Title: Part 2' Converged networks and services 4' Convergence of fixed networks
1Part 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
24.1. 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)
3PSTN
LE
LE
PBX
PSTN
PBX
PBX
Branch office
HQ office
4PSTN Call Processing and Protocol Flows
54.1. 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
6Data network
App server
Server
Router
Router
Public/private network
Modem/router
HQ office
Res. house
Branch office
7Web Browser, MS Outlook, LOTUS
HTTP, FTP
H.323, SIP, RTP, RSVP, MGCP, MEGACO/H.248
TCP
UDP
IP
Ethernet, ATM, FR, PPP
Physical layer
84.1. 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
94.2. 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
10Internet 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
11 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
12VoIP Call Processing and Protocol Flows
13B. 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)
14VoIP 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
15VoIP components
Gatekeeper
Intranet/ Internet (IP Network)
VoIP Terminals
Gatekeeper
Router
Router
VoIP Terminals
Gateway (Voice IWU)
Gateway (Voice IWU)
PSTN/ ISDN
ATM
PBX
16C. 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.
17VoIP 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
18H.323 call setup process
19VoIP 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.
202. 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.
21SIP Proxy operation
22SIP Redirect Server
23VoIP 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
24How MGCP coordinates the Media Gateways
25Megaco/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)
26Real-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.
27Real-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.
28Which 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.
29Which 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.
30Which 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.
31Details of signaling protocols
32D. 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.
33VoIP 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.
34VoIP 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)
35- 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.
36Business 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.
37TeleGeography VoIP market predictions for 2005
- In 2005 the international VoIP traffic will
exceed 40 billion minutes with more than 30 - annual growth.
38Roadblocks 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.
394.4. 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
40A. 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
41Nines of availability and corresponding downtime
42Reliability 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
43Reasons 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.
44Why 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
45Common 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.
46Benefits 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
47Commonly 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
48B. 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
50- 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).
51Availability (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
52Delay
- 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
53Delays
- 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.
55Sources 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).
57QoS 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
58QoS 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
59Summary 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
60Internet performance measurements RTT (from
Belgium to a specific region)
1200
RTT round-trip time
Source Alcatel
61Internet performance measurements
Source NetIQ Corp.
One-way delay receiver timestamp sender
timestamp
62Delays 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
63Internet performance measurements Packet Loss
(from Belgium to a specific region)
30
Source Alcatel
64C. 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?
65State 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
66Key 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
67D. 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
681. 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.
691. 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
70Reservation 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
72Integrated 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
73IntServ 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
74IntServ/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
76Differentiated 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)
77Differentiated 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
78Differentiated 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
804.5. 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
81B. 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)
82This 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.
84R-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.
85R-factor values from the E-model and
corresponding MOS values (Cntd)
86R-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.
87Calculating 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).
88C. 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
89Parameters 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
91Common 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
92Codecs 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
94List 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
95List 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
96Use 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)
97QoS - 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
98TeleGeography VoIP market predictions for 2005
- In 2005 the international VoIP traffic will
exceed 40 billion minutes with more than 30 - annual growth.
99Convergence 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
-
-
101Codec comparison