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
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
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
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
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
In general, H.323 was too broad standard without sufficient
efficiency. It also does not guarantee business voice quality
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.
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
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) , 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.
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.
Reliability in PSTN networks is already for 10s of years equal to the famous 99.999%, also called the 5 nines
Operators are so used to this reliability that they take it for granted
Why is it so important?
99% means downtime of 3.7 days per year
99.9% means downtime of 9 hours per year
99.99% means downtime of 53 minutes per year
99.999% means downtime of 5.5 minutes per year
Traditional IP data equipment does not offer 5 nines reliability
Nines of availability and corresponding downtime
Reliability is a fundamental philosophy Source: Infonetics Research, November 2001 The Tier 1 Service Provider Opportunity, US/Canada 2001 0% 25% 50% 75% 100% Product Reliability Best Price-to-Performance Ratio Financial Stability Leading-Edge Technology Manufacturer’s 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
Reasons for system unavailability Source: Gartner Group
User Errors and Process: Change management, process inconsistency
• 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
36% Router Operations
21% Router Failures
Hardware fault intolerance
Physical Links 27%
Malicious 2% Unknown 2% Source: University of Michigan
MPLS traffic engineering
Diversity of paths
Software process isolation and redundancy
99.999 percent available hardware
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
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 QoS Applications Interactive TV Streaming media E-mail, file transfer Voice Web browsing
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:
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
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.
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).
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).
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
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
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
# 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.
# 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.
# Target - a control of a packet delay and jitter and elimination of
# Based on priority level and type of traffic
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
Quality goal for a VoIP call the PSTN level of quality (“toll” quality)
But what is in IP networks???
We need to understand some of the different measurement standards for voice quality
The leading subjective measurement of voice quality - Mean Opinion Score (MOS) – Recommendation ITU P.800 – but for telephone equipment!
The Mean Opinion Score (MOS) described in ITU P.800 is a subjective measurement of call quality as perceived by the receiver. A MOS can range from 5 down to 1, using the following rating scale (see Table)
B. Standards for measuring call quality
The MOS is measured on a scale from 5 down to 1 This mapping between audio performance characteristics and a quality score makes the MOS ( Mean Opinion Score) standard valuable for network assessments, benchmarking, tuning, and monitoring
MOS and other methods are based in older telephony approaches. These approaches are not very well suited to assessing call quality on a data network, as they can’t take into account the network issues of delay, jitter, and packet loss.
Models don’t take into account E2E delay between the telephone speaker and listener. Excessive delay adversely affects MOS.
• Models show quality in one direction at a time.
• Models don’t scale to let you see the effect of multiple, simultaneous calls.
Recommendation ITU G.107 introduced the E-model. The E-model is better suited for use in data network call quality assessment because it takes into account impairments specific to data networks.
The output of an E-model calculation is a single scalar, called an “R-value” or R-factor derived from delays and equipment impairment factors. Once an R value is obtained, it can be mapped to an estimated MOS.
E-model R - factor values from the E-model and corresponding MOS values The R value, the output from the E-model, ranges from 100 down to 0, where 100 is excellent and 0 is poor. The calculation of an R value starts with the undistorted signal.
R - factor values from the E-model and corresponding MOS values (Cntd)
R - factor values from the E-model and corresponding MOS values (Cntd) MOS
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
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.
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)