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شبکه‌های بی‌سیم (628-40) مدل‌های کاربرد و معیارهای عملکرد

شبکه‌های بی‌سیم (628-40) مدل‌های کاربرد و معیارهای عملکرد. نیمسال دوّم 98-97 افشین همّت یا ر. دانشکده مهندسی کامپیوتر. Introduction. Last chapter : Understanding of the issues and techniques involved in carrying bit streams over wireless channels.

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شبکه‌های بی‌سیم (628-40) مدل‌های کاربرد و معیارهای عملکرد

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  1. شبکه‌های بی‌سیم (628-40)مدل‌های کاربرد و معیارهای عملکرد نیمسال دوّم 98-97 افشین همّتیار دانشکده مهندسی کامپیوتر

  2. Introduction • Last chapter: Understanding of the issues and techniques involved in carrying bit streams over wireless channels. • The resources required to carry a bit stream depend on: • The characteristics of the stream (e.g., the average rate, peak rate, and rate variability) • The performance required by the application generating the bit stream (e.g., an interactive voice call requires end-to-end delay bounds, but can tolerate some data loss) • This chapter: Discuss the major types of applications that telecommunication networks are used for and the performance issues related to these applications, particularly when wireless networks are involved. 2

  3. Network Architecture • CO: Central Office • TX: Trunk eXchange • GW: Gate-Way • R: Router Simplifiedview of telecommunication networks

  4. Components (1) Packet Switching Cellular Networks Circuit Switching • The Public Switched Telephone Network (PSTN) has carried telephone calls for nearly a century. • PSTN: calls are multiplexed onto the links by using circuit switching. • Packet switched public data networks have evolved from the early X.25 networks to the present ubiquitous Internet. • Cellular networks have provided mobile access since the early 1980s. 4

  5. Components (2) • In campuses and enterprises, mobile devices such as laptops and Personal Digital Assistants (PDAs) obtain access to Internet services via Wireless Local Area Networks (WLANs). • Wireless Metropolitan Area Networks (WMANs)and Ad-hoc Multi-hop Wireless Mesh Networks are also another types of wireless networks.

  6. Components (3) • GateWays (GW), interconnecting the PSTN, the Internet, and cellular networks. • Image in slide 3 shows only bearer gateways that are in the path of the actual application traffic. • Signaling gateways are also needed because signaling protocols are different in networks. • For example, a signaling protocol called Session Initiation Protocol (SIP) is used to set up voice calls in the Internet, whereas the Signaling System No. 7 (SS #7) is used in the PSTN. 6

  7. Components (4) Packet Radio Network (PRN) • Ad-hoc Multi-hop Wireless Mesh Network may be attached to the Internet. • Although multi-hop mobile wireless networks have been studied for more than three decades, in the early years under the name Packet Radio Networks, the deployments of such networks are still experimental. • The IEEE 802.16 suite of protocols (popularly known as WiMax) now contains the definition of a mesh networking standard. • Under this standard, nodes that cannot directly access a WiMaxBase Station (BS)can form a static mesh network that is connected at some point to the WiMaxBS.

  8. Point-to-Point Communication (1) • One instance is a voice call between a fixed line telephone instrument on the PSTN (e.g., A) and a Mobile Station (MS) such as a cell phone, or PDA Station (STA) or Subscriber Station (SS). • One of the functions of the GW is converting the Constant Bit Rate (CBR) flow of voice bytes in the PSTN to a lower bit rate voice coding scheme over the resource limited cellular wireless network. 8

  9. Point-to-Point Communication (2) GSM multiple access scheme: FDMA + TDMA • In cellular networks, typically, voice is handled as a CBR stream of a lower bit rate than that in the PSTN. • In an FDM-TDMA system, such as GSM, there are channels, each of which can carry one call at a fixed rate. • An accepted call is assigned to a channel for the entire duration of the call needing circuit multiplexing system. • A call for which no free channel is available is blocked and the main performance measure for the system is the Blocking Probability (Pb). B(t): The number of calls blocked in the same time A(t): The number of call arrivals until time t uplink spectrum 9

  10. Point-to-Point Communication (3) • In a CDMA cellular system, the voice connection is handled by assigning to it the required coded rate. • Unlike typical FDM-TDMA systems, a variety of rates can be assigned. • Whether or not a call is accepted, depends on: • The rate requirements of the other calls that have already been accepted. • The resulting interference levels once the new call is accepted. • Each accepted call needs to be assigned a transmit power level. • Again, the performance measure of interest is the Blocking Probability. 10

  11. Point-to-Point Communication (4) • Another instance is a voice call between the PSTN instrument A and a Voice Over IP (VoIP) endpoint B. • The GW between the PSTN and the Internet converts the CBR flow of voice bytes in the PSTN to an asynchronous flow of voice packets in the Internet. • Packets flow asynchronously in the packet network, and can be queued in buffers in the network routers. 11

  12. Point-to-Point Communication (5) • There could be a voice call between either A or B and the endpoint D, which accesses the Internet via a WLAN or a WMAN. • The packet multiple access mechanism over the wireless access network affects the performance of the voice call. 12

  13. Point-to-Point Communication (6) Three-Way Handshake (TCP) • As another instance, the MS C being used to browse the contents of the web server E which is attached to the Internet. • This kind of application is quite different from voice, as there is no intrinsic rate at which the data should be transferred from the web server to the mobile phone. • Feedback-based rate control algorithms are employed to ensure some sort of rate fairness between such connections, and efficient utilization of network resources. • In the Internet: TCP is in conjunction with implicit or explicit congestion feedback from the network. 13

  14. Point-to-Point Communication (7) • The MS C or the device D could be displaying a video that is stored in the server E. • Once the video starts playing, the network should provide this connection the average rate required to transport the video stream. • Variability in the rate at which the network transports the video can be compensated by buffering a sufficient amount of the video in the playout device. • Such buffering must be done in such a way that the playout does not starve (i.e., the buffer empties out), nor does the buffer overflow. 14

  15. Types of Traffic • Applications generate one of the following types of traffic resulting from previous scenarios: • Elastic traffic: WWW browsing, FTP file transfers, and electronic mail • Real-time stream traffic: packet voice telephony • Store-and-forward stream traffic: streaming movies or music over the Internet 15

  16. Elastic Traffic (1) Elastic WLAN system topology Overview of Elastic WLAN • Although the human (or some machine application,) that wishes to achieve file transfer, would like to have the transfer completed as soon as possible, the source of data itself does not demand any specific transfer rate. • If the data transfer does not lose data, no matter how fast or slow it is, the file will sooner or later get transferred to the destination device. • Elastic traffic does not have an intrinsic temporal behavior, and can be transported at arbitrary transfer rates. 16

  17. Elastic Traffic (2) QoS Requirements: • Transfer delay and delay variability can be tolerated. • An elastic transfer can be performed over a wide range of transfer rates, and the rate can even vary over the duration of the transfer. • The application cannot tolerate data loss. • This does not mean, however, that the network cannot lose any data. • Packets can be lost in the network (owing to uncorrectable transmission errors or buffer overflows,) provided that the lost packets are recovered by an automatic retransmission procedure. • So effectively the application would see a lossless transport service. • Since elastic sources do not require delay guarantees, the delay involved in recovering lost packets can be tolerated. 17

  18. Elastic Traffic (3) • In practice, users will not tolerate arbitrarily poor throughput, high throughput variability, and large delays. • So a network carrying elastic traffic will need to manage its resource-sharing mechanisms in a way such that some minimum level of throughput is provided. • Also some sort of fairness must be ensured between the ongoing elastic transfers. • Elastic traffic can be carried over circuit multiplexed networks (e.g., the PSTN or GSM cellular networks), or over networks that allocate a fixed rate to the elastic connection (e.g., a second generation CDMA cellular network). • Shaping of the traffic so as to match the allocated rate should be carried out by the source. 18

  19. Real-Time Stream Traffic (1) • Digitized speech emanating from an end-device involved in interactive telephony could be: • A periodic stream of bytes or packets. • An on-off stream of bytes or packets, if silence suppression is employed. • This source of traffic has an intrinsic temporal behavior, and this pattern needs to be preserved for faithful reproduction of the speech at the receiver. • The network will introduce delay: • Fixed propagation delay • Queuing delay that can vary from packet to packet in packet networks. 19

  20. Real-Time Stream Traffic (2) • Playoutdelay introduced at the receiver (to mitigate the effect of random packet delay variation,) will be larger, when the packet delay is more variable. • The network cannot serve such a source at arbitrary rates, as it could in the case of elastic traffic. • Real time interactive speech or video telephony are examples of real-time stream traffic. 20

  21. Real-Time Stream Traffic (3) QoS Requirements: • Delay (average and variation) needs to be controlled. • Real-time interactive traffic such as that from packet telephony would require tight control of source-to-sink delay. • For wide area packet telephony, the delay may need to be controlled to less than 200 ms, with a probability more than 0.99. • Packets that do not conform to the delay bound are considered to be lost. • There is tolerance to data loss. • From the point of view of the receiver, packets can be lost for two reasons: • Buffer overflows, or unrecovered link losses in the network. • Late arrivals at the receiver. • Owing to the high levels of redundancy in speech and images, a certain amount of data loss is imperceptible. • Stream traffic expects the intrinsic loss rate from the packet transport service to be bounded. 21

  22. Store-and-Forward Stream Traffic (1) • Applications such as streaming audio and video basically involve a one-way transfer of an audio or video file stored on the disk of a media server. • In order for the received video to be useful, the playout device should be continuously fed with video frames. This can be achieved by: • Providing a guaranteed transfer rate, as would be done, for example, in a CDMA cellular system. • A more economical way is to treat the transfer as elastic, and buffer the video frames as they are received. • This would be the approach taken when the video is transferred over the random access WLAN. 22

  23. Store-and-Forward Stream Traffic (2) • Playout is initiated only after a sufficient number of video frames has been buffered, so that a smooth video playout can be achieved in spite of a variable transfer rate across the contention-based WLAN. QoS Requirements: • The average transfer rate provided in the network should match the average rate at which the stored data has been encoded. • The transfer rate variability should not be too large. 23

  24. Store-and-Forward Stream Traffic (3) • Store-and-forward stream traffic is like real-time stream traffic since it has anintrinsic average rateat which it must be transported, but it does nothave strict delay bounds, and hence the network can provide it a time varying transfer rate. • TCP can be used to transport store-and-forward streaming media, provided the average TCP throughput does not drop below the average coded rate of the media. • The added benefit of TCP is that, it recovers lost packets. 24

  25. Closed and Open Loop Traffic • Refer to real-time stream traffic as open loop, as it has an intrinsic temporal behavior: • The rate of flow on a connection is determined by the application, and these sources are not controlled by the network. • A limited amount of controllability is possible in some systems, by the sink alerting the source of poor playoutquality, to which the source can respond by using a lower bit rate coder. • Closed loop controls invariably are used when transporting elastic traffic, so such traffic can be called closed loop. • The source of the traffic is made to continually adjust its rate of emitting data by means of implicit feedback (packet loss) or explicit feedback (control bits in packet headers). 25

  26. Real-Time Stream Sessions • Traffic modeling and QoS issues for real-time stream sessions (Guaranteed delay) in the context of voice telephony are: • Constant Bit Rate (CBR) Speech • Variable Bit Rate (VBR) Speech • Speech playout • Quality of Service (QoS) Objectives • Network Service Models 26

  27. CBR Speech (1) PCM • Consider a voice call between a pair of endpoints: • Electrical signals from a microphone are digitized and coded by a speech codec in each end device. • PCM (Pulse Code Modulation) • Sample the analog signal from the microphone at a rate of 8000 samples per second. • Quantize the resulting continuous amplitude samples into 256 predetermined levels. • Encode each of these levels into 8 bits (one byte). • The PCM encoder (ITU G.711), yields a CBR source that produces 1 byte every 125 μsec, equals to 64Kbps. voltage 7 6 5 4 3 2 1 0 levels Binary Codes voltage 27

  28. CBR Speech (2) • A PCM source can be compressed to yield CBR sources at various rates. • ITU G.729 Vocoder • Takes PCM as input and produces 10 bytes of coded speech every 10 ms, thus yielding a coded bit rate of 8Kbps. • However this coder has a coding delay of 15 msand a decoding delay of 7.5 ms. • Thus the minimum delay between a sound being produced at the source device and this being heard at the other end is 22.5 ms. 28

  29. CBR Speech (3) • It is necessary for the network to use a service rate greater than or equal to the voice bit rate in order to carry a CBR voice source. • Although for CBR sources it is sufficient to allocate the CBR rate, but If the allocated service rate is exactly same as the source rate, there will not be any queuing. • Considering a voice call between the PSTN phone A and the cell phone C: • GW converts PCM speech arriving over the PSTN to CBR speech at rate R. • So cellular network can just allocate resources so that the voice call is provided a service rate of R. 29

  30. VBR Speech (1) • In speech generated byinteractive telephony: • There are low energy periods that correspond to silences while the speaker listens, or to gaps between words, sentences, and utterances. • Coder output corresponding to these inactive periodscan be discarded or encoded at alower rate. • This yields a Variable Bit Rate (VBR) coded speech. 30

  31. VBR Speech (2) • VBR speech can be handled: • as a variable rate bit stream, • or can be packetized for transport over a packet network. • Take a certain number of bytes from the source (e.g., 160 bytes or 20 ms of speech from a PCM source,) and generate a packet from these. • Short packet is generated when a talk-spurt finishes before 160 bytes have been collected. 31

  32. VBR Speech (3) • Bytes that arrive early in the packet have to wait for those that arrive later, until the packet is formed. So packetizer must wait to accumulate a packet. • This results in a packetization delay, which can be reduced by using shorter packets. • Packets cannot be very short, otherwise significant amount of header overhead will be in each packet, (e.g. in the Internet, at least 12 bytes for RTP (Real-time Transport Protocol), 8 bytes for UDP (User Datagram Protocol), and 20 bytes for IP). 32

  33. VBR Speech (4) • Although the inactive periods do not have speech information in them, the duration of the gaps is indeed information that needs to be conveyed to the receiver. • One of the difficulties in the transport of packetized VBR speech is the retention of the mentioned timing information. • The inactive periods can only be approximately replicated at the receiver. • The resulting errors are not noticeable if the inactive periods are long. 33

  34. VBR Speech (5) • Suppose the VBR speech source is allocated the service rate C in the cellular network. • We denote by R,the peak rate of the VBR source, and by rav , the average rate. • If the on-off VBR source has an average on-duration of 400ms and average off-duration of 600ms, and R= 64Kbps, then we have: rav=(400/400+600) x R = 25.6Kbps • It is waste of bandwidth to make C> R,but it is necessary that C≥ rav. buffer fills when source rate > service rate 34

  35. VBR Speech (6) • When the voice source is emitting data at the rate Rp, then the link buffer builds up at the rate of (R-C) Kbps. • But we don not know how long this rate mismatch will last. • Hence, if we want to bound the delay of the voice bytes in the link buffer, in the absence of any other information about the source, our only recourse is to use C = R. • This approach ofpeak rate allocation is typically adopted in FDM-TDMA and CDMA cellular systems. • In CDMA systems, even though the peak rate is allocated to a call, the on-off nature of VBR speech is exploited, because during the voice silence periods a call does not cause interference. 35

  36. Speech Playout (1) • Consider VBR coded packetized speech between the device B and D, or B and C. • At devices C and D there is the problem of playing out the individual packets. • The original voice pattern should be reproduced, in spite of the random network delays introduced by the packet network. • Suppose that the cellular network handles only CBR speech; then the gateway, GW, between the Internet and the cellular network should convert the asynchronously arriving speech packets, to CBR speech. 36

  37. Speech Playout (2) 37

  38. Speech Playout (3) • Immediate playout: Playout each packet as soon as it was received. Cause to frequent lost packets, and frequent interpolations. Thus would lead to very poor speech quality. • Deferred playout: A playout delay is applied to each packet to allow trailing packets to catch up. 1) Considering the maximum delay. • How we can know in advance the maximum delay? • The worst-case delay could be very large! 2) The packet network may have the ability to provide a delay guarantee at call setup. 3) The receiver needs to estimate the delay by means of time stamps, as the call progresses (no guarantee!). 38

  39. QoS Objectives (1) • Mouth-to-Ear (MtoE) delay, is delay between a sound being produced at the source device and this being heard at the other end. MtoEdelay = coding delay + packetization delay + network propagation delay + network transmission and queuing delay + receiver playout delay + decoding delay Expected:Pr ( MtoE delay > 200mS ) < 0.02 39

  40. QoS Objectives (2) • Coding delay, packetization delay, network propagation delay, and decoding delay are fixed. • Network transmission and queuing delay (X1), and receiver playout delay (X2) are variable. Expected: Pr ( Xi> Ti) < εi • Packet Loss is due to: • Buffer overflows in routers • Arriving packet after scheduled playouttime • Errors in the network Expected: Pr (Packet loss) < 0.05 40

  41. Network Service Models (1) • In access networks which accept calls based on peak rate allocation, the only issue is to design the network for a desired call blocking probability, and the Erlang blocking model can be used. • In access networks which assign a fixed rate less than the peak rate of a VBR call, then the buffering model can be used. 41

  42. Network Service Models (2) • In some access networks the service rate applied to a connection may not be constant. • For example, in OFDMA systems, the number of bytes to be served from a queue can vary from frame to frame depending on the fading in the various carriers, the competing traffic, and the power constraints. • In this case we have a dynamically controlled server, and detailed analysis to obtain buffer occupancy distributions or delay distributions is difficult. 42

  43. Network Service Models (3) • WLANs, based on the 802.11 standard, are contention-based systems. • Hence the service applied to a queue is time varying because the number of contenting nodes varies over time, as some queues empty out while others receive new traffic. • Some progress has been made on developing analytical models for the performance analysis of WLANs. 43

  44. Elastic Transfers (1) • Elastic traffic is generated by applications whose task is to move chunks of data between the disks of two computers connected to the network. • Elastic flows can be speeded up or slowed down depending on the number of flows contending for the capacity of the network. • The basic problem is to transfer each file entirely from the source to the destination machine. There is no intrinsic rate at which the files must be transferred, and the transfer rate can vary as a file is transferred. • Further, there is no intrinsic packet size that the files need to be segmented into during their transfer. The transfer protocol can view each file as a byte stream, and transfer varying amounts of it in each packet. 44

  45. Elastic Transfers (2) Several users downloads files form a server attached to a operator’s own high speed LAN. 45

  46. Dynamic Control of Bandwidth Sharing (1) • Consider a very simple network comprising a single link over which several users, on their respective hosts, are downloading files from some web servers. • The link capacity is limited to C bps, but the local networks attaching the users and the web servers to this link are infinitely fast. • Suppose, one user initiates file transfer with a throughput of C bps. • Then another user starts a session, while the first file transfer is still progressing. Read More Article: Dynamic Bandwidth Shaping Algorithm for Internet Traffic Sharing Environments. Site: https://pdfs.semanticscholar.org/c2db/0e943633d7d021a6212e9529d0dc61506b59.pdf 46

  47. Dynamic Control of Bandwidth Sharing (2) • There is two kinds of issue when 2 (or more) client are using network : • Congestion • Unfairness • Both of these issues require that there be some kind of feedback (explicit or implicit) to the data sources so that the rate of the first transfer is reduced, and that of the second transfer is increased, so that ultimately each transfer obtains a rate of C/2bps. • When the first session departs, the source of the second session should increase its transfer rate so that a throughput of C bps is obtained. 47

  48. Dynamic Control of Bandwidth Sharing (3) Explicit Feedback Scenario ImplicitFeedback Scenario • Explicit feedback: control packets flow between the traffic sources, sinks, and the network, and these packets carry information (e.g., an explicit rate or a rate reduction signal) that is used by the sources to adapt their sending rates. • Implicit feedback: can be provided by packet loss or increase in network delay; that is, a source can reduce its rate on sensing that one of the packets it sent may not have reached the destination. 48

  49. Control Mechanisms: MAC and TCP • In wideband cellular wireless networks, the entire system bandwidth is not used as one fat pipe as we discussed. • Instead, radio resource allocation is done on an MS by MS basis, depending on the channel conditions to the mobiles. • Thus, even at the medium access layer it is possible to implement control strategies that achieve some sort of a rate allocation objective over the MSssuch as: • Equal rate allocation • This is an example of a more general fairness objective called max-min fairness. • Maximize the total rate over the MSs • Might be very unfair as MSs with poor connectivity might obtain no throughput. 49

  50. Transmission Control Protocol (TCP) • Bandwidth sharing in the wide area Internet is controlled by the TCP that resides in all end-systems attached to the Internet, including Internet-enabled cellular phones. • Sits between the applications and IP, the Internet’s packet routing and forwarding protocol. • is connection oriented • Enhances the unreliable, non-sequential packet transport service provided by IP to a reliable and sequential packet transport service. • Uses a window-based packet loss recovery mechanism to achieve this function. Also this mechanism is employed for two other major functions that TCP provides: • sender-receiver flow control • adaptive bandwidth sharing in the network 50

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