Outline Overview Streaming Multimedia Protocols for Realtime Conversational

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • LTE Quality of Service • Qo. S Measurements • Summary Reference: Jim Kurose, Keith Ross, Computer Networking: A Top Down Approach 7 th Edition, Pearson, 2016 2

Properties of Heterogeneous Wireless Networks • Benefit: expand the network capacity and coverage in a dynamic fashion • Challenge: Quality of Service (Qo. S) evaluation is very challenging due to the presence of different communication technologies – Different communication technologies have different characteristics • different performance measurement parameters – The applications that utilize them have unique Qo. S requirements • They may use different radio access networks but have the same Qo. S requirements 3

Qo. S • Three types of Qo. S: intrinsic, perceived and assessed – Intrinsic Qo. S is directly provided by the network itself and may be described in terms of objective parameters as, for instance, loss and delay – Perceived Qo. S (P-Qo. S) is the quality perceived by the users; • It is measured by the “average opinion” of the users by assigning Mean Opinion Score (MOS) rating (1 to 5) – Assessed Qo. S is the will of a user to keep on using a specific service • Most of the Qo. S provision is offered in terms of intrinsic (objective parameters) Qo. S by using a Service Level Specification (SLS) which is “a set of parameters and their values which together define the service offered to a traffic” [RFC 3206] 4

Qo. S Class Proposed by ITU-T Recommendation 5

Qo. S Class Proposed by ETSI-TR 102 6

Typical Qo. S Parameters • Latency / delay • Jitter • Bandwidth / Bit rate • Loss • Throughput / Admission rate 7

IP Qo. S Classes and Objective Performance-metric Upper Limits • • IPTD – IP Packet Transfer Delay IPDV – IP Packet Delay Variation (known as Jitter) IPLR – IP Packet Loss Ratio IPER – IP Packet Error Ratio 8

Example of Qo. S Service Level Specification 9

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • LTE Quality of Service • Qo. S Measurements • Summary 10

Multimedia Networking Applications • Audio – Analog audio signal sampled at constant rate • Telephone: 8, 000 samples/sec • CD music: 44, 100 samples/sec – Each sample quantized, i. e. , rounded • Each quantized value represented by bits, e. g. , 8 bits for 256 values • Video – Sequence of images displayed at constant rate – Digital image: array of pixels • Each pixel represented by bits – Coding: use redundancy within and between images to decrease bits used to encode image • Spatial (within image) • Temporal (from one image to next) 11

Multimedia: Audio and Video spatial coding example: instead of sending N values of same color (all purple), send only two values: color value (purple) and number of repeated values (N) ……………………. . . … audio signal amplitude quantization error quantized value of analog value analog signal frame i time sampling rate (N sample/sec) temporal coding example: instead of sending complete frame at i+1, send only differences from frame i+1 12

Multimedia Networking: 3 Application Types • Streaming, stored audio, video – Streaming: can begin playout before downloading entire file – Stored (at server): can transmit faster than audio/video will be rendered (implies storing/buffering at client) – e. g. , You. Tube, Netflix, Hulu • Conversational voice/video over IP – Interactive nature of human-to-human conversation limits delay tolerance – e. g. , Skype • Streaming live audio, video – e. g. , live sporting event (futbol) 13

Cumulative data Streaming Stored Video 1. video recorded (e. g. , 30 frames/sec) 2. video sent network delay (fixed in this example) 3. video received, played out at client (30 frames/sec) time streaming: at this time, client playing out early part of video, while server still sending later part of video 14

Challenges Of Streaming Stored Video • Continuous playout constraint: once client playout begins, playback must match original timing • Network delays are variable (jitter), so will need client-side buffer to match playout requirements • Other challenges: • Client interactivity: pause, fast-forward, rewind, jump through video • Video packets may be lost, retransmitted 15

Streaming Stored Video: Revisited constant bit rate video transmission client video reception variable network delay client playout delay constant bit rate video playout at client buffered video Cumulative data • Client-side buffering and playout delay: compensate for network-added delay, delay jitter time 16

Client-side Buffering, Playout buffer fill level, Q(t) playout rate, e. g. , CBR r variable fill rate, x(t) video server client application buffer, size B client • Initial fill of buffer until playout begins at tp • Playout begins at tp • Buffer fill level varies over time as fill rate x(t) varies and playout rate r is constant 17

Playout Buffering buffer fill level, Q(t) playout rate, e. g. , CBR r variable fill rate, x(t) video server client application buffer, size B client Average fill rate (x), playout rate (r): • x < r: buffer eventually empties (causing freezing of video playout until buffer again fills) • x > r: buffer will not empty, provided initial playout delay is large enough to absorb variability in x(t) – Initial playout delay tradeoff: buffer starvation less likely with larger delay, but larger delay until user begins watching 18

Streaming Multimedia: UDP • Server sends at rate appropriate for client – Often: send rate = encoding rate = constant rate – Transmission rate can be oblivious to congestion levels • Short playout delay (2 -5 seconds) to remove network jitter • Error recovery: application-level, time permitting • RTP [RFC 2326]: multimedia payload types • UDP may not go through firewalls 19

Streaming Multimedia: HTTP • Multimedia file retrieved via HTTP GET • Send at maximum possible rate under TCP variable rate, x(t) video file TCP send buffer server TCP receive buffer application playout buffer client • Fill rate fluctuates due to TCP congestion control, retransmissions (in-order delivery) • Larger playout delay: smooth TCP delivery rate • HTTP/TCP passes more easily through firewalls 20

Voice-over-IP (Vo. IP) • Vo. IP end-delay requirement: needed to maintain “conversational” aspect – – Higher delays noticeable, impair interactivity < 150 msec: good > 400 msec: bad Includes application-level (packetization, playout), network delays • Session initialization: how does callee advertise IP address, port number, encoding algorithms? • Value-added services: call forwarding, screening, recording • Emergency services: 119 21

Vo. IP Characteristics • Speaker’s audio: alternating talk spurts, silent periods – 64 kbps during talk spurt – pkts generated only during talk spurts – 20 msec chunks at 8 Kbytes/sec: 160 bytes of data • Application-layer header added to each chunk • Chunk+header encapsulated into UDP or TCP segment • Application sends segment into socket every 20 msec during talkspurt 22

Vo. IP: Packet Loss, Delay • Network loss: IP datagram lost due to network congestion (router buffer overflow) • Delay loss: IP datagram arrives too late for playout at receiver • Delays: processing, queueing in network; endsystem (sender, receiver) delays • Typical maximum tolerable delay: 400 ms • Loss tolerance: depending on voice encoding, loss concealment, packet loss rates between 1% and 10% can be tolerated 23

Delay Jitter constant bit rate transmission variable network delay (jitter) client playout delay client reception constant bit rate playout at client buffered data Cumulative data • End-to-end delays of two consecutive packets: difference can be more or less than 20 msec (transmission time difference) time 24

Vo. IP: Fixed Playout Delay • Receiver attempts to playout each chunk exactly q msecs after chunk was generated – Chunk has time stamp t: play out chunk at t+q – Chunk arrives after t+q: data arrives too late for playout: data “lost” • Tradeoff in choosing q: – Large q: less packet loss – Small q: better interactive experience 25

Playout Schedule • • Sender generates packets every 20 msec during talk spurt First packet received at time r First playout schedule: begins at p Second playout schedule: begins at p’ 26

Adaptive Playout Delay • Goal: low playout delay, low late loss rate • Approach: adaptive playout delay adjustment: – Estimate network delay, adjust playout delay at beginning of each talk spurt – Silent periods compressed and elongated – Chunks still played out every 20 msec during talk spurt • Adaptively estimate packet delay: (EWMA - exponentially weighted moving average, recall TCP RTT estimate): di = (1 -a)di-1 + a (ri – ti) delay estimate after ith packet small constant, e. g. 0. 1 time received - time sent (timestamp) measured delay of ith 27

Estimate Playout Delay • Also useful to estimate average deviation of delay, vi : vi = (1 -b)vi-1 + b |ri – ti – di| • Estimates di, vi calculated for every received packet, but used only at start of talk spurt • For first packet in talk spurt, playout time is: playout-timei = ti + di + Kvi • Remaining packets in talkspurt are played out periodically 28

Recovery From Packet Loss Recover from packet loss given small tolerable delay between original transmission and playout • Each ACK/NAK takes ~ one RTT • Alternative: Forward Error Correction (FEC) – send enough bits to allow recovery without retransmission Simple FEC • For every group of n chunks, create redundant chunk by exclusive OR-ing n original chunks • Send n+1 chunks, increasing bandwidth by factor 1/n • Can reconstruct original n chunks if at most one lost chunk from n+1 chunks, with playout delay 29

Another FEC Scheme • Piggyback lower quality stream • Send lower resolution audio stream as redundant information • Non-consecutive loss: receiver can conceal loss • Generalization: can also append (n-1)st and (n-2)nd lowbit rate chunk 30

Interleaving to Conceal Loss • Audio chunks divided into smaller units – e. g. four 5 msec units per 20 msec audio chunk • Packet contains small units from different chunks • If packet lost, still have most of every original chunk • No redundancy overhead, but increases playout delay 31

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • LTE Quality of Service • Qo. S Measurements • Summary 32

Protocols for Real-time Conversational Applications: RTP, SIP • Real-Time Protocol (RTP) [RFC 3550] – RTP specifies packet structure for packets carrying audio, video data – RTP runs in end systems – RTP packets encapsulated in UDP segments – Interoperability: if two Vo. IP applications run RTP, they may be able to work together • SIP: Session Initiation Protocol [RFC 3261] – SIP provides mechanisms for call setup – Determine current IP address of callee – Call management 33

RTP Runs on Top of UDP • RTP libraries provide transport-layer interface • That extends UDP: – – Port numbers, IP addresses Payload type identification Packet sequence numbering Time-stamping 34

RTP and Qo. S • RTP does not provide any mechanism to ensure timely data delivery or other Qo. S guarantees • RTP encapsulation only seen at end systems (not by intermediate routers) – Routers provide best-effort service, making no special effort to ensure that RTP packets arrive at destination in timely matter 35

Real-Time Control Protocol (RTCP) • Works in conjunction with RTP • Each participant in RTP session periodically sends RTCP control packets to all other participants • Each RTCP packet contains sender and/or receiver reports – Report statistics useful to application: # packets sent, # packets lost, inter-arrival jitter • Feedback used to control performance – Sender may modify its transmissions based on feedback 36

Multiple Multicast Senders • Each RTP session – Typically a single multicast address – All RTP /RTCP packets belonging to session use multicast address • RTP, RTCP packets distinguished from each other via distinct port numbers • To limit traffic – Each participant reduces RTCP traffic as number of conference participants increases sender RTP RTCP receivers 37

Stream Synchronization • RTCP can synchronize different media streams within a RTP session – e. g. , videoconferencing app: each sender generates one RTP stream for video, one for audio • Timestamps in RTP packets tied to the video, audio sampling clocks – Not tied to wall-clock time • Each RTCP sender-report packet contains (for most recently generated packet in associated RTP stream) – Timestamp of RTP packet – Wall-clock time for when packet was created • Receivers uses association to synchronize playout of audio, video 38

Session Initiation Protocol (SIP) • All telephone calls, video conference calls take place over Internet • People identified by names or e-mail addresses, rather than by phone numbers • Can reach callee (if callee so desires) – No matter where callee roams – No matter what IP device callee is currently using 39

Setting Up Call to Known IP Address • SIP messages can be sent over TCP or UDP – Here sent over RTP/UDP • Media can be sent over RTP or some other protocol 40

Comparison with H. 323 • H. 323 – Another signaling protocol for real-time, interactive multimedia – Complete, vertically integrated suite of protocols for multimedia conferencing • Signaling, registration, admission control, transport, codecs – Comes from the ITU (telephony) • SIP – – Single component Works with RTP, but does not mandate it Can be combined with other protocols, services Comes from IETF, borrows much of its concepts from HTTP – Uses KISS(Keep It Simple Stupid) principle 41

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • Quality of Service • Qo. S Measurements • Summary 42

Network Support for Multimedia 43

Providing Multiple Classes of Service • One-size fits all service model – Partition traffic into classes – Network treats different classes of traffic differently • Differential service among multiple classes, not among individual connections H 1 0111 H 2 R 1 1. 5 Mbps link R 2 R 1 output interface queue H 3 H 4 44

Principles for Qo. S Guarantees • Packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly R 1 R 2 • Provide protection (isolation) for one class from others 1 Mbps phone R 1 R 2 1. 5 Mbps link packet marking and policing 45

Principles for Qo. S Guarantees (Cont. ) • While providing isolation, it is desirable to use resources as efficiently as possible – Allocating fixed (non-sharable) bandwidth to flow – Inefficient use of bandwidth if flows doesn’t use its allocation 1 Mbps logical link 1 Mbps R 1 phone R 2 1. 5 Mbps link 0. 5 Mbps logical link • Call admission: flow declares its needs, network may block call (e. g. , busy signal) if it cannot meet needs 1 Mbps phone R 1 R 2 1. 5 Mbps link 46

Policing Mechanisms • Limit traffic to not exceed declared parameters • Three common-used criteria – Average rate: how many pkts can be sent per unit time (in the long run) – Peak rate – Burst size: max number of pkts sent consecutively (with no intervening idle) token bucket: limit input to specified burst size and average rate 47

Policing and Qo. S Guarantees • Token bucket, WFQ combine to provide guaranteed upper bound on delay arriving traffic token rate, r bucket size, b per-flow rate, R WFQ arriving traffic D = b/R max 48

Differentiated Services marking • Scalability: simple functions in network core, relatively complex functions at edge routers (or hosts) r b – Signaling, maintaining per -flow router state difficult with large number of flows • Don’t define service classes, provide functional components to build service classes scheduling . . . 49

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • LTE Quality of Service • Qo. S Measurements • Summary 50

Quality Of Service in LTE • Qo. S allows both subscribers and services to be differentiated – The importance of Qo. S increases during periods of congestion – Qo. S impacts admission control decisions • Qo. S is applied between the UE and the PDN Gateway within the LTE network – i. e. it is applicable to an EPS bearer generated from a combination of an E-RAB and S 5/S 8 bearer – EPS bearers and E-RAB can be categorized as Guaranteed Bit Rate (GBR) or non-Guaranteed Bit Rate (non-GBR) 51

Qo. S Class Identifier (QCI) • A pointer to a set of standardized Qo. S characteristics • Reduce both the signaling requirement and the maximum number of possible parameter combinations • The QCI determines which bearers are categorized as GBR and which are categorized as non-GBR • The priority associated with each QCI is applied when forwarding packets across the network • The packet delay budget associated with each QCI defines an upper bound for the packet delay between the UE and the Policy and Charging Enforcement Function (PCEF) within the PDN Gateway – The delay budget figure is applicable to both the uplink and downlink with a 98% confidence level • The packet error loss rate defines the percentage of higher layer packets 52

Allocation and Retention Priority (ARP) • Pre-emption Capability (shall not trigger preemption, may trigger pre-emption) • Pre-emption Vulnerability (not pre-emptable, pre -emptable) • Priority (I to 15) – 15 corresponds to no priority – 14 corresponds to the lowest priority – 1 corresponds to the highest priority 53

Bit Rate • Guaranteed Bit Rate (GBR) – Defines the minimum bit rate which can be expected to be made available to the bearer when required – Configured with values between 0 and 10, 000 Mbps • Maximum Bit Rate (MBR) – Defines the maximum bit rate which can be expected to be made available to the bearer when required – Configured with values between 0 and I 0, 000 Mbps • APN Aggregate Maximum Bit Rate (APN-AMBR) – Defines the maximum allowed throughput for an individual UE based upon the sum of its non-GBR bearers to a specific APN – i. e. the total non-GBR throughput generated by a UE to a specific APN is not allowed to exceed this limit – Configured with values between 1 kbps and 65, 280 Mbps • UE Aggregate Maximum Bit Rate (UE-AMBR) – Defines the maximum allowed throughput for a UE based upon the sum of all its non. GBR bearers – The MME sets the UE-AMBR to the sum of the APN-AMBR of all active APN up to the value of the subscribed UE-AMBR – Configured with values between 0 and I 0, 000 Mbps 54

Channel Quality Indicator (CQI) • CQI reports provide a measure of the downlink channel conditions experienced by the UE • The e. Node. B uses CQI reports within its scheduling and link adaptation algorithms • UE report the CQI value which corresponds to the largest transport block that can be received with a transport block error rate which does not exceed 10 % • When generating CQI values, UE assume they are allocated the modulation scheme and coding rate indicated by the next page table • The CQI Reference Resource is defined by a specific set of Resource Blocks in the frequency domain and a single subframe in the time domain 55

CQI Values 56

UE Configured Transmission Mode • UE configured to use transmission mode 9 derive CQI values from the CSI Reference Signal when configured to report Precoding Matrix Indicators (PMI) and Rank Indicators (RI) – Otherwise, they derive CQI values from the cell specific Reference Signal • UE configured with transmission modes 1 to 8 derive CQI values from the cell specific Reference Signal • If the cell specific Reference Signal is used to derive the reported CQl then the UE accounts for the ratio of the PDSCH Energy Per Resource Element (EPRE) to the cell specific Reference Signal EPRE – If the CSI Reference Signal is used to derive the reported CQI then the UE accounts for the ratio of the PDSCH EPRE to the CSI Reference Signal EPRE 57

Example Throughputs Associated with Each CQI Value (All Figures in Mbps) 58

Network Traffic Measurement • The process of measuring the amount and type of traffic on a particular network • Important with regard to effective bandwidth management • Network performance could be measured using either active or passive techniques – Active techniques (e. g. i. Perf) are more intrusive but are arguably more accurate – Passive techniques have less network overhead and hence can run in the background to be used to trigger network management actions 59

Throughput and Transmission Delay • Throughput – The rate of successful message delivery over a communication channel – The throughput of a communication system may be affected by various factors, including • The limitations of underlying analog physical medium • Available processing power of the system components • End-user behavior • Transmission delay – The amount of time required to push all the packet's bits into the wire – Most packet switched networks use store-and-forward transmission at the input of the link – A switch using store-and-forward transmission will receive (save) the entire packet to the buffer and check it for CRC errors or other problems before sending the first bit of the packet into the outbound link 60

Outline • Overview • Streaming Multimedia • Protocols for Real-time Conversational Applications • Network Support for Multimedia • LTE Quality of Service • Qo. S Measurements • Summary 61

Qo. S Measurements • The description or measurement of the overall performance of a service • Related aspects of the network service – Such as packet loss, bit rate, throughput, transmission delay, availability, jitter, etc. • These performance measures are affected by mechanisms – Such as mobility management, radio resource management, admission control, fair scheduling, channel-dependent scheduling, etc 62

PRACH Preamble Transmit Power • PRACH Preamble Transmit Power = min {PCMAX, PL + Preamble. Rx. Target. Power)} • Preamble. Rx. Target. Power = Initial. Rx. Target. Power + Delta. Preamble + (Preamble. Counter - 1) x Ramping. Step PCMAX : According to the UE power class, e. g. 23 d. Bm for power class 3 PL = Reference Signal transmit power - RSRP measurement Initial. Rx. Target. Power : Range from -120 d. Bm to -90 d. Bm Delta. Preamble : Preamble formats 0 and 1 -> 0 d. B power offset, preamble formats 2 and 3 -> -3 d. B power offset Preamble. Counter : A counter maintained by the UE Ramping. Step : value of 0, 2, 4 or 6 d. B 63

Random Access Response Window • The response window size can be configured as 2, 3, 4, 5, 6, 7, 8 or 10 subframes 64

Reference Signal • The purpose of this Reference Signal is to deliver the reference point for the downlink power • When UE try to figure out DL power – UE measure the power of this reference signal and take it as downlink cell power – i. e. , the power of the signal from a e. Node. B • These reference signal are carried by multiples of specific Resource Elements in each slots • The location of the resource elements are specifically determined by antenna configuration 65

Sounding Reference Signal • Reference signal for e. Node. B to figure out the channel quality of uplink path for each subsections of frequency region • UE is transmitting SRS at the last symbol of a slot • Let e. Node. B to know which section across the overall bandwidth has better channel quality comparing to the other region – Network can allocate the specific frequency region which is the best for each of the UEs 66

Android LTE Measurement • Use Android API level 17 Cell. Signal. Strength. Lte to get - Channel quality indicator Signal strength Signal level Reference signal received power Reference signal received quality Reference signal-to-noise ratio • Measure LTE network delay

Android Wi. Fi Measurement • Use Android API Wifi. Info to get - Basic service set identifier (BSSID) of the current access point - Current frequency - IP address - Current link speed - Received signal strength indicator of the current 802. 11 network

i. Perf / i. Perf 3 • The TCP, UDP and SCTP network bandwidth measurement tool • TCP and SCTP – Measure bandwidth – Report MSS/MTU size and observed read sizes – Support for TCP window size via socket buffers • UDP – – Client can create UDP streams of specified bandwidth Measure packet loss Measure delay jitter Multicast capable • Cross-platform: Windows, Linux, Android, Mac. OS X, Free. BSD, Open. BSD, Net. BSD, Vx. Works, Solaris, . . . 69

i. Perf Features • Client and server can have multiple simultaneous connections • Server handles multiple connections, rather than quitting after a single test • Print periodic, intermediate bandwidth, jitter, and loss reports at specified intervals • Use representative streams to test out how link layer compression affects your achievable bandwidth • Ignore TCP slowstart • Set congestion control algorithm • Dynamic server (client/server parameter exchange) 70

i. Perf General Options Command line option Description -p, --port n The server port for the server to listen on and the client to connect to -f, --format A letter specifying the format to print bandwidth numbers in -i, --interval n Sets the interval time in seconds between periodic bandwidth, jitter, and loss reports -F, --file name client-side: read from the file and write to the network, instead of using random data server-side: read from the network and write to the file, instead of throwing the data away -B, --bind host Bind to host, one of this machine's addresses -J, --json Output in JSON format 71

i. Perf Server Specific Options Command line option Description -s, --server Run i. Perf in server mode -D, --daemon Run the server in background as a daemon -I, --pidfile write a file with the process ID, most useful when running as a daemon 72

i. Perf Client Specific Options Command line option Description -c, --client host Run i. Perf in client mode, connecting to an i. Perf server running on host --sctp Use SCTP rather than TCP -u, --udp Use UDP rather than TCP -b, --bandwidth n[KM] Set target bandwidth to n bits/sec (default 1 Mbit/sec for UDP, unlimited for TCP) -t, --time n The time in seconds to transmit for -P, --parallel n The number of simultaneous connections to make to the server -w, --window n[KM] Sets the socket buffer sizes to the specified value -N, --no-delay Set the TCP no delay option -O, --omit n Omit the first n seconds of the test, to skip past the TCP slowstart period 73

i. Perf Measurement Environment 74

i. Perf Measurement Result • Single TCP stream 75

Streaming Qo. S (You. Tube 1080 p HD) Play You. Tube Video with Chrome, right click mouse button, and choose Properties 76

Streaming Qo. S (You. Tube 480 p) 77

Media Streaming (VLC Player) 78

Multi-Streaming (VLC Player) 79

Streaming Qo. S (VLC Player) 80

Summary • Qo. S of heterogeneous networks is very challenging • Fundamentals of streaming multimedia applications are introduced • Major protocols for real-time conversational applications: RTP, SIP • Network support for multimedia and their Qo. S are examined 81

References • Jim Kurose, Keith Ross, Computer Networking: A Top Down Approach 7 th Edition, Pearson, 2016 • Chris Johnson, Long Term Evolution in Bullets 2 nd Edition, Create. Space Independent Publishing Platform, 2012 • Mario Marchese, Quality of Service over Heterogeneous Networks, 2007, John Wiley & Sons, Inc. • Farnaz Farid, Seyed Shahrestani, Chun Ruan, "Qo. S Evaluation of Heterogeneous Networks Application-based Approach, " International Journal of Computer Networks & Communications (IJCNC) Vol. 8, No. 1, 2016 • Alessio Botta, Antonio Pescape, Giorgio Ventre, "Quality of Service Statistics over Heterogeneous networks: Analysis and Applications, " European Journal of Operational Research Vol. 191, No. 1, 2008 • Ammar Alshamrani, Xuemin Shen, Liang-Liang Xie, "Qo. S Provisioning for Heterogeneous Services in Cooperative Cognitive Radio Networks", IEEE Journal of Selected Areas in Communications, Vol. 29, No. 4, 2011 • Share. Technote, http: //www. sharetechnote. com/ • i. Perf, https: //iperf. fr/ • Open. Air. Interface 5 G, http: //www. openairinterface. org/ • Wiki. Pedia, https: //en. wikipedia. org • VLC, https: //www. videolan. org/vlc/index. html 82