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IP QOS Mechanisms. Prepared by Νικήτας Γιαννακός Last update: 12-05-2003. QoS Metrics QoS Introduction QoS Architecture RSVP Protocol IntServ, Integrated Services Model DiffServ, Differentiated Services Model. QoS Metrics. What means Quality of Service ?.
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IP QOS Mechanisms Prepared by Νικήτας Γιαννακός Last update: 12-05-2003
QoS Metrics QoS Introduction QoS Architecture RSVP Protocol IntServ, Integrated Services Model DiffServ, Differentiated Services Model
What means Quality of Service ? • QoS is the ability of a network to servicea given application efficiently, withoutaffecting its function or performance • Depending on the kind of network different metrics are defined
At circuit switched networks QoS is measured (defined) by: • Technical factors • Reliability • Expandability • Effectiveness • Maintainability of the system • Congestion waiting • Transmission quality ... • Human factors • Stability of service quality • Availability of subscriber lines • Waiting times • Fault clearance times • Subscriber information • Stability of operation of the system • ...
At packet switched networks QoS is measured (defined) by: • Availability • Bandwidth (Throughput) • Delay (Latency) • Delay Variation (Jitter) • Error Ratio (Packet Loss)
Quality of Service Parameters Analogy: • Availability • What portion of the time bank clerk is at his position • Bandwidth (Throughput) • How fast bank clerk is • Delay • Time that clients have to wait at queue • Delay Variation (Jitter) • Variation of time clients spent at queue • Error Ratio (Loss) • Percentage of clients that clerk cannot be served, rejected • Bank high QoS means: Clients served quickly, not waiting too much and they know from before their servicing time
Availability • A network service can never be 24/7 available • From service provider point of view : • For example a 99% availability means • 24hours/day x 365 days/year x 0.01 = 88 hours/years unavailability • From customer point of view:
Bandwidth (Throughput) • Bandwidth at IP networks is actually a promise: Sometimes applications get close to promised BW but never, never reach it • In other words BW is not guaranteed, and no one and all are responsible for that, it is because of the statistical nature of a packet switching network
Delay (Latency) • Time it takes for a packet to reach the targetdestination • ITU recommendation I.380 defines • IPTD, IP Packet Transfer Delay • The delay for a IP datagram between two reference points • Typically an end-to-end delay or a delay within one network • Mean IP Packet Transfer Delay • Arithmetic average of the IP packet transfers delays for packets we are interested about
Why we concern about Delay? • Some applications are intolerant to excessive delays • Voice apps • Too much delay will cause man on moon effect • ITU guidelines: • Delay < 150ms acceptable • 150ms < Delay < 400ms acceptable, depending on the voice quality desired • Delay > 400ms deemed unacceptable • Internet games • Too much delay will destroy interactivity
Why we concern about Delay? • Multimedia apps • For example a MPEG stream requires delay to be lower that the GOP, Group Of Pictures size, in order decoder to decode the GOP • Time sensitive apps • Applications such asSNA and Fax transmission can timeout (expire) • Or information can be useless (unusable) for the application after the timeout
Delay is comprised of : • Fixed Delay • Processing • Serialization • Quantization • Propagation • Variable Delay • Queuing • Routing • One-way delay : Time it takes on average for packet to traverse a particular network path • Round-trip time : One way delay plus the average time for a response to get back
Serialization Delay t1 Transmission media frame Serialization Delay Transmitter buffer t2 Next frame Transmission media frame
Serialization Delay • Time it takes to actually transmit the packet • A 1500-byte packet transmitted at 256 Kbps • Serialization Delay = 47 ms • It only takes 3us to transmit a 4000 bytes packet on a 100Mbps connection
Processing Delay • Coding/compression/decompression/decoding (i.e processing of a voice codec) • Packetization (i.e by TCP/IP protocol stack) • It is an application-based delay • Propagation (Distance) Delay • Based on transmission distance, transmission media characteristics • Approximately 6ms/Km
Queuing Delay • Packets need to be hold in a queue during a congestion on the physical link • Time a packet spends in router queues before it is serviced • Ingress queuing delay for traffic entering anetwork node • Egress queuing delay for traffic exiting anetwork node • •Dejitter buffers • Variance in queue depth caused by: • Variable packet sizes • Unloaded network Queuing delay is negligible • Heavily congested network Queuing delay becomes the main delay component • Routing / Switching (Forwarding) Delay • Time to take switching decision
Typical Delay Budgets at networks over the world • OK within regions, N. America OK with Europe, Japan
When Delay becomes a problem? • For LAN links, delay can be ignored due to short distances and high bandwidths • Low-speed dial-up and WAN connections introduce measurable latency • For earth-bound connections, the latency is mostly a function of the bandwidth • Satellite connections always involve significant delay • When networks are not congested (transmission capacity is greater than flow of data) , delay is very low • Typical delay in a not congested network can be 50ms, while in a congested one can be 400ms or more
56kb WAN Serialization Delay can be a Problem Voice Packet 60 bytes Every 20 ms Voice Packet 60 bytes Every >214 ms Voice Packet 60 bytes Every >214 ms ~214ms Serialization Delay Voice 1500 Data Bytes Voice Voice 1500 Data Bytes Voice Voice 1500 Data Bytes Voice 10mbps Ethernet 10mbps Ethernet
Serialization Delay • Depending on bit-rate it may be a significant portion of overall delay • ATM has the lowest serialization delay because priority cells need only wait for 53 bytes • Token Ring networks may have significant delays if large packet sizes are used • Rule of thumb : Serialization Delay is only significant in WAN links that are 768 Kbps or slower and are being used for real-time voice or video • In modern world where high bit-rate connections are becoming the norm serialization delay is becoming more and more irrelevant
When Delay becomes a problem? • In general • Propagation delay is the most important factor in very large distances like the satellite links • Serialization delay is the most important factor in case of large packets transmitted at low bitrates • Queuing delay is the most important contributor in most of the cases
Delay variation (jitter) • Difference in the arrivaltimes of packets
Why we concern about Jitter? • Jitter is induced by the network congestion severity • As congestion increases, delay increases too • As congestion appearsand disappears, depending on network load, delay increases and decreases • Jittering effect can be worst than delay • When delay varies, it is more difficult to design an algorithm that will compensate for that • A jitter buffer is used to smooth outarrival times, but there are instantaneous and total limitson the buffer’s ability to smooth out arrival time (too large buffer will increase delay too much)
Why we concern about Jitter? • Elastic applications (like TCP) • Small delay variations are not important • Large ones may cause packet retransmissions or long delays beforeretransmit • Real-time,delay-sensitive applications such as voice andvideo • A minimal amountof jitter may be acceptable but as jitterincreases, the application may become unusable • For example in IP telephony a packet arrived too late will be discarded, resulting in a small amount of distorted audio (Jitter forVoIP should be <30 ms)
Packet Loss • Refers to the percentage of packets dropped • ITU recommendation I.380 defines IPER, IP Packet Error Ratioand IPLR, IP Packet Loss Ratio • Spurious IP Packet Rate • Number of spurious packets observed in time interval • Measured in Packets per Service-Second
Packet Loss • Protocols like TCP deal with losses, asking retransmission and slowing of transmission rate • These protocols are unresponsive (continue to transmit even if network is congested, worsening things)
Why we concern about Packet Loss? • Non Real Time apps • Protocols like TCP deal with losses, asking retransmission and slowing of transmission rate • Real Time apps (no retransmission possible) • Light and fast protocols like UDP, used for real time transmissions (multimedia frames, encapsulated over RTP, Real Time Protocol and RTP over UDP) • These protocols are unresponsive (continue to transmit even if network is congested, worsening things) • For VoIP packets should be <1% • Male voice with no impairment (G.711) PSQM score = 0.00 • •Male voice after 3 percent packet loss (G.711) PSQM score = 2.39 • •Male voice after 10 percent packet loss (G.711) PSQM score = 3.15
Why we concern about Packet Loss? • Applications react on packet lossin different ways: • Fragile:If packet loss exceeds certain threshold, value of application is lost • Tolerant : Application can tolerate packet loss, but the higher the packet loss the lower is the value of application • Critical threshold levels are defined • Performance: Application can tolerate even very high packet loss ratio but its performance can be very low in high packet loss ratio
Why Packets get Lost? • Congestion • Errors introduced by thephysical transmission medium • To prevent congestionswitches discard packets (i.e Random Early Discard mechanism) • Most landline connections have very low lossas measured in the Bit Error Rate (BER) • Wireless connections (satellite,mobile or fixed wireless networks) have a highBER that varies due to environment orgeographical conditions • Transmission infrastructure becomes more and more reliable (optical fibre, redundant paths ..) so congestion becomes the important reason
Congestion Loss • Speed Mismatch • Aggregation • Confluence • Buffer exhaustion • Oversubscription
Speed Mismatch • The #1 Reason for Congestion! • Possibly Persistent when going from LAN to WAN • Usually Transient when going from LAN to LAN!
Aggregation • Transient Congestion fairly typical!
Confluence • Always need mechanisms to provide guarantees! • Transient Congestion occurs!
At packet switched networks • Default service in packet networks is the best effort service • Actually it means no effort at all, no guaranteed • Network gives the same service to all applications • Best effort service not satisfactory for many applications needs: • Video and audio conferencing : Bounded delay and loss rate • Video and audio streaming : Bounded packet loss rate • Time-critical applications (real-time control) : Bounded delays • Valuable applications : Better service than less valuable applications
Different Applications have Different requirements • In simple words applications want a specific degree of regular and robust receipt ofpackets • In other words network should provide bounded delay, jitter, packet loss to applications
Different Applications have Different requirements • To simplify things, traffic is grouped into classes that have similar QoS requirements
Apps have requirements but network has limited resources • There are parts of the network in which resources are unable to meet demand • Network devices (such as switches and routers) forward traffic among themselves using interfaces. If the rate at which traffic arrives at an interface exceeds the rate at which that interface can forward traffic to the next device, then congestion occurs
QoS Strategy ? • Different applications have different requirements of Quality metrics • Network shall be engineered in a such way such to satisfy these requirements Apps compete for network resources What we should do? Keep the existing Internet traffic model and provide more network resources Keep the network resources and apply QoS strategies (Means give plenty of resources to all - overshoot) (Means arbitrate) • A combination of both ways seems to be the right way
Overprovision Network • Over-provisioning both bandwidth and IP routing capacity • There is no need to apply a QoS strategy • Keep the Internet working with a best effort traffic model • Every packet is treated equally, transmission is Sent and Pray, network will try to do its best but no guarantee exist
Overprovision Network, Advantages • QoS is easy to achieve if the network over allocates resources for each flow • A very simple and efficient model • Assumption that bandwidth will be infinite • Optical fiber has enormous capacity • Most IP network service providers have yet to implement any type of QoS • Simple priority is sufficient, QoS is all about giving some traffic higher priority over other traffic • Most applications can adapt to even extreme delays
Overprovision Network, Disadvantages • Quite expensive approach • Open invitationfor each subscriber to send still more traffic, causingtheneed to upgrade continuously • This guides to profitless growth cycles • Meanwhile, predicting traffic volumes and assuring service levels • becomes ever more difficult • • Allocating bandwidth at the peak rate, yields deterministic • QoS, but is very inefficient use of network resources
or Apply QoS strategy • Build a network that can provide QoS with the least • amount of resources • Using better algorithms has the same effect as adding • bandwidth (or other resources) • QoS will allow to sensitive applicationssuch as IP telephony and videoconferencing to have a better that the “best effort” transmission