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  • 4

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  • 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
  • 2. 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 PBX LE LE PBX PBX PSTN 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 – it’s not critical for data, but critical for other possible apps
  • 6. Data network Server Router Router App server Branch office HQ office Res. house Modem/router Public/private network
  • 7. Ethernet, ATM, FR, PPP Physical layer TCP UDP HTTP, FTP H.323, SIP, RTP, RSVP, MGCP, MEGACO/H.248 Web Browser, MS Outlook, LOTUS IP
  • 8. 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
  • 9. 4.2. PSTN/Internet convergence for data services: Narrowband Internet access (local area) LEX LEX (local area) (local area) LEX LEX ISP LEX Access PoP LEX LEX LEX Access PoP Access PoP Central PoP PSTN trunk (ISDN PRI) trunk (SS7) LEX LEX - Local Exchange PoP – Point-of-Presence ISP – Internet Service Provider
  • 10. Internet access methods Home Network Intermediate Network modem bank/ access server/ router Access Devices ISP X ATM/FR/LL ISP PoP Corporate PoP access server / router POTS ISDN xDSL cable modem X CATV FR - Frame Relay LL – Leased line Virtual PoP (VPOP) FR/ATM/LL POTS/ISDN Narrowband dial-in access Narrowband dial-in access with virtual POP Broadband access Corporate leased line access ISP backbone PSTN PSTN X Broadband access
  • 11. 4.3. PSTN/Internet convergence for voice services A. Converged network Gateway Gateway LAN PC LAN LAN LAN Modem/Router Router Server Router App server Res. house PBX Branch office HQ office IP-based public/private network
  • 12. VoIP Call Processing and Protocol Flows
  • 13. B. Network scenarios for VoIP PSTN/ISDN Gatekeeper Call Processing Name’s Server OAM Server RAS POP PSTN/ISDN Internet 64 kbit/s speech Voice over IP Message interface to central server Voice Voice Voice IWU (Gateway) RAS POP Voice IWU (Gateway) S 0 u r c e Destination Registration, Admission, and Status Protocol (RAS) PC to Phone Phone to Phone Phone to PC PC to PC
  • 14. 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 Intranet/ Internet (IP Network) Router Gatekeeper VoIP Terminals Router Gateway (Voice IWU) PSTN/ ISDN ATM PBX VoIP Terminals Gatekeeper Gateway (Voice IWU)
  • 16. 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.
  • 17. 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
  • 18. H . 323 call setup process
  • 19. 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.
  • 20. 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.
  • 21. SIP Proxy operation
  • 22. SIP Redirect Server
  • 23. 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
  • 24. How MGCP coordinates the Media Gateways
  • 25. 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)
  • 26. 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.
  • 27. 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.
  • 28. 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.
  • 29. 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.
  • 30. 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.
  • 31. Details of signaling protocols
  • 32. D. VoIP scenarios: Phone-to-Phone PSTN/ISDN VoIP Server (Gatekeeper) RAS POP PSTN/ISDN Internet Voice Voice Voice IWU (Gateway A) RAS POP Voice IWU (Gateway B)
    • Basic Call "Phone-to-Phone"
      • A-Subscriber dials IWU E.164 number
      • Normal Call Setup (a) between A-Subscriber and A-IWU
      • Announcement from A-IWU to user
      • Input of A-Subscriber E.164 Number, PIN and B-Subscriber E.164 Number (via multi-frequency code)
      • ( SP) Call setup (b) within the Internet between A-IWU and B-IWU (routing functions are in gatekeeper)
      • Normal Call Setup (a) between B-IWU and B-Subscriber.
    A B A B MGCP (a) (b) (a)
  • 33. VoIP scenarios: PC-to-Phone PSTN/ISDN VoIP Server (Gatekeeper) RAS POP PSTN/ISDN Internet Voice Voice Voice IWU (Gateway) RAS POP Voice IWU (Gateway)
    • Basic Call "PC-to-Phone"
      • PC needs VoIP software (support on of Signaling Protocols)
      • Normal Internet login (a) of A-Subscriber
      • Access to VoIP Server
      • Input PIN and B-Subscriber E.164 Number
      • ( SP) Call setup (b) within the Internet between A-subscriber and B-IWU (routing functions are in gatekeeper)
      • Normal Call Setup (a) between B-IWU and B-Subscriber.
    (b) (b) (a) (a) A B A B
  • 34. VoIP scenarios: Phone-to-PC PSTN/ISDN VoIP Server (Gatekeeper) RAS POP PSTN/ISDN Internet Voice Voice Voice IWU (Gateway) RAS POP Voice IWU (Gateway)
    • Basic Call "Phone to PC"
      • PC needs VoIP software (support on of Signaling Protocols)
      • Normal Internet login (a) of B-Subscriber and registration at gatekeeper (E.164 to IP address mapping)
      • A-Subscriber dials IWU E.164 number
      • Normal Call Setup (a) between A-Subscriber and A-IWU
      • Input of A-Subscriber E.164 Number, PIN and B-Subscriber E.164 Number
      • ( SP) call setup (b) within the Internet between A-IWU and B-subscriber PC (routing functions and address mapping are in gatekeeper)
    MGCP (a) (a) (b) (b) A B A B
  • 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.
  • 36.
    • Business VoIP service is defined as a high quality, reliable service capable of
    • sustaining mission-critical communications. High quality is defined as clear audio with
    • the absence of echo. A reliable service connection provides an error free transmission
    • with no service interruptions.
    • IP-Telephony uses IP as the transport mechanism but it uses the public data
    • network (i.e., the Internet) to transmit voice packets. Because the Internet is an
    • unmanaged, non-voice engineered conglomerate of many networks, it cannot
    • guarantee bandwidth and timely delivery of voice packets, resulting in unacceptable
    • voice quality for business communications.
    • By transmitting voice over a private managed IP data network, you can control all of
    • the network characteristics required to ensure high-quality, reliable voice
    • communications over a data network.
    Business VoIP and IP-T
  • 37. TeleGeography VoIP market predictions for 2005
    • In 2005 the international VoIP traffic will exceed 40 billion minutes with more than 30%
    • annual growth.
  • 38. 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.
  • 39. 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
  • 40. 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
  • 41. Nines of availability and corresponding downtime
  • 42. Reliability is a fundamental philosophy Source: Infonetics Research, November 2001 The Tier 1 Service Provider Opportunity, US/Canada 2001 0% 25% 50% 75% 100% Product Reliability Best Price-to-Performance Ratio Financial Stability Leading-Edge Technology Manufacturer’s Products Already Installed Pre-and post-sales service and support Manufacturer reputation Manufacturer’s future product offering Leasing and Financing Options Lowest Price Sales and Marketing Services Network Integration and Design Services 100 82 73 73 64 64 45 45 27 27 18 9 Percent of Respondents Rating 6 to 7 Manufacturer Selection Criteria (Q61, n-11) Source: Contingency Planning Research, a division of Eagle Rock Alliance Ltd Reliability moved up the value scale and now rates highest for Tier_1 Service Providers
  • 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.
  • 44. Why are traditional IP Routers Unreliable? 7% Customer Premises Equipment
    • 36% Router Operations
      • Software/hardware updates
      • Configuration errors
    • 21% Router Failures
      • Hardware fault intolerance
      • Software quality
    Physical Links 27%
    • Congestion 5%
      • Network Engineering
    Malicious 2% Unknown 2% Source: University of Michigan
    • MPLS traffic engineering
    • Diversity of paths
    • Fast Restoration
    • Software process isolation and redundancy
    • 99.999 percent available hardware
    • Software upgrades
    • Hardware upgrades
  • 45. 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%.
  • 46. 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
  • 47. 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
  • 48. B. QoS parameters - system performance metrics QoS Applications Interactive TV Streaming media E-mail, file transfer Voice Web browsing
    • Bandwidth (Network Throughput )
    • Network/Devices Availability
    • Packet Delay
    • Packet Delay Variation
    • - Jitter
    • Packet Loss
  • 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).
  • 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. 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).
  • 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.
    • 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
  • 58.
    • Packet loss
      • E2E packet loss for voice should be < 2%
      • E2E 64k transparent should be more stringent < x %
      • ETSI TIPHON (voice)
        • <0.5% class 1 = gold
        • 0.5% to 1% class 2 = silver
        • 1% to 2% class 3 = bronze
      • Provided the E2E delay < 150 ms all above classes are acceptable
    QoS: Voice transport requirements (Cntd)
  • 59. Summary of network QoS requirements Optimal network QoS parameters Limits of network QoS parameters Delay – one way <= 100ms Delay – one way <= 150ms Jitter <= 40ms Jitter <= 75ms Packet loss <= 1% Packet loss <= 3%
  • 60. Internet performance measurements: RTT (from Belgium to a specific region) 1200 Source: Alcatel RTT – round-trip time Europe North- America South- America Asia Oceania Africa Middle- East 1000 800 600 400 200 0 RTT (ms) 1998 353.3 417.3 882.6 841.3 738.8 808.4 1270.6 Sept-Oct 1998 2001 204.4 219.7 509.6 461.8 441.0 521.4 620.9 Mar-Apr 2001
  • 61. Internet performance measurements Source: NetIQ Corp. One-way delay = receiver timestamp – sender timestamp
  • 62. Distance 10 100 1000 10.000 100.000 km STR – Stratosphere balloon LEO – Low-orbit satellite MEO – Middle-orbit satellite GEO – Geostationary-orbit satellite Delays for different satellite communications systems
  • 63. Internet performance measurements: Packet Loss (from Belgium to a specific region) 30% Source: Alcatel Europe North- America South- America Asia Oceania Africa Middle- East 25% 20% 15% 10% 5% 0% Packet Loss (%) 1998 11.2% 15.3% 17.0% 26.6% 12.6% 14.4% 23.4% Sept-Oct 1998 2001 3.7% 2.4% 5.8% 12.1% 3.0% 10.1% 10.2% Mar-Apr 2001
  • 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
        • 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
  • 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 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
  • 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:  
    • @ 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.
  • 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
    • Cl ass-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
    • 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
  • 80.
    • 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
    4.5. Estimation of call quality
  • 81.
    • Quality goal for a VoIP call the PSTN level of quality (“toll” quality)
    • But what is in IP networks???
    • We need to understand some of the different measurement standards for voice quality
    • The leading subjective measurement of voice quality - Mean Opinion Score (MOS) – Recommendation ITU P.800 – but for telephone equipment!
    • The Mean Opinion Score (MOS) described in ITU P.800 is a subjective measurement of call quality as perceived by the receiver. A MOS can range from 5 down to 1, using the following rating scale (see Table)
    B. Standards for measuring call quality
  • 82. The MOS is measured on a scale from 5 down to 1 This mapping between audio performance characteristics and a quality score makes the MOS ( Mean Opinion Score) standard valuable for network assessments, benchmarking, tuning, and monitoring
  • 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 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.
  • 84. E-model R - factor values from the E-model and corresponding MOS values The R value, the output from the E-model, ranges from 100 down to 0, where 100 is excellent and 0 is poor. The calculation of an R value starts with the undistorted signal.
  • 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.
    • R = R 0
    • R = R 0 – I s – I d – I e + A
    • where:
    • I s : channel’s noise impairments to the signal
    • I d : delays introduced from end to end
    • I e : impairments introduced by the equipment, including packet loss
    • A : advantage factor (for example, mobile users may tolerate lower quality because of the convenience).
    Calculating an R value
  • 88. 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
  • 89. Parameters of VoIP codecs m a
    • MOS and R value include Pack delay and Jitter buffer delay
    • Common bandwidth – real bandwidth consumption:
    • # Payload = 20 bytes/p (40 bytes/s)
    • # Overhead includes 40 bytes of RTP header (20 IP + 8 UDP + 12 RTP )
    • G.723.1 – Quality is“Acceptable” only
  • 90. a m 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. m a m a
    • Codecs’ comparison (Cntd)
    • Any lost datagram impairs the quality of the audio signal. Data loss is thus a key call-quality impairment factor in calculating the MOS.
    • Random loss –simplest loss model
    • # One datagram is lost or two datagrams are lost time by time
    • # Small effect inside of delay limit (<=150 ms)
    • Bursts of loss
    • # Quality degrades most significantly
    • # More than two consecutive datagrams are lost
    • 5% random packet loss (upper Figure)
    • # MOS starts at around 4 for the G.711 codec
    • 5% bursty packet loss (Figure below)
    • # MOS starts at around 3.5 for the same codec
    • The effect of bursty loss is even greater on the other codecs
    Codec R-factor MOS G.711 93.2 4.4 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 <= 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 s ize
    • # Decreasing the bandwidth consumption
    • # BUT - it adds latency.
  • 96. Use call admission to protect against too many concurrent calls # Call Admission Control Use priority scheduling for voice traffic # DiffServ (EF) # Queuing mechanisms - giving VoIP higher priority Get your data network ready for VoIP # In general, unsatisfactory data networks # Network should be fully upgraded and tuned, before starting a VoIP deployment List of VoIP network design tips (Cntd)
  • 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 (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
  • 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