV
Broadbandaccess
Narrowbanddial-in access withvirtual POP
Virtual PoP (VPOP)
FR/ATM/LL
Home Network
Intermediate Network
Corporate leased lineaccess
FR - Frame Relay LL Leased line
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4.3. PSTN/Internet convergence for voice
servicesA. Converged network
App server
Server
Router
Router
IP-based public/private network
Modem/Router
Gateway
Gateway
LAN
LAN
PC
LAN
LAN
PBX
Res. house
Branch office
HQ office
12
VoIP Call Processing and Protocol Flows
13
B. Network scenarios for VoIP
Internet
Voice
Voice
POP
POP
RAS
RAS
PSTN/ISDN
PSTN/ISDN
Voice IWU (Gateway)
Voice IWU (Gateway)
Gatekeeper Call Processing Names Server OAM
Server
Destination
S 0 u r c e
64 kbit/s speech Voice over IP Message interface
to central server
Registration, Admission, and Status Protocol (RAS)
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VoIP components and their functions

• Media 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

15
VoIP components
Gatekeeper
Intranet/ Internet (IP Network)
VoIP Terminals
Gatekeeper
Router
Router
VoIP Terminals
Gateway (Voice IWU)
Gateway (Voice IWU)
PSTN/ ISDN
ATM
PBX
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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.

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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

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H.323 call setup process
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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.

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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.

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SIP Proxy operation
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SIP Redirect Server
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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

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How MGCP coordinates the Media Gateways
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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)

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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.

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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.

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Which Standard?

• 1. H.323
• H.323, with its roots in ISDN-based
  video-conferencing,
• has served its purpose of helping to transition
• the industry to IP telephony. Today, however, its
• circuit switched heritage makes H.323 complex to
• implement, resource intensive, and difficult to
• scale.
• Vendors and service providers are now
  de-emphasizing
• H.323s role in their IP voice communications
• strategies.

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Which Standard?(Cntd.)

• 2. SIP
• SIP is ideal for IP voice and will play an
  important
• role for next generation service providers and
  distributed
• enterprise architectures. SIP suffers from some
• of the limitations of H.323 in that it has become
  a
• collection of IETF specifications, some of which
  are
• still under definition. The other similarity with
• H.323 is that SIP defines intelligent end points
  and
• vendors have found this approach to be more
  costly
• and less reliable.

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Which Standard?(Cntd.)

• 3. MGCP/MEGACO/H.248
• In 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.

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Details of signaling protocols
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D. VoIP scenarios Phone-to-Phone
Voice
Voice
Internet
A
B
POP
POP
RAS
RAS
PSTN/ISDN
PSTN/ISDN
(a)
(a)
(b)
Voice IWU (Gateway A)
Voice IWU (Gateway B)
A
B
MGCP
VoIP Server (Gatekeeper)

• 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.

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VoIP scenarios PC-to-Phone
Voice
Voice
Internet
A
B
POP
POP
(a)
RAS
RAS
(b)
PSTN/ISDN
PSTN/ISDN
(a)
(b)
Voice IWU (Gateway)
Voice IWU (Gateway)
A
VoIP Server (Gatekeeper)
B

• 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.

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VoIP scenarios Phone-to-PC
Voice
Voice
Internet
(b)
A
POP
POP
B
RAS
RAS
(a)
PSTN/ISDN
PSTN/ISDN
(a)
(b)
Voice IWU (Gateway)
Voice IWU (Gateway)
MGCP
A
VoIP Server (Gatekeeper)
B

• 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 (b) within the Internet between
  A-IWU and B-subscriber PC (routing functions and
  address mapping are in gatekeeper)

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• 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.

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Business VoIP and IP-T

• 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.

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TeleGeography VoIP market predictions for 2005

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

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Roadblocks to Convergence

• Quality 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.

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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

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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

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Nines of availability and corresponding downtime
42
Reliability is a fundamental philosophy
Manufacturer Selection Criteria (Q61, n-11)
Product Reliability
100
Reliability moved up the value scale and now
rates highest for Tier_1 Service Providers
82
Best Price-to-Performance Ratio
Financial Stability
73
Leading-Edge Technology
73
Manufacturers Products Already Installed
64
Pre-and post-sales service and support
64
45
Manufacturer reputation
Manufacturers futureproduct offering
45
Leasing and Financing Options
27
27
Lowest Price
Sales and Marketing Services
18
Source Contingency Planning Research, a division
of Eagle Rock Alliance Ltd
Network Integration and Design Services
9
0
25
50
75
100
Source Infonetics Research, November 2001 The
Tier 1 Service Provider Opportunity, US/Canada
2001
Percent of Respondents Rating 6 to 7
43
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, scaling
• On 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.

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Why are traditional IP Routers Unreliable?
7 Customer Premises Equipment
Unknown 2
Malicious 2

• 36 Router Operations
• Software/hardware updates
• Configuration errors

• Congestion 5
• Network Engineering

Physical Links 27

• 21 Router Failures
• Hardware fault intolerance
• Software quality

Source University of Michigan
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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.

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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

47
Commonly used techniques to solve reliability

• Instead of one reliable router, provide a
  reservation for each router
• Not quite the solution, isnt 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

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B. QoS parameters - system performance metrics

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

QoS Applications
Interactive TV
Voice
Streaming media
Web browsing
E-mail, file transfer
49

• 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

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• 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).

51
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
52
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

53
Delays

• Propagation delay the time to travel across the
  network from end to end. Its 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).

54

• 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 problem
• Results 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.

55
Sources of delays within the VoIP network
56

• 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).

57
QoS Voice transport requirements

• Delay
• E2E delay (Customer to Customer) lt 250ms (no
  echo canceling is required)
• objective is lt 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 lt 40ms
• Delay variation example of ETSI TIPHON
• lt10 ms class 1 gold
• 10 ms to 20 ms class 2 silver
• 20 to 40 ms class 3 bronze

58
QoS Voice transport requirements (Cntd)

• Packet loss
• E2E packet loss for voice should be lt 2
• E2E 64k transparent should be more stringent lt x
  
• ETSI TIPHON (voice)
• lt0.5 class 1 gold
• 0.5 to 1 class 2 silver
• 1 to 2 class 3 bronze
• Provided the E2E delay lt 150 ms all above classes
  are acceptable

59
Summary of network QoS requirements Optimal
network QoS parameters Limits of
network QoS parameters Delay one way lt 100ms
Delay one way lt
150ms Jitter lt 40ms
Jitter
lt 75ms Packet loss lt 1
Packet loss lt 3
60
Internet performance measurements RTT (from
Belgium to a specific region)
1200
RTT round-trip time
Source Alcatel
61
Internet performance measurements
Source NetIQ Corp.
One-way delay receiver timestamp sender
timestamp
62
Delays for different satellite communications
systems
Distance
10 100 1000 10.000 100.000
km STR Stratosphere balloon LEO Low-orbit
satellite MEO Middle-orbit satellite GEO
Geostationary-orbit satellite
63
Internet performance measurements Packet Loss
(from Belgium to a specific region)
30
Source Alcatel
64
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?

65
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
• youre 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

66
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

67
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

68
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  
• _at_ The supply of long-distance
  fiber in the ground currently exceeds
• the demands for it
• _at_ 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.

69
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

70
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

71

• 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

72
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

73
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

74
IntServ/RSVP problems

• Scalability (processing of every individual flow
  on core Internet routers)
• Lack of policy control mechanisms

75

• 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

76
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)

77
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

78
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

79

• Example - QoS in LANs
• Ethernets 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

80
4.5. Estimation of call quality

• A. Data and Voice network performance
  requirements.
• DATA
• File transfer applications - big volumes, big
  resources,
• E-mails - small volumes, tolerance to delays and
  losses
• Using TCP
• VoIP applications
• Relatively little bandwidth, but cant 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

81
B. Standards for measuring 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)

82
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
The MOS is measured on a scale from 5 down to 1
83

• 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 cant take into account the
  network issues of delay, jitter, and packet loss.
  
• Models dont 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 dont 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.

84
R-factor values from the E-model and
corresponding MOS values
E-model
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.
85
R-factor values from the E-model and
corresponding MOS values (Cntd)
86
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.

87
Calculating an R value

• R R0
• R R0 Is Id Ie A
• where
• Is channels 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).

88
C. Codecs 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

89
Parameters of VoIP codecs

• 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 isAcceptable only

m
a
90
m
a
1) Based on the specified bit-rate 2) Based on
two voice frames per packet
91
Common voice codecs and corresponding audio
quality
-
Codec
R-factor MOS
G.711
93.2
4.4 G.729
82.2
4.1 G.732.1m
78.2
3.9 G.723.1a
74.2
3.75
92
Codecs comparison
m a
Codec
R-factor MOS
G.711
93.2
4.4 G.729
82.2
4.1 G.732.1m
78.2
3.9 G.723.1a
74.2
3.75
93

• 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 (lt150 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

m a
m a
Codec R-factor MOS
G.711 93.2
4.4 G.729 82.2
4.1 G.732.1m 78.2
3.9 G.723.1a 74.2
3.75
94
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
95
List of VoIP network design tips (Cntd)

• Actively minimize one-way delay, keeping it below
  150ms
• E2E Delay PrD TD PcD JBD lt 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

96
Use call admission to protect against too many
concurrent callsCall Admission ControlUse
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)
97
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

98
TeleGeography VoIP market predictions for 2005

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

99
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 (RB) 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

100

• Appendix
• iLBC (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
•                                                   
                                                    
                      
•  

101
Codec comparison