Download Convergence VoIP

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
LE
LE
PBX
PSTN
PBX
Branch office
PBX
HQ office
PSTN Call Processing and Protocol Flows
4.1. Network characteristics (Cntd)
• Data networks – 60s, ARPA
• Basic principals – packet switching, connectionlessoriented (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
Server
App server
Router
Router
Public/private network
Modem/router
Branch office
Res. house
HQ office
Web Browser, MS Outlook, LOTUS
H.323, SIP, RTP, RSVP, MGCP,
MEGACO/H.248
HTTP, FTP
TCP
UDP
IP
Ethernet, ATM, FR, PPP
Physical layer
4.1. Network characteristics (Cntd)
Characteristics of PSTN and IP networks
Bandwidth
PSTN
IP Network
Fixed
Variable
Technology
Circuit-switched
Call handling
Connection-oriented
Quality
Guaranteed limit
on delay, jitter and loss
Packet-switched
Connectionless-oriented
No guarantee
on transmission quality
4.2. PSTN/Internet convergence for data services:
Narrowband Internet access
trunk
(ISDN PRI)
Access
PoP
LEX
(local area)
LEX
ISP
trunk
(SS7)
LEX
Central
PoP
(local area)
Access
PoP
LEX
LEX
LEX
PSTN
LEX
(local area)
LEX
LEX
Access
PoP
LEX - Local Exchange
PoP – Point-of-Presence
ISP – Internet Service Provider
Internet access methods
Narrowband
dial-in access
Access Devices
ISP PoP
Corporate PoP
POTS/ISDN
POTS
ISDN
PSTN
xDSL
cable
modem
Broadband
access
X
X
access
server /
router
CATV
PSTN
X
Broadband
access
ISP backbone
ISP
modem bank/
access server/
router
ATM/FR/LL
Virtual PoP
(VPOP)
Narrowband
dial-in access
with
virtual POP
FR/ATM/LL
Home Network
Corporate
leased line
access
Intermediate Network
FR - Frame Relay
LL – Leased line
4.3. PSTN/Internet convergence for voice services
A. Converged network
App server
Server
Router
Router
IP-based public/private network
Modem/Router
Gateway
LAN
LAN
PC
Gateway
LAN
PBX
Branch office
Res. house
HQ office
VoIP Call Processing and Protocol Flows
B. Network scenarios for VoIP
Internet
Voice
PSTN/ISDN
POP
POP
RAS
RAS
Voice IWU
(Gateway)
Voice IWU
(Gateway)
PSTN/ISDN
Voice
Gatekeeper
Call Processing
Name’s Server
OAM Server S
64 kbit/s speech
Voice over IP
Message interface to central server
Registration, Admission, and Status Protocol (RAS)
0
u
r
c
e
Destination
PC to PC
Phone to PC
PC to Phone
Phone to Phone
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
VoIP components
Gatekeeper
Gatekeeper
Intranet/
Internet
(IP Network)
VoIP
Terminals
Router
Router
VoIP
Terminals
Gateway
(Voice IWU)
PSTN/
ISDN
Gateway
(Voice IWU)
ATM
PBX
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 callestablishment 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 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.323’s role in their IP voice communications
strategies.
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.
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.
Details of signaling protocols
D. VoIP scenarios: Phone-to-Phone
Voice
(a)
A
PSTN/ISDN
POP
Voice
B
POP
RAS
RAS
(b)
Voice IWU
(Gateway A)
MGCP
Voice IWU
(Gateway B)
PSTN/ISDN
A
Internet
(a)
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 multifrequency 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.
B
VoIP scenarios: PC-to-Phone
Voice
(b)
A
PSTN/ISDN
(a)
POP
RAS
Voice IWU
(Gateway)
B
POP
RAS
(b)
Voice IWU
(Gateway)
PSTN/ISDN
Internet
A
Voice
VoIP Server
(Gatekeeper)
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.
(a)
B
VoIP scenarios: Phone-to-PC
Voice
(a)
A
PSTN/ISDN
POP
(b)
POP
RAS
B
RAS
(b)
Voice IWU
(Gateway)
MGCP
Voice
Voice IWU
(Gateway)
(a)
PSTN/ISDN
A
Internet
VoIP Server
(Gatekeeper)
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)
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 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.
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 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 voiceover-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%
99.9%
99.99%
99.999%
means downtime of
means downtime of
means downtime of
means downtime of
3.7 days per year
9 hours per year
53 minutes per year
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
Manufacturer Selection Criteria (Q61, n-11)
Product Reliability
Reliability moved up
the value scale
and
now rates
highest for Tier_1
Service Providers
100
82
Best Price-to-Performance Ratio
Financial Stability
73
Leading-Edge Technology
73
Manufacturer’s Products
Already Installed
64
Pre-and post-sales
service and support
64
Manufacturer reputation
45
Manufacturer’s future
product offering
45
Leasing and Financing Options
27
Lowest Price
27
Sales and Marketing Services
Source: Contingency Planning Research, a division of
Eagle Rock Alliance Ltd
Source: Infonetics Research, November 2001
The Tier 1 Service Provider Opportunity, US/Canada 2001
Network Integration and
Design Services
18
9
0%
25%
50%
75%
100%
Percent of Respondents Rating 6 to 7
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.
Why are traditional IP Routers Unreliable?
7% Customer Premises Equipment
Unknown 2%
Malicious 2%
Congestion 5%
 Network Engineering

36% Router Operations
 Software/hardware
updates
 Configuration errors
MPLS traffic
engineering


Software upgrades
Hardware upgrades
Physical Links 27%


21% Router Failures
 Hardware fault intolerance
 Software quality
Diversity of paths
Fast Restoration


Software process isolation and redundancy
99.999 percent available hardware
Source: University of Michigan
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 operations
Credit card/sales
authorisation
Pay-per-view
Home shopping (TV)
Airline reservations
Tele-ticket sales
Package shipping
Automated 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
• Bandwidth (Network Throughput) QoS Applications
• Network/Devices Availability
• Packet Delay
Interactive TV
• Packet Delay Variation
- Jitter
Voice
• Packet Loss
Streaming media
Web browsing
E-mail, file transfer
• 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 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.
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).
QoS: Voice transport requirements
• 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 (Cntd)
• 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
Summary of network QoS requirements
Optimal network QoS parameters
Delay – one way <= 100ms
Jitter
<= 40ms
Packet loss
<= 1%
Limits of network QoS parameters
Delay – one way <= 150ms
Jitter
<= 75ms
Packet loss
<= 3%
Internet performance measurements:
RTT (from Belgium to a specific region)
1200
RTT (ms)
Sept-Oct 1998
Mar-Apr 2001
1000
800
600
400
200
0
1998
2001
Europe
NorthAmerica
SouthAmerica
Asia
Oceania
Africa
MiddleEast
353.3
204.4
417.3
219.7
882.6
509.6
841.3
461.8
738.8
441.0
808.4
521.4
1270.6
620.9
RTT – round-trip time
Source: Alcatel
Internet performance measurements
One-way delay = receiver timestamp – sender timestamp
Source: NetIQ Corp.
Delays for different satellite communications systems
Distance
STR – Stratosphere balloon
LEO – Low-orbit satellite
MEO – Middle-orbit satellite
GEO – Geostationary-orbit satellite
10
100
1000 10.000 100.000
km
Internet performance measurements:
Packet Loss (from Belgium to a specific region)
30%
Packet
Loss (%)
Sept-Oct 1998
Mar-Apr 2001
25%
20%
15%
10%
5%
0%
1998
2001
Europe
NorthAmerica
SouthAmerica
Asia
Oceania
Africa
MiddleEast
11.2%
3.7%
15.3%
2.4%
17.0%
5.8%
26.6%
12.1%
12.6%
3.0%
14.4%
10.1%
23.4%
10.2%
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 guarantees
Possible 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 overprovisioning - 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
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 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
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)
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
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.
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.
R-factor values from the E-model and corresponding MOS
values (Cntd)
R-factor values from the E-model and corresponding
MOS values (Cntd)
•One-way delay
•Percentage of packet loss
•Packet loss burstiness
•Jitter buffer delay
•Data lost due to jitter buffer overruns
•Behaviour of the codec.
MOS
Calculating an R value
R = R0
R = R0 – Is – Id – Ie + A
where:
• 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).
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
• 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
m
a
m
a
1) Based on the specified bit-rate
2) Based on two voice frames per packet
Common voice codecs and corresponding audio quality
-
Codec
G.711
G.729
G.732.1m
G.723.1a
R-factor
93.2
82.2
78.2
74.2
MOS
4.4
4.1
3.9
3.75
Codecs’ comparison
m
a
Codec
G.711
G.729
G.732.1m
G.723.1a
R-factor
93.2
82.2
78.2
74.2
MOS
4.4
4.1
3.9
3.75
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
m
a
# 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
m
a
Codec
G.711
G.729
G.732.1m
G.723.1a
R-factor
93.2
82.2
78.2
74.2
MOS
4.4
4.1
3.9
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 150ms
E2E 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.
List of VoIP network design tips (Cntd)
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
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
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
Codec comparison