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1. Packet voice transmission requirements Bits per second per voice channel Voice Sample Weice Paitkete ber PPP or Frame Hanne bit rate time ayload second Einermet Relay iii RTP cRTP jer b4 Kbps 20 msec heo bytes 50 z 5 R Oo a G amp at 23 4 jar 64 Kops 30 msec 240 bytes 333 2 2 a s E ho A E ot in le7208 Kbps 20 msec 2o bytes 5c g s 7208 e Kbps 30 msec so bytes 333 oe a 2 oe 7 2 lto ERA P Kbps O msec 0 bytes 5 Kbps Kbps Kbps Note RTP assumes 40 octets RTP UDP IP overhead per packet Compressed RTP icRTP assumes 4 octets RTP UDP IP overhead per packet Ethernet overhead adds 18 octets per packet PPP Frame Relay overhead adds 6 octets per packet ey E a 87 2 A Kbps bp 6 2 66 Kbps bp er bakes ke 3 z 3 2 66 G711 B4 Kbps 40msec 320 bytes 2 Kbps bps i B4 12 Kbps bp 0 2 to Kbps bp pem B L2 L3 payload codec _ bandwidth Bandwidth requirement payload 12 Appendix B Medium Complexity High Complexity G 711 a law and m law G 728 G 726 all versions G 723 all versions G 729a G 729ab G 729a G 729 G 729b G 729 AnnexB AnnexB S3 Codec Complexity Table
2. Technical report IDEO958 November 2009 Evaluation of VolP Codecs over 802 11 Wireless Networks A Measurement Study Master s Thesis in Computer Network Engineering Arbab Nazar hih siiedeni on Chantal E He Lapecp Pi wah Site eater E ohe Oo meee boots EP Lengas Dos Core with Mie softphoos Asterisk Server 6 VT Meanager gt A i School of Information Science Computer and Electrical Engineering Halmstad University y m ni ASG Preface This report is submitted in partial fulfillment of the requirements for the degree of Master in Computer Network Engineering in the School of Information Science Computer and Electrical Engineering at Halmstad University Sweden I express my profound and cordial gratitude to offer thanks to my learned kind and experienced supervisor Per Arne Wiberg for his kind behaviour encouraging attitude constructive suggestions and ever helping supervision My sincerest thanks are to my kind friend Sohail Akhtar who willingly helped me during my project work Arbab Nazar Halmstad University November 2009 Abstract Voice over Internet Protocol VoIP has become very popular in recent days and become the first choice of small to medium companies for voice and data integration in order to cut down the cost and use the IT resources in much more efficient way Another popular technology that is ruling the world after the year 2000 is 802 11 wireless networks The Organization
3. ia cd IT Of Od naaa Pe Tao OO 1F Jc Sd db oF g Fo oof 38 KG Fig 8 9 Wireshark Protocol Analyzer 39 9 Results Discussion and Conclusion In this chapter we discuss the results that we take from the scenario that we setup in the last chapter We established a VoIP call from Node A to Node B and then Node B to Node A so that we can find the more precise results The duration of each call was about 10 minutes approximately We took the time period of each call so long because we wanted to notice the more and more changes during the call The results that we got did not reflect any RFC or standard It is our own described scenario and we want to choose that which codec can do well under bad condition means interference We created a heavy interference as described early the interference was so strong that sometime we drop the wireless connection So there is also chance that there is some error in our results but we conducted these test three to four times before finalizing these results but still there is chance that we get some wrong results But all the results match to the defined characteristics of the codecs We have a lot of VoIP codecs that are used these days but we have chosen these famous codecs for our test G 711 G 729 and Speex The average CPU utilization before starting any test was 10 Now we described the result of each codec one by one that we got in our experiments 9 0 Available Bandwidth Before running any expe
4. 23 no 5 May 2005 p 1056 66 33 SWAN SWAN Service Differentiation in Stateless Wireless Ad Hoc Networks http www iks inf ethz ch education ss04 seminar 41 pdf SWAN Service Differentiation in Stateless Wireless Ad Hoc Networks 34 ZHA05 Baoxian Zhang Hussein T Mouftah QoS Routing for Wireless Ad Hoc Networks Problems Algorithms and Protocols IEEE Communications Magazine v 43 no 10 October 2005 p 110 17 60 35 www wi fi org 36 Wi Fi CERTIFIED for WMM Support for Multimedia Applications with Quality of Service in Wi Fi Networks 37 CCNP ONT Official Exam Certification Guide by Amir Ranjbar 38 QoS over WLAN for the CE World by www metalinkBB com 39 msdn microsoft com 40 http www gss co uk 41 http www voipresource net 42 http www voip news com 43 Solutions to Performance Problems in VoIP Over a 802 11 Wireless LAN by Wei Wang Soung Chang Liew and Victor O K Li From IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY VOL 54 NO 1 JANUARY 2005 44 ntrg cs tced 1e 45 VQ Manager User manual http www manageengine com products vqmanager 46 en wikipedia org wiki G 711 47 www bandcalc com 48 www iana org assignments rtp parameters 59 people xiph org jm papers speex pdf 50 Speex manual from speex org 51 www aero org 52 www ip voip service com 53 www vocal com Ne l l a d a il a d al 6l 11 APPENDIX A
5. Length 40274129 bytes Format Wiresharklicpdumpl libpcap Encapsulation Ethernet Packet size limit 65535 bytes Capture InterFace Intel R PRO Wireless 3945466 Network Connection Microsoft s Packet Scheduler Dropped packets 0 Capture filter none Display Display Filter tcp stream eq 0 Traffic 4 Captured 4 Displayed 4 Marked 4 Packets 41580 1311 0 Between first and last packet 296 759 sec 22 245 sec 58 934 Avg packet size 952 593 bytes 976 363 bytes Bytes 39605025 1202634 Avg bytesisec 133471 271 57658 779 Avg MBik sec 1 065 0 461 Help Figure 8 4 Average Packets per Second Codec Sample Size Packet Second Average Interference Packet Second G 711 20ms 50 600 700 50 50 600 700 G 729 600 700 Table 8 1 Summarized table of codec and interference packets second 8 3 QoS Model and Queuing Mechanism We implement the best effort QoS model because we just treat the voice as regular data Because in other QoS models implementation voice perform reasonable well no matter what codec we are using so we also neglect this factor and do in normal situation implementing Best Effort model Further more we use the FIFO as a queuing mechanism because we want to forward the traffic in the order as it came and do not want to give the priority to the voice traffic In case if want to prioritize the voice traffic than we use the LLQ Low Latency Queue 8 4 Asterisk Server Asterisk is free open source
6. ctivsccs ie teniasatacetsasdsteincesceetee ondncinaniodendtnbecdsieatieaaeddaas 5 QoS for Wireless Net works cccscccccccsccscccccccccccsccccscccscccsccesccesece 5 1 Challenges Involved 1n Wired QOS reari ccc cece cece cece eee e eee e enna eee eee eee eee eA 5 2 Additional QoS Challenges involved in Wireless Networks ccceeeeeeees 5 3 02 11 Medium Access Method 2 5 f 3sey acai ve daws eveucaeaekib biiad eased a a seek 5 4 IEEE 802 lt 1 le Wireless Standards sa4acerndna2hscy ad beeateGukeeaadoiness wheter ieee 6 Implementation of QoS in Wireless Networks sccccsccccccsscssscecs Gale Why is QOS NECESSARY 7 scchediccactste crac tadiatestevateut waa bi tocsensttarax tote caren etree iom 6 2 QoS for Wi Fi Networks offered by WMM ccc ccc cece cece cece eeeeeeeeeeees 0 ACCESS CALC ores xia dasa na hase satan ask orcas eee ae OAN MM ODETA O ensis a hea taxa a Cas hate oelcree aen eh ot ueenm ete cuales 7 Shortcomings of Wireless VoIP Today cccccccccccccccscccccccccceees Dll SCE VIC CAT CA eevee sie cdtou asad Mans ica Satie ueta ce neh a5 Hate Slee Seals eee ae outa kal 7 2 RC ADIN ernea onsets pidacadeeseatins AT A eawechones To HUMES CNCY Fl All Sara tatten cote rac at nct a E E TA Load Balane iNo serrian beter tat tak etek ih E nat beste E Mh inate hic Bes 7 Limited Noof Calls SuppOtt a iie3s0hoh deentdtee ened iaiasaeseiaiiecdad aa iuidiaes 107 Foor Chunks in Pach PACKel 22 05 c
7. license It can be transmitted directly over UDP RTP It 1s a lossy codec which means that during the compression and then decompression processes the original data reduces its quality The designers of the Speex codec have been focused to make it best for VoIP to provide the high quality sound at a low bit rate It uses multiple sampling rates of 32 KHz 16 KHz and 8 KHz Speex uses the Code Excited Linear Predication CELP for coding and encoding The choice of CELP for Speex is because of its performance at both high and low bit rates It is flexible in such a way that it uses multiple sample rates and variable bit rates from 2 Kbps to 44 Kbps which make it so powerful that it continuously maintains the quality of voice It also provides the facility of VAD Voice Activity Detection and it is not so demanding on the processor Furthermore some effort has been made to reduce the noise during the coding encoding process 3 7 Transmission For the transmission of VoIP there are three steps e Signaling is used to setup and end the call For this purpose SIP H 323 and MGCP protocol are used e Packetization is used for sending the voice in the form of packets As discussed above e QoS is used to give the priority to VoIP traffic discuss in next chapter There are four reasons that the VoIP calls are not as clear as PSTN calls these are e Packet loss e Latency e Jitter 14 e Echo 3 7 1 Packet Loss packet
8. with 802 11b hardware and gives the same three Clean channels It also suffers the same kind of interference as 802 11b because this RF range is used by all kinds of devices like microwave ovens cordless phones etc l gt l 2 3 4 5 6 7 a J3 I0 ll 802 11 b g US Channels 22 MHz gt 2 401 GHz 473 GHz Fig 2 5 802 11b g channel 3 Voice over Internet Protocol VoIP is a technology that is meant to replace the analog PSTN phone technology It is a technology that makes it possible to make a phone call over the Internet in the same way as we can do it over standard analog PSTN phone this is the simplest definition of VoIP It is a revolutionary technology that converts the standard internet connection in such a way that we can make free phone calls on it 4 For a technologist the VoIP is a combination of software and hardware that enables us to use the Internet as the medium to place a call which is analog signal into the form of digital data packets discuss later instead of PSTN network 3 1 Ways to make a VoIP Call There 1s not a single way to make a VoIP call There are a lot of ways to make a VoIP call Some of the well known types are as follows 3 1 1 Analog Telephony Adaptor ATA This is the simplest way that uses the existing analog phone With the help of ATA we connect the analog phone with the VoIP network or computer on the LAN ATA is a two way solution it converts the analog signa
9. 2 1 Typical use WLAN at home 2 2 1 Facts about Wireless LAN A Wireless LAN Access Point works like a HUB which means that only one user can send receive data at any given time Shared Signal and Half Duplex It uses the unlicensed bands of Radio Frequency RF Wireless LAN uses the Carrier Sense Multiple Access Collision Avoidance CSMA CA instead of Carrier Sense Multiple Access Collision Detection CSMA CD 2 2 2 Benefits of Wireless LAN Wireless LAN has a lot of benefits here are some of them 2 2 2 1 Mobility With the development of public wireless network a user can access the Internet even if he is not within his normal working area For example almost all the coffee shops and big malls are offering free of charge Internet to their customers 2 2 2 2 Cost Stability As compared to the Wire LAN every time you add a new device you have to run cable while with Wireless LAN include an AP Wireless Router and you can connect a sufficient number of devices to it without any further cost 2 2 2 3Easy to Install It is really easy to install the Wireless AP Router even for a home user To install the wireless equipment at home or SOHO Small Office Home Office there is no need for a technician On the other hand you need a technician every time for installing an RJ45 jack or running a ceiling cable 2 2 3 Deficiency in Wireless LAN There are also some weaknesses in Wireless LAN but we can overcome these w
10. Networks telephony and VoIP as well as the increase in the bandwidth now a day that are used by commercial and home users for transportation of VoIP services Today most IP networks are not designed for carrying the real time and delay sensitive data voice and video Current IP networks only provide the best effort service and also there is no guarantee that the VoIP speech quality will be the same like that PSTN speech quality The implementation of VoIP in wireless networks are rapidly increasing and the reason behind it is mobility The wireless IP network gives the benefit of mobility to its users while the wireless networks have their own special characteristics It is more challenging task when wireless networks are used to carrying the real time traffic with the data traffic 1 Because VoIP is real time application and it is particularly sensitive to packet loss that can be caused in the wireless networks by the weak signals limitation in coverage area and interference from other device that are using the same frequency range as the wireless networks are using In order to deploy the VoIP over wireless networks one should meet a lot of specific requirements for efficient transmission of voice over wireless networks Because of the delay sensitivity of VoIP applications competition for transmission with the data on the same wireless medium causes the degradation in voice quality So it 1s important to properly implement QoS in wir
11. VoIP service providers see the opportunity to get back into the market by acquiring cellular companies There is tough competition between VoIP service providers and cellular companies and this competition 1s in favor of consumers in the long run However in the short run to spend a lot of cash for wireless technology could prove to be the wireless phone equivalent to someone investing in a Betmax video recorder a few decades ago 33 8 Network Scenario and Measurement Tools In this chapter we will discuss the scenario that we built up for measuring the values of different speech codecs and also the tools that we used to accomplished our task 8 1 Network Topology To conduct this experiment we setup a real network environment that shown in Fig 8 1 to measure the value of VoIP codec we use three computers all have Windows Xp out of them one is dedicated Asterisk server and other two laptops are acting as clients with Xlite softphone is installed on it to make a calls between each other We also use VQ Manager Voice Quality Manager to measure the necessary parameters for voice on which we can show that which codec can perform better in wireless networks He Lapip Fi ah Se eee rac Boothe Och reese boot PF Lempar Goel Core with ibe entiphoos sierigk Server E Vl Manage Fig 8 1 Measurement Scenario 8 2 Interference Creation Figure 8 2 gives the overview of our working place Sctereck cere amp o AF f ee e
12. a critical problem When two packets are sent there is no guarantee that they follow the same path to reach the destination Maybe one packet takes a path that contains fewer hops or is less congested so in these cases the packets will not reach the destination at the same time and this causes an unacceptable delay jitter P 5 2 Additional QoS Challenges involved in Wireless Networks Wireless networks have the same challenges that wired networks have but they also involve additional challenges Due to these additional challenges involved in wireless networks QoS methods used in the wired networks are not feasible for wireless networks There are a lot of additional challenges involved some of them are 5 2 1 Interference In wired networks the cause of packets loss is due to severe network congestion Only a negligible amount of data is lost due to corrupt transmission wire On the other hand the wireless link typically suffers more packets loss because of interference during transmission This interference is because of using the free range of frequency bands that other applications use and these create interference with wireless signals Another cause of interference is electrical noise which causes the severe damage to the wireless signal 28 5 2 2 Hidden Node Problem In wireless hidden node is also a serious problem that occurs when a node is visible to the central point Access Point but it is not visible for other nodes
13. are varying for each AC e Minimum wait time e Contention window or random backoff time Both values are larger for low priority traffic and smaller for high priority traffic For each AC the random backoff timer is calculated as the sum of minimum wait time and random value from 0 to the contention window After each collision the value of random backoff time becomes doubled until it reaches to the maximum defined value for each AC After each successful transmission of packets it comes to its initial position at 0 The higher AC packets get the access to the medium quickly because they have smaller random backoff time eiri sirs E Slots 0 3 Slots jie 2 Slots 0 7 Slots Best Effort Priority SIFS_ 3 Slots 0 15 Slots Video Priority racks aaah i3 8 7 Slots Priori Minimum Wait Time Fig 6 6 WMM Access Category Contention Timing Once the client gets access to the medium it is allowed to send the packets for a specific time which depends on the AC and physical rate Typically the medium access 0 15 Slots Random Backoff Time 37 30 time limits are from 0 2 to 3 ms for background priority and video priority respectively in 802 11 a g network For 802 11b network it is ranged from 1 2 to 6 ms 6 5 1 Scheduled Access Scheduled access is a mechanism used by applications in which a client sends a request to the AP to reserve the network resources based on the traffic types By using the central s
14. if 11 call is made This kind of problem is solved by using Admission Control in EDCA in which AP advertise the available bandwidth in its beacons and client check the available bandwidth before transmitting more traffic on the network P 24 5 4 1 2 HCCA HCF Controlled Channel Access HCCA works quite similarly to PCF but instead of waiting for idle time and back off mechanism it uses the access point in this case acting as hybrid coordinator as central control authority that can guarantee the time as well as the time during which each connected node can send data transmission time Every station that wants to join the network must make a request to the AP the request includes the QoS parameter indeed the traffic type the AP Hybrid Coordinator analyzes the QoS parameters to decide whether it can or can not provide the required QoS to the requested node and then admit deny The AP also maintains the centralized schedule of the entire register device to it and it grants the permission to each node to access the wireless medium according to the scheduler as all the parameters were determined at the time of registration P The other differences between PCF and HCCA are that the HC 1s not limited to per station queuing and it can also provide the per session service B1 These are some problems with HCCA However the biggest problem is that the HCCA does not have an ability to work with legacy networks in its neighborhood because
15. it samples at the rate of 8000 samples second The reason for variable sampling rate can be this because we used the latest version of freeware software for testing purpose A recent extension to G 711 G 711 1 allows the addition of narrowband and or wideband 16000 samples s enhancements each at 25 of the bit rate of the included base G 711 bit stream leading to data rates of 64 80 or 96 Kbit s G 711 1 1s compatible with G 711 at 64 Kbit s hence an efficient deployment in existing G 711 based voice over IP VoIP infrastructures is foreseen The G 711 1 coder can encode signals at 16 kHz with a bandwidth of 50 7000 Hz at 80 and 96 Kbit s and for 8 kHz sampling the output may produce signals with a bandwidth ranging from 50 up to 4000 Hz operating at 64 and 80 Kbit s 9 In codec detail this 1s also mentioned that the frame size is not applicable for this codec and the reason behind this is that G 711 is a sample based codec and this field is not implemented on this codec and the codec that works with the frame size is G 723 1 because it is a frame based codec There is also a field for the payload type which indicate that which kind of data the packet is containing and the payload type 0 means that it is PCM audio data according to the IANA Internet Assigned Numbers Authority The formula and table for the values that we used for calculating the bandwidth can be found in Appendix A with reference The bandwidth calcul
16. loss occurs due to the network congestion and it can be solved with proper QoS policy 3 7 2 Latency This is the time delay between end to end VoIP conversations It should not be more than 150 ms for one way 3 7 3 Jitter The variable delay that can cause the voice packet to arrive late or out of order and it can be solved by using jitter buffer 3 7 4 Echo This is the phenomenon in which the callers hear their own voice back and it can be solved by using the echo cancellation G 168 Here we only discuss the SIP because of its simplicity and features 3 7 5 Session Initiation Protocol SIP SIP is the most widely used signaling protocol to control voice and video call over the Internet It is an Application Layer protocol which is based on TCP IP but it is not dependent on the underlying transport layer it can run on anything like UDP or TCP It based on many of the previous protocols that are already in use like HTTP SMTP and DNS etc It is more of an all in one protocol rather than the protocol suite of H 323 It uses the same model that HTTP uses for request response as Fig 3 4 shows For each request of the client a particular function is invoked on the server SIP can work together with many other protocols but it only does the signaling for communication session SIP client uses UDP TCP port 5060 5061 to connect to the other SIP device or SIP server Port 5060 is used for un encrypted signaling traffic while
17. other hand Speex perform reasonable well by mean of reasonable that it did not drop the call and adopt the change in the bandwidth It is also a free open source codec but it did not give the voice quality like G 711 because the reason behind it compresses the voice using Code excited linear prediction algorithm that is a lossy format which means the quality is permanently degraded while reducing the file size while on the other hand it gives the flexibility by compressing the voice at the bit rates ranging from 2 to 44Kbps It uses the sampling rate of 8 KHz Narrowband 16 KHz wideband and 32 KHz ultra wideband If we are not too much quality conscious but we want a robust codec that adapt the change in our wireless environment then SpeexFEC is the reasonable codec It creates some load on the processor and has little bit more delay then other proprietary 58 codecs but this load did not create a significant effect on the modern PC because we were using old computer in our environment may be that is the reason the CPU utilization value is too much If we want constant and low bandwidth codec then G 729 is the best choice but we have to figure out the solution for echo cancellation as well as it is the proprietary codec which means we have to buy a licensed for it For the echo cancellation G 168 1s the best choice G 168 address the performance issue of echo canceller in PSTN by strictly limits the convergence time allow residual echo tole
18. software based telephone private branch exchange PBX implementation which was originally created by Mark Spencer in 1999 It allows the connected telephones or softphones to make calls to each other and also allows connecting to other telephone services like VoIP services It was originally designed for Linux system but now it is also available for Microsoft Windows which is known as AsteriskWin32 In our network environment we use AsteriskWin32 because of its user friendly graphical interface and configuration It did not give as much as option that it originally 36 give on Linux system but still it gives all the options that we need in our study Figure 8 4 shows the Startup window of Asterisk Win32 server Williicoce PRE MAMAGER Free Editon man comrae Took ien Haipo func callerid oe feller D peleted iania itish iino ceo EH Related Fert ions iu Wise UR jee Ti bons hetet ia Asay 8 5 Softphone fered cies Prabin ence Erpa Sp Erca rics 12 FEC rt DAN Wiebe FLL ala GILL ulpa Fei i LLS FOH tetera Speek hajati Epor Adee FEC Fig 8 6 Xlite Softphone Codec selection option a7 On the other hand the software Xlite that we used for making a call is free softphone and it did not have any feature except to that we can adjust the speech codec on it which we want to use as shown in Figure 8 5 and 8 6 It cannot perform any error correction or voice quality enhancement Everything is oper
19. source that generates the desired traffic so that it saves the resources and it need to happen once only in the network All other devices just look at the marking and set the policy according to it 4 6 1 1 Layer 2 Marking CoS uses three bits in the layer header and provides eight levels of markings from 0 to 7 po temetwork Control po A100 lash Override p00 Routine O Layer 2 Marking 19 4 6 1 2 Layer 3 Marking The original TCP IP standard defined a ToS byte The first implementation of marking using the ToS byte was IP precedence and it has only used the leftmost three bits and provides the same eight levels of markings as CoS When the classes of applications increased and we required more level of markings then they introduced the DSCP Differentiated Service Code Point maintaining backward compatibility with IP precedence DSCP uses the left most 6 bits currently three bits for per hop behavior PHB three bits for drop probability in which last 1s always zero and last two bits for flow control but it is not used now Expedited Forwarding 101 110 PHB High Drop Medium Drop 0 Best Effort 000 000 Layer 3 Marking 4 6 2 Queuing Management Queuing mechanism gives us the possibility to control the congested and send it to interface by putting the congested traffic in its own assigned queue according to the configured policy There are two types of queues Hardware Queue in w
20. the video streaming continuously while we were making the call We did this because we do not want to create an ideal situation as we want to create the terrible situation and then want to check that which codec is perform better in such a situation although we know that in working or industry environment these things are taken under consideration that two AP are not operate on the same channel Figure 8 4 gives the summary of the average utilization of the link between access point and one of the node from 5 nodes that connect to the HH access points which shows that how much packets it send and receive in one second In our project we used the sample size of 20 ms for each codec and according to this the total packet that each codec send in one second One packet size sample size 20 ms Total packet in 1 second 1 20 x 10 50 packets second When we take that value of average packet per second from fig 8 4 140 packet second and multiplied it with 5 we get 700 packets per second which means that we have a lot of interference Table 8 1 gives the overview of the average packets second for each codec and the interference packets second on the network These calculations are based upon the parameters that we taken defined during our project rather than the measurement so if we change these parameters than the calculated values also change 35 a Wireshark File Mame CA DOCUME ARBABN 1IL OCALS 1 Tempiwiesharki kasana 700
21. time and this increases the efficiency There are also some moments when no party was speaking and this also reduces these silent periods from conversation and uses even less bandwidth It does not establish a dedicated circuit between the two parties the Internet connection will be really slow if it establishes a circuit between our computer and Web server at any time we were viewing the web page instead it sends the information in the form of packets over the network internet with thousands of redundant paths That is why it is called Packet Switching Circuit switching opens a connection constantly while packet switching opens a connection shortly while it has to send a block of data which is called packet from one side to other and it works in this manner 10 1 Sending node computer breaks the data into small blocks called packets and assigns an address of destination node to each of them 2 Inside the packet it has a payload that contains the actual information that it wants to send to the destination node 3 The computer sends these packets to its default gateway a nearby router 4 The default gateway sends the packets to another router along the way and every router is doing the same until the packets reach the default gateway of the destination computer 5 The default gateway of the destination computer takes out the address label and sends the packet to the desired computer that uses the information
22. wants to implement the VoIP on the wireless network The wireless medium has different nature and requirement than the 802 3 Ethernet and special consideration take into account while implementing the VoIP over wireless network One of the major differences between 802 11 and 802 3 is the bandwidth availability When we implement the VoIP over 802 11 we must use the available bandwidth is an efficient way that the VoIP application use as less bandwidth as possible while retaining the good voice quality In our project we evaluated the different compression and decompression CODEC schemes over the wireless network for VoIP To conduct this test we used two computers for comparing and evaluating performance between different CODEC One dedicated system is used as Asterisk server which is open source PBX software that is ready to use for main stream VoIP implementation Our main focus was on the end to end delay jitter and packet loss for VoIP transmission for different CODECs under the different circumstances in the wireless network The study also analyzed the VoIP codec selection based on the Mean Opinion Score MOS delivered by the softphone In the end we made a comparison between all the proposed CODECs based on all the results and suggested the one Codec that performs well in wireless network Table of Contents LeIM LOGUCHON susiiices acaeveniasiieeenseaiuaiiaswaweminahieeendeaunaiiaswewenugranmeeeetas 2e Wireless NCOWOEK kesicinin EE
23. 105 Bits per sample Not 4Aonfica hie Sampling Rate 5000 Hz Frame Size Not 4onfica bie Packet Size 20 ms RTF Clock Rate 000 Hz Mo of Channels 1 Fig 9 3 1 SpeexFEC Codec Details The field of bits per sample is also not applicable for this codec because it did not use exact bits per sample it uses variable bit rate It also uses the sampling rate of 8 KHz There is also mention that the frame size 1s not applicable for this codec and the reason behind this is that SpeexFEC is also a sample based codec and this field is not implement on this codec and the codec that work with the frame size is G 723 1 because 47 it is a frame based codec There is also field about the payload type which indicate that which kind of data the packet is containing and the payload type 105 is from the dynamic range that defined by the IANA Internet Assigned Numbers Authority for the voice and video application 9 3 2 Initial Call Setup Fig 9 3 1 shows the SIP session that it establishes between the two softphones using the SpeexFEC codec Fig 9 3 2 SpeexFEC Call set between two Nodes 48 9 3 3 Voice Quality 75 Ne ua ee ee 4 0 4 E 5 3 F p a aa e J ag Time Min Max Ag Delay ms E E mf Loss 5 mi Mm z Mm Mas O 3 4 3 5 E 2 5 Hop Err PL IF DMNSMamEe Avg Min Max Cur Jttr Graph g 90 192 168 2 1 wl Belkin 1 1 1 ERR 0 00 D pea E q 90 192 166 101 2 2 ERR 00O 000 palthbr 9 90 194 47 1
24. 5 150 2 2 ERR 00O SiG packets Round Trip 2 2 2 ERR 0 00 134 47 15 130 hop 5 Graph time 10 minute oe 30 i a a 1002p isp TE 1005p Tee 1007p 10 p 1003p 10 10p i liip Fig 9 3 3 Voice Quality Parameters The figures above show the average delay mean opinion score MOS and packet loss The delay was 7 ms which is acceptable because the maximum delay permitted for VoIP call is 150 ms for one end The loss rate is about 3 which is not good but acceptable in the environment where there is lot of interference The MOS is 3 5 which is fairly good when we compare to the Speex codec The value of jitter is O which is incredible which shows that the SpeexFEC is able to manage the end to end delay efficiently X axis represents the call duration time and Y axis represents the random scale for jitter All the units are in millisecond In the delay graph it shows that it has control in variation of delay because the curve slightly changes during the call In the packet loss graph it shows that the packet loss is less because it tries to recover the corrupted packets during transmission 9 3 4 Incoming and Outgoing Call Quality These sub graphs represent the delay loss and mos separately for both incoming and outgoing instead of average They just give the overview of incoming voice quality and outgoing while on the other hand the graphs in 9 3 3 give the detail information about these parameters in combined form 49 MOS MO
25. 504 21514 al be 2534 Fig 9 1 3 Voice Quality Parameters The figures above show the average delay mean opinion score MOS and packet loss The delay was 9 ms which is acceptable because the maximum delay permitted for VoIP call is 150 ms for one end as well as the packet loss is also acceptable and fulfill the described requirement for VoIP On the other hand the MOS is really good which is 4 on the scale of 5 The jitter graph shows that the average jitter value is not too much it is only 0 38 ms where the red bars represent the delay variation According to definition of jitter it is variation in end to end delay and red bars represent where these variations take place during the call X axis represents the call duration time and Y axis represents the random scale for jitter In the graphs of delay loss and mos we see that there are only dots which represent that there was not too much variation in these parameters during the call and only for these and situation or time where the changing took place 9 1 4 Incoming and Outgoing Call Quality These sub graphs represent the delay loss and mos separately for both incoming and outgoing instead of average They just give the overview of incoming voice quality and outgoing while on the other hand the graphs in 9 1 3 give the detail information about these parameters in combined form 43 MOS 3 7 Mos 3 9 Z 4 Poor Excellent F 4 Poor Excellent Outgoing call from Node A Incoming call t
26. 5061 is used for encrypted signaling INVITE SDP Sx Redirect INVITE SDP 100 Trying 180 Ringing SDP e 200 OK a T RA a 100 ACK a a e E N eo RTP RTCP Fig 3 5 SIP Call Setup Its main purpose is to set up and tear down the voice video call but it is also found in many other applications Voice video stream in SIP application is carrying by RTP Real time Transport Protocol and parameters such as port number codec and 15 protocol for this stream are defined by SDP Session Description Protocol SIP is a peer to peer protocol but it requires simple network infrastructure with some intelligence in the endpoints P 3 8 Voice Activity Detection This is a feature in VoIP that stops sending traffic during silence This means it saves our bandwidth On average phone calls are about 35 silence though it really does depend on the call types and also languages matter in VAD Background noise decreases the efficiency of VAD and it does not work with MOH Music on Hold 16 4 Quality of Service QoS This 1s the mechanism used in data communication and networking field to prefer or prioritize some traffic over other so that the preferred traffic moves through the network as quickly as possible It gives different level of preference to different applications It is specially used for VoIP IPTV and video streaming If the network is not congested then there is no need of QoS 7 For VoIP t
27. 7 CPU Utilization Measurement To measure the CPU utilization we used the NTGM freeware software Although it is not a very popular software but still it gives us the idea that how much processor resources used by each codec we used these result as comparison not to precisely defined that how much processor dependent is each codec SSS a a a a a a a a a E a _ Intel R PROAWireless 3945486 Network Connect Broadcom MetLink Tht Fast Ethernet Packet Sc VMware Virtual Ethernet Adapter for vMnett k Reset Counters Renew IF Fig 8 8 NTGM CPU Utilization window 38 8 8 Wireshark Protocol Analyzer Wireshark is free protocol analyzer that is mainly used for troubleshooting in computer networks It also provides support to troubleshoot the VoIP call setup We used Wireshark to check the initial call setup in our scenario It s really easy to use as it gives us GUI graphical user interface We just need to select the interface on which we have to analyze the traffic i711 Wireshark Teil Dia Edi yar ga Capture giym Statistics nlp Bette EXxZEAJA o FSF AABAH ewe a Prone od i rdo Taw _ u n m em H Frame 1 42 Bytes On wire 42 bytes aptur ad a Etherrmet II Sro D Link ocas oA aeaa Gst Intelcor_sdidbiet Coos saciod H Address Reo TUT ion Protocol PeT ood OO LF 3c Sd db Gf go BO lt A ca p a OS ob OO Ol ta Sa Ea saranan ELD 6 Oo G Oo Do b0 c da g
28. Cat 5 to connect devices and computers for the purpose of communication within a small area such as home office or a device within one building The series of 802 11 are referred to as Wireless LAN 2 1 3 Wireless Metropolitan Area Network WMAN Wireless MAN is used to connect the two or more networks that are a distance away from each other like in different cities It is also used to provide the WLAN services like Internet to the entire city WiMAX is usually used as the reference of the Wireless MAN 2 2 Wireless LAN Wireless LAN connects two or more devices using Orthogonal Frequency Division Multiplexing OFDM or Direct Sequence Spread Spectrum DSSS modulation techniques to establish communication between devices within a limited range This gives the advantage that users can move freely within the prescribed area without the fear of disconnection to the network or the burden of changing the position of the wire from one jack to another At home the Wireless LAN is used because of its easy to installation to get rid of wires to move freely and to avoid the drill of every time having to add a new jack when adding a new computer It is also less expensive then the Wire LAN At coffee shops and malls it is used to attract the costumers so that they can surf the Internet while having a coffee or shopping DSL Modem th Ww PC with Wireless Card Wireless AP Laptop with built in Wireless Card Fig
29. Communications February 2005 10 www wi1 fiplanet com tutorials 11 www ciscopress com articles wireless 12 Computer and Communication Networks by Nader F Mir ISBN 10 0 13 174799 1 Pub Date November 02 2006 13 www technology ku edu network services data wireless fcc 802 shtml 14 communication howstuffworks com ip telephony htm 15 searchunifiedcommunications techtarget com 16 www livinginternet com 1 1w_packet_inv htm 17 Signal Processing First by James H McClellan Ronald W Schafer and Mark A Yoder ISBN 10 0130909998 and Publisher Prentice Hall 18 www ip voip service com 19 www cisco com 20 speex org 21 RFC 2543 SIP Session Initiation Protocol 22 http www com dtu dk teletraffic handbook telenook pdf 23 CCNP ONT Reference Sheets by Denise Donohue ISBN 1 58705 315 2 24 www oreillynet com pub a network 25 en wikipedia org wiki Quality of service 26 RFC 791 DARPA INTERNET PROGRAM PROTOCOL SPECIFICATION 27 www cs wustl edu 28 ACOE422 Wireless Networks Notes by Dr Haris Haralambous 29 www wi fiplanet com SS d l al 30 QoS over WLAN for the CE World by www metalinkBB com 31 en wikipedia org wiki IEEE 802 11e 2005 32 LIU05 Qingwen Liu Shengli Zhou Georgios B Giannakis Cross Layer Scheduling With Prescribed QoS Guarantees in Adaptive Wireless Networks IEEE Journal on Selected Areas in Communications v
30. HCCA AP takes the access over the wireless medium when working with the legacy network in both CFP and CP situations so it will interfere with the legacy network Fig 5 3 shows the enhancement at the data link layer to provide QoS for wireless network Fig 5 1 OSI model with modification for 802 11e standard Data Link Layer Physical Layer 5 4 2 Additional 802 11e Specifications There are some additional specifications that are defined by 802 1 le in addition to EDCA HCCA and TXOP to enhance the MAC layer QoS 5 4 2 1 Automatic Power Save Delivery APSD The Automatic Power Save Delivery is a mechanism to save power efficiently as compared to the legacy power saving method used in original 802 11 standards APSD works very well with VoIP phones because the data rates in both directions are approximately the same When voice data are sent to Access Point the AP start sending the buffered voice data in the other direction so after the voice data transmission 1s completed the IP phone goes into a stand by mode until the next voice data has to be sent to the AP P 25 5 4 2 2 Block Acknowledgement Block acknowledgement allows that the block of total TXOP will be acknowledge instead of single frame by doing this it provide less protocol overhead when longer TXOPs are used P 5 4 2 3 NoAck In QoS there are two values for the service class frames that are QoSAck and QoSNoAck The frames with the QoSNoAck wil
31. Operation WMM gives priority to the traffic according to the four classes that are defined in table 6 2 which means the higher the access category you are the higher the chance to transmit By doing that it overcomes the weakness of DCF Distributed Coordination Function which was unable to give the desired service to the multimedia applications The Access Categories AC are designed in such a way that they map to 802 1d 3 bit prioritization tag to the same QoS policy across wired and wireless network segments P If the packet is not assigned a specific AC then it is categorized by default the best effort priority Applications are assigned an AC to each data packet and then this packet is added to one queue out of four voice video best effort and background independently in the client as shown in fig 6 5 The client also has an internal collision resolution mechanism to resolve the collision between different queues as it selects the frame from the highest priority queue first and then lowers it for transmitting The same kind of mechanism is used for external collision resolution to find which client should be granted the access to the medium P 29 Backgronal Best Effort Queue Queue Video Queue tB Fig 6 5 Transmit Queues inside the Clien Fig 4 6 shows an algorithm which is responsible granted the access to the medium based upon priority and it is totally dependent on two timing parameters that
32. S 3 4 3 5 ne 4 2 4 Poor Excellent Foor Excellent Outgoing call from Node A Incoming call to Node A Delay Mi ms Delay W ms Loss Bix Loss Ma Fig 9 3 4 Incoming and Outgoing Call parameters 9 3 5 CPU Utilization Reset Counters Fig 9 3 5 CPU Utilization The CPU utilization was high with the SpeexFEC codec and it was about 22 32 10 before the call establishment the CPU utilization was about 10 to 11 The reason behind the high CPU utilization is because the CPU has to work on the error correction also during the encoding and decoding process 9 4 Speex Wideband Then we enabled the Speex Wideband codec in Xlite softphone on both nodes and established between them and find some results which are as follows 9 4 1 Codec Details CODEC Details Name SPEEM Payload Type 100 Bits per sample Mot Apofica ble Sampling Rate 16000 Hz Frame Size Not 4Aonficable Packet Size 20 ms RTP Clock Rate 000 Hz No of Channels i Fig 9 4 1 SpeexWideband Codec Details Speex Wideband uses the 16 KHz sampling rate and it divide into two bands of 8 KHz signal in which one is representing the low band 0 4KHz and other high band 4 8 KHz There is also stated that the frame size is not applicable for this codec and the reason behind this is that Speex Wideband is also a sample based codec and this field is not implement on this codec There is also field about the payload type which indicate that which kind of data the packe
33. aeda ala Fig 9 6 3 Voice Quality Parameters erlba ilyva elea 9 6 4 CPU Utilization Reset Counters Fig 9 6 4 CPU Utilization The CPU utilization of this codec 1s not too much when compare to pervious codec It seems to be CPU friendly codec it only create a load of about 2 to 2 5 load on the CPU 57 9 7 General Comment on Results We established twelve calls to get these six codecs results in reality we established more than 60 calls to get the average of these 60 calls It was a long process and we noticed a lot of changes during the calls The most notable change was that when we established a call using G 729 codec the call dropped if the interference was too heavy while on the other hand Speex codec never dropped the calls no matter how much the interference was this is true in our case in reality it is not promised but the quality of the call was degraded a lot due to the interference While we were making the call using Speex codec we create an extra interference using Microwave oven because it also operates on 2 4 GHz still the call was not dropped but the quality was so poor We record some of these calls are uploaded at http www ziddu com download 6904326 CodecsResults rar html in the form of wav file In general all the codec perform according to standard description but our purpose was not to check that the standard says right or wrong Our main purpose was to check that which codec perform reasonable eve
34. aeons esanenneses E E 2 Types OF Wireless INCIWOT Kernent a o E 2 V1 PetsonalwAted NetWork crina Ta a wc daardereeiadee aed 2 12 Wireless Local Area Network jacwisicasctavcs tasers edacss dss adetes vedios EEA 2 1 3 Wireless Metropolitan Area Network 0 0 cece cece eee eee eee e teen eee e ees 22 gt War eless LAN aeara a beaten ducal ta a aterm Agee wibari nett eal eee he tacts 25 OO ZA Stand d Seer Gees ar bed hae tedden dane each aan aaa a 5 Voice over Internet Protocole ss cccicevs cet ccctcnadspececdsdeseesesmeceuseeipacocesnans Su ays omake a OLE CaN cacra Site den annua rsa aatamenre nen a ea 32 Methodsof Transmitting VO1CC yaviu esse eaaauitartina OAA IEE TOR 3 3 Transmission of Voice in VoIP Network c cc cece ccc cece cece eee eeeeeeeeeeeaaas SA Pa ketia Oka e e a E E E E NESE 3 PROCESS ol Ovan Za ON eaae AEE A E N a EA VOC OS ariaa ae ear eae ets E EEE A E E EN SATE ONG 2 42 N E E E E E E E T A eens 320 VOICSUACH VIL Dele COM ice nats E a du adiondiaciucs evedeedane dence A OQUALICY OF SERVICE OOS was c te cenvinesss a a a a aa A UW live WWiG NCCU OOS laeron E tata ana ae eae A2 Models ised or OOS esene dacs Siete dla Mohan re Sait bes ts Bian Oe De aiaeesnaaa ds 4 3 Applications that need QOS ccs saccsedawseedeseereadsadenaantese eiaws A e a dese A AC utrenthy Problemi With OOS serna EAEE oeaareaneaWecm ewan N AS Proposed Solution TOL QOS rennandi EErEE nanen EEEN E E ERES AG Hows OQ0S Mmplenented c15
35. and the payload type 106 is from the dynamic range that defined by the IANA Internet Assigned Numbers Authority for the voice and video application CODEC Details Name SPEE FEC Payload Type 106 Bits per sample Mot 4Aooficable Sampling Rate 16000 Hz Frame Size Not 4Apoficable Packet Size 20 ms RTP Clock Rate 6000 Hz Mo of Channels 1 Fig 9 1 1 SpeexWidebandFEC Codec Details 9 5 2 Initial Call Setup Fig 9 4 1 shows the SIP session that it establishes between the two softphones using the Speex WidebandFEC codec 53 Fig 9 5 2 SpeexWidebandFEC Call set between two Nodes 9 5 3 Voice Quality 4 30 E ao 20 10 fi 4 a oiy g 25 A Time o Min Max Awg Delay ms Mm 34 41 Mm 37 Loss 2 M6 mf 7 Mos E 1 3 E i E 1 5 Hop Erre PL IF DA SHame Avg Min Max Cur Jttr Graph 197 168 2 1 wl Belkin 3 i 7 5 244 ie l 10 192 166 10 1 2 2 4 2 075 Nik pe 194 47 15 130 netlogoni30 hh se 13 2 ve 2 24 44 Round Trip 13 2 fe 224 44 netlogon130 hh se 194 47 15 130 hop 3 Graph time 10 minutes 2056 30 i F lisp 1il24p 1125p Iip 1127p 1123p 1123p 1120p 1131p 1132p Fig 9 5 3 Voice Quality Parameters The graphs above show the average delay mean opinion score MOS and packet loss The delay was 37 ms that is not too high when we compare to the other version of the Speex codec The loss rate is about 7 which is too high for voice traffic and is no
36. and the consequential result is that the access point becomes easily overloaded There is no automatic mechanism that will help to balance the load to switch the signals to idle or less busy access points when possible 7 7 5 Limited No of Calls Support The capacity of taking the calls by the wireless standard is also the shortcoming in the way of wireless VoIP A popular standard of wireless is 802 11b which provides the 32 basis of 802 lle has the capacity of 11 Mbps A typical VoIP call consumes about 10Kbps So theoretically the number of VoIP calls that it can take simultaneously is 11Mbps 10Kbps 1100 which correspond to 550 calls each with two streams If we use the GSM 6 10 codec then it only supports about 12 calls because of a lot of overheads in the packet header 1 7 6 A Lot of Chunks in Each Packet A typical VoIP packet contains 40 Bytes of IP UDP User Datagram Protocol RTP Real time Transport Protocol plus 6 to 22 Bytes of Data Link layer overhead for carrying the 20 Bytes of actual voice payload This means that we waste 70 of bandwidth in the form of overheads although the voice is real time traffic and it does not need all these overheads for all the packets Fig 7 1 shows the typical voice packet Header Header Header Header Fig 7 1 Typical Voice Packet 7 7 Collisions VoIP signals are unpredictable in nature due to factor of asynchronous data flow In busy wireless networks the access points could be
37. ated using this formula is not absolute because it is theoretical and can be added or subtracted 6 404160 64 160 Bandwidth usage 82 4Kbp Bandwidth usage 41 9 1 2 Initial Call Setup Fig 9 1 1 explains the SIP session that it establishes between the two softphones In the first step the calling softphone sends the invitation in which it gives the information that it is using G 711 codec and it is representing with INVITE keyword Then the called softphone sends back the response which represent with 100 Trying When the called softphone begins ringing it sends back response which is represented as 180 Ringing When the caller picks up the phone the called softphone send back its response as 200 OK The calling softphone sends the response as ACK Then the actual conversation is transmitted via RTP G 711 When one end hangs up the softphone sends the response as BYE and the other end is responding with 200 OK Fig 9 1 2 G 711 Call set between two Nodes 42 9 1 3 Voice Quality E T 5 1 0 F a A 0 5 d A 25 Time Min Max Avg Loss mi mi mi MOS E E Mm 4 Hop Err PLS IF DA Shame avg Min Max Cur JIttr Graph i 10 192 168 2 1 wl Belkin 1 1 3 2 D501 12 1 10 197 168 10 1 z 2 12 12 1 36 l i 10 194 47 15 130 netlogoni30 hh se 2 2 3 3 0 38 mo Pa Ps Round Trip 3 3 0 38 netlogon 130 hh se 194 47 15 130 hop 3 Graph time 10 minutes 13 E r z F ziga 2a 2a fhe era 2ta 2
38. ating at normal situation 8 6 Voice Quality Measurement Software To measure the voice quality parameters we used the VQ Manager VQ Manger is the most reliable and trusted software for monitoring the VoIP network for voice quality call traffic bandwidth utilization and keep track of active calls and failed calls VQ Manager can monitor any device or user agent that supports SIP Skinny and RTP RTCP It gives the summary of all the parameters for specified time period as well as it show in the form of graphs It also gives the Information on what is going on in VoIP network and how it performs are presented in the form of comprehensive intuitive and informative reports Roran Page SUMtnheary Remar Call Volume a ok 4 Waice Quality An A wee Pan i 3 F i x p E a Seas ERES EE a ae i a om chil ce a Erga Oe Oe BD i i M a a ge eh yg ga at a gi a alte aT a ggg gt Time Tare Moat data overy i hr Plot data every i Be T iit F iias ag Total Cats 1934 RSuccessfulig2s9 Ol Wetting 3 Blunsuctesstul b 78 tl Delay tmp Wi Mr Boia Seed Qualy Calle ETD Antever Geure Robo ASR 65 0 Ci teria MS E E j Poor Quality Cats g Peak Usage Penod OF hrs 132 ET woes Bo Bo Bo Unindwored Cade ag low Usage Pernod 26 hre i ay mE EE B36 Error ads eer average Cal Durabon 3 ming 26 pect C R Factor E 6 E yi E Minmanitored Gals l CallRate 1Mmuta E c d O Tolerable W Poor Configure Fig 8 7 VQ Manger 8
39. cause it 1s theoretical and can be plus or minus 6 40 20 8 20 Bandwidth usage 26 4 Kbps Bandwidth usage 9 6 2 Initial Call Setup Fig 9 5 1 shows the SIP session that it establishes between the two softphones using the G 729 codec Fig 9 6 2 G 729 Call set between two Nodes 9 6 3 Voice Quality The graphs below represent the jitter loss and average delay for G 729 codec The delay of 11 ms is acceptable and within the range of 150 ms for one end of VoIP call The MOS for this codec is really good that is 4 out of 5 Jitter graph represent the average jitter value that 1s about 2 which shows that the there 1s not too much delay variation X axis represents the call duration time and Y axis represents the random scale for jitter In the graphs of delay loss and jitter we see that there are only dots which represent that there was not too much variation in these parameters during the call and only these and situation or time where the changing took place 56 G wo 54 10 gt T E s5 1 0 f x r S ps l Time Min Max Awg Delay ms 1i ii W ii Loss 5 mi mi mi hos lt 4 m4 Hop Err PL IP DA SMame vq Min Max Cur Jttr Graph 192 168 2 1 wl Belkin i i 2 2 Dai i 192 168 10 1 1 1 2 2 D43 194 47 15 130 netlogonl30 hh se 2 l g 1 2 00 Round Trip 2 1 8 1 2 00 Graph time 10 minutes 30 a netlogoni30 hh se 194 47 15 130 hop 3 21198 Ziea aleia EEE alza
40. ccen sida vatiwiah assem heley ere dost area aE Ter OSOD S apse amp cata cdo e en fora es EE E aha EE Sis wi Patents seated onices TAO Te AWC A E AEE E E E OO eee one iis need o aan aie Aca E E A TDSC CULL rc oh hh at hs and tea landanes teticet hee telat e OE TIO Tee way Waleran T EE E AT A 8 Network Sce nario and Measurement Tools ccccccccccccccccccccccccccccces Sl INGtwOrk T OpOlOGysa 245 shana te a ned cect ta a a ada B72 MMEren Ce TAO race s aici ttn ct cae cixeane a Mores vaecien daa aot tvon nua AAN 8 3 QoS Model and Queuing Mechanism 0 ccc ccc cece cece cece cece eee e enn nnnnes BE SUSE Ol arses heed cect N tas Seo narnia tr a chatted E BO LOD MOM Ge tik eautine red ous ligt once ncaa ed anus Aeneas uaa A 8 6 Voice Quality Measurement Software 00 c cece cece cece eee cece ee eee eee nnnees 8 7 CPU Wil Zation Measure Ment s cc2e nice downs a E sa cedemanhceaceca pees 8 8 Wireshark Protocol AnalyZer 0 cc ccc cece cece eee e eee eee aa eee eee ee eee bees 9 Results Discussion and COnclusiOn c cccccccccccccccccccccccccccccceccceccees D OFA Vailable Bandwidth tusks eoeiedesseseateareansehudout Jean peewee 9 3 SpeexFEC 9A SPEE W IGG DANG cc recce conse shat cannon suas deeuunea de een oom E ieee D5 SPCex Wideband FEC 252553 fad hat E E A boa Gan anmren awieheddeeaadad 9 6 G 729 codec 9 7 General Comment on Results 0 cece ccc cc cecccccccccecuceccucecse
41. cheduling control mechanism average network latency can be reduced It is designed to meet the latency and throughput requirement of the applications by assigning different times to the applications as to when they can transmit The client sends a reservation request to the AP and the AP assigns the access to the medium by checking different Transmission Specification TSPEC parameters of the request like packet sizes data rate service interval etc In scheduled access the client knows what kind of resources it needs in advance and the AP makes the assumptions of all the parameters as defined above to effectively schedule concurrent traffic 6 5 2 EDCA plus Admission Control EDCA assures that high priority traffic gain access to the medium quickly as compared to the low priority traffic and it does not degrade the performance of high priority traffic Another unique addition to the admission control also prevents the traffic from the same priority class from disturbing the already admitted traffic if the network is not capable of handling both traffic streams E g EDCA assures that the voice traffic stream gets priority over the data stream while the addition of admission control prevents another voice stream from entering the network if it is not capable of providing the same level of service to both and both streams crashing Admission control continuously evaluates the network s resources and only allows the addi
42. contained inside the packets to resemble these packets so that it gets the data in its original format This provides very efficient redundant and cheap lines for data transmission Its also frees the two computers that are communicating with each other so that meanwhile they are also communicating with other computers and sharing information also with them 3 3 Transmission of Voice in VoIP Network The voice in the VoIP network Internet is transmitting in the form of data packets The figure 3 2 represents the communication between the IP phone and analog phone but any combination of devices can be involved in the real world example transmission between analog to analog devices as well as digital to analog etc 4 Sound 1 Sound i wih Router Prone N Voigt Gateway 3 Analog 2 Packets E J Fig 3 2 Transmission of Voice packets in VoIP network These are the general steps that are involved in the transmission of voice over VolP network 1 Sounds that came from IP phones are grouped into small packets of sound usually 20ms then sampled and converted into digital format 2 Each of these packets of sound is then assigned header that consists of data link IP UDP and Reliable Transport Protocol RTP and shipped on the VoIP network 11 3 The analog phone on the other end is unable to understand the packets so the gateway in this case router will do the job and convert the packets back into ana
43. cs are there 3 6 1 G 711 Codec This uses the Pulse Code Modulation PCM to code encode compress and decompress the analog telephony signal by using 8 KHz sampling and 64Kbps bit rate It literally codes each and every sample into its own binary calculation for the transmission on the Internet 3 6 2 G 729 Codec This operates at 8 Kbps and 8 KHz sampling frequency by using the Conjugate Structure Algebraic Code Excited Linear Prediction algorithm It also uses a human voice Codebook as a dictionary to work and Look Ahead of 5 ms Special mathematical algorithms are used for voice synthesis The complexity requirement of processing power lies at 15 Appendix B because of its low bandwidth requirement It is the most favorable codec that is used in VoIP applications nowadays 13 3 6 3 Mean Opinion Score Mean Opinion Score MOS is a scale that is used to describe the quality of the perceived media after compression and transmission It lies between 1 and 5 where 1 indicates the lowest quality and 5 indicates the best quality Here is the MOS used for the codec for VoIP applications P Codec Bit Rate kbps MOS Score Compression Delay ms 6 G 729 CS ACELP 88 892 G 729aCS ACELP 88 8 Table 3 1 Mean Opinion Score Table 3 6 4 Speex Speex is a free open source speech codec that can be used for VoIP applications and it is absolutely free licensed under the Berkeley Software Distribution BSD
44. cucteceeeeces 9 8 Conclusion 10 Referen e aeaa a a nd a sates naa ewe 11 Appendix A 12 Appendix B 34 34 34 36 36 37 38 38 39 40 40 41 44 47 50 53 55 58 58 60 62 62 1 INTRODUCTION VoIP Voice over Internet Protocol is a technology that gives you the feature of making calls using the internet connection instead of using your analog phone line Some VoIP services allow you to make call to other people who are using the same VoIP services while other give you chance to make call to anyone who has telephone number including local and international Some VoIP services only work on computer or special type of VoIP phones while other services work on traditional phone 1f you connect it with VoIP adapter VoIP technology used IP based networks to carry the voice With VoIP the service providers can offer the telephony services as well as the traditional data service using the same existing IP infrastructure By doing this it will not only increase their revenue but customers also save a lot of money while comparing to buy these services individual In recent years growth of VoIP has been increased dramatically which gains a lot of attention from the network engineering research communities consequence the result of rapid commercial solution as well as the network improvements Other factors that make this popular are ongoing decrease in quality differences between the existing PSTN Public Switched Telephone
45. d Distributed Channel Access also referred to as Contention based medium access e HCCA HCF Controlled Channel Access also known as Controlled medium access 802 lle also operates in two modes CP and CFP EDCA is only used in Contention Period where as can be used in both modes The reason it is called hybrid is 1t combined both DCF and PCF methods 5 4 1 1 EDCA Enhanced Distributed Channel Access EDCA Enhanced Distributed Channel Access gives chance to high priority traffic to be sent first then low priority traffic afterwards which means that the node with the low priority traffic has to wait more than high priority traffic In addition when defining priority it also assigns Transmit Opportunity TXOP to each priority level TXOP is a specific time interval during which a node can traffic as much traffic as possible If the frame is too big that it can be not be transmitted in one TXOP then it should be fragmented into smaller frames The uses of TXOP also solve the problem for low rate stations that gained the access to the medium for a long time to transmit its frames in the original legacy 802 11 DCF MAC M The basic purpose of EDCA is to provide the QoS mechanism upon the classes types to send high priority data first than low priority data but there can also be situations in which data that belong to the same priority class have to protected from each other e g where the network can only accommodate 10 VoIP call and
46. due to fact that the router fails to deliver some packets because its buffer memory is full A router can drop some packets or all packets and this depends upon the status of the network i e whether it is fully congestion or not This packets loss can degrade the performance of voice and video traffic The router in its 17 default behavior drops all packets once its queue become full This leads towards some serious problems gt e TCP global synchronization TCP buffer starvation 4 2 Models used for QoS Basically there are three types of models that are used to implement QoS e Best Effort e Integrated Services IntServ e Differentiated Services DiffServ 4 2 1 Best Effort This is a default model in which the traffic is being sent in the order in which it arrives It gives the equal treatment to all traffic types and does not give any guarantee of delivery The advantage of its usage is that its scalability Internet used the best effort model for the delivery of all kinds of traffic 4 2 2 IntServ IntServ guarantees some specific level of service to each flow of traffic throughout the network for specific period of time It uses Resource Reservation Protocol RSVP to reserve the path throughout the network An RSVP enabled router requests specific level of service to its next hop router and each router along the way reserving the specific bandwidth for that flow for some length of time If the network is not able to p
47. e lt 9 co Mode A users using connected to HH AP H AP Fig 8 2 Working Environment We create a lot of interference for taking the results For this we setup our experiment wireless router Belkin on the same wireless channel Channel 6 as of the other two wireless APs hh netlogon and hh student are already configured on channel 6 as we can see in Figure 8 3 34 Channel Use cop YolF_ over Wireless Channel oOo MAC Address 00 1c df 8c Oc 63 TxPower 100 mw 4 Strength 34 dBm TU Antennas 2 5 T 5 Speed Mbit 54 Using GPS Mo PEL Auth Type Open GPS Signal M A Frag Threshold satelites N A ATS Threshold 2347 2437 MHz Frequency im Di Ja mm Status de iv Not Ava iv Not Ava iv Not va SSID Channel hh netlogon hh netlogon Security ASSl of Not A LO NAA L af Noth CO NAIL Requir CO NAA L Rates Su dO Mbs dO Mbs dO Mbs dO Mbs MAC Add 00 b 55 00 b 85 8 00 0b 85 8 OO 0b 55 amp Network OFDM 24 OFDM 24 OFDM 24 OFDM 24 Infrastruc Infrastruct Infrastruct Infrastruct Infrastruct hh student Requir LA NAA LL VolP_over Wi al Noth MD 34 A40 Mbs 001e di8 OFDM24 lt tH ig Not Ava p Connec E hh studert G 6 Infrastruct Figure 8 3 Access Point Channel Description Five users are connected these access points and doing
48. e is also mention that the frame size is not applicable for this codec and the reason behind this 1s that speex is a sample based codec 44 and this field is not implement on this codec and the codec that work with the frame size is G 723 1 because it is a frame based codec There is also field for the payload type which indicate that which kind of data the packet 1s containing and the payload type 97 is from the dynamic range that defined by the IANA Internet Assigned Numbers Authority for the voice and video application The formula used for calculating the bandwidth can be found in Appendix A with reference and parameters used in this formula are taken from reference 5 The bandwidth calculated using this formula is not absolute because it is theoretical and can be plus or minus 6 40 4 30 11 30 Bandwidth Usage 27 8 Kbps Bandwidth Usage 9 2 2 Initial Call Setup Fig 9 2 1 shows the SIP session that it establishes between the two softphones using the speex codec Fig 9 2 2 Speex Call set between two Nodes 45 9 2 3 Voice Quality Delay La Loss F a 2 Wi n i j Time Min Max Avg Delay rns Bs Mii Mio Loss E i E 7 E MoS 2 6 as Hop Err PL IP DN SPlane Avg Min Max Cur Jttr Graph 197 166 2 1 wl Belkin 5 1 ood 1 6 5 197 166 10 1 2 2 5 ae yi 194 47 15 130 netlogoni30 hh se 6 2 on 3 6 00 Round Trip 6 2 22 3 6 00 netlagondsdhhse 194 47 15 130 ho
49. e solutions Temporal Key Integrity Protocol TKIP Message Integrity Code MIC and 802 1x 2 2 4 3 WPA2 IEEE 802 11 This was released by the Wi Fi Alliance in 2004 It needs hardware upgrade and uses the AES Advance Encryption Standard as encryption It is backward compatible with TKIP hardware also which means that if client connect to AP that only support TKIP encryption it provides that and if the client supports the AES then it will provide this 2 2 5 Interference Because the Wireless LAN uses the unlicensed band so the other devices in the wireless network that are using the same channel such as microwave ovens and cordless phones can interfere the wireless signal and this can create significant impacts on the performance of Wireless LAN Properly designed networks can mitigate the impact of interference 2 2 6 Architecture of Wireless LAN 2 2 6 1 Station All devices that have the ability to connect to the wireless medium are called stations All stations have wireless NICs Wireless station can be Access Point AP or clients Access Points are devices that are capable of sending and receiving RF to the backbone network for wireless clients that associate with it Wireless clients are any devices that have Wireless NIC e g laptops IP phones PDA etc 2 2 6 2 Basic Service Set Basic Service Set BSS is a collection of all the Access Point and clients that can communicate with each othe
50. eaknesses by using the correct design model and strategy 2 2 3 1Security In Wire LAN the hacker or malicious person must have to be inside the building and have the access to the RJ 45 jack to attack the network or sniff the packets However for Wireless LAN the situation is totally different because of the nature of radio transmission the intruder does not have to be inside the building Radio signals leak outside the building and anyone within the range can use them to access the internal network of the company due to which we must have to implement proper security so that no unauthorized person can access our Wireless LAN 2 2 4 Types of Security 2 2 4 1 WEP Wired Equivalent Privacy WEP is the original wireless security standard release in 1997 to secure the wireless networks but it 1s the weakest form of wireless security It consist of 40 104 bits WEP static key 24 bits Initialization Vector IV 40 104 bits WEP static key 24 bits IV Static WEP key is always the same while the IV changes for every packet It sends the keys in a beacon header and if you are able to catch enough packets then you can break the security In 2005 an FBI team made an experiment with WEP and broke it in 3 minutes 7 2 2 4 2 WPA Wi Fi Protected Access It was released by the Wi Fi Alliance to overcome the weakness of WEP in 2003 It uses the same hardware as WEP but it gives the far better security than WEP It gave the thre
51. eless networks because it gives the priority to voice packets over data packets Security is also a major problem in VoIP wireless adds another layer of security concerns as the packets are transmitting over the open medium air instead of cables So it is easy to capture these packets and convert them back to wav form Common VoIP protocols like SIP Session Initiation Protocol have their own security vulnerabilities End to end delay and jitter have significant impact on the quality of voice in VoIP In our project end to end delay as well as jitter will be investigated and analyzed for different codec and suggested the best codec upon these parameters There are a number of CODEC used for VoIP these days and each of them have its different and unique characteristics When different codecs are used and the information that we want to transmit is arranged into different frame sizes then the transmission result also change This result is more prominent when the conversion of transport stream changes into the Real time Transport Protocol RTP packets Normally one or more voice frames are put into one RTP packet RTP packets are then put into the UDP packets and at the end into IP packets before transmitting across the network IP packets are then encapsulated into MAC frames switch from one node to another Delays are added during the processing that occurs at each node while across the path from source to destination The measurement of th
52. ender moved and there was also a possibility that the new route did not provide the level of service that the previous route guaranteed The problem was also similar if the destination moved 5 3 802 11 Medium Access Method 802 11 originally used the DCF Distributed Coordination Function as the basic medium access control mechanism but it could also implement PCF Point Coordination Function on the top of DCF optionally 5 3 1 DCF Distributed Coordination Function DCF was originally used by the 802 11 MAC layer to share the access to the medium between the multiple nodes DCF uses CSMA CA Carrier Sense Multiple Access with Collision Avoidance to share the access of medium between nodes but it also optionally uses RTS CTS Request to Send Clear to Send It has limitations but those limitations mean you cannot use it for QoS 1 If many nodes start sending data at the same time many collisions occur which causes the lower available bandwidth and there is no concept of high and low priority traffic in DCF Only once can a station get access to the medium it can keep the medium as long as it wants and if it has a low bit rate then it will create problem for all other stations by taking a long time to send its traffic DCF does not provide any QoS guarantee 5 3 2 PCF Point Coordination Function 802 11 standards optionally defined the PCF Point Coordination Function which is used for the transmission of time sens
53. etworking research community QoS working group gave a suggestion that increasing bandwidth is a more suitable solution than to implement the QoS 4 5 Proposed Solution for QoS Multi Service Access Everywhere MUSE proposed the QoS concept first and that we must agree on discrete jitter value for each class that 1s implementing in the network This solution has some benefits and these are P gt e End users are able to notice the difference in service quality e Easy to implement e Weare able to predict the end to end delay 4 6 How is QoS Implemented There are a lot of factors that are taken into account while implement the QoS Some of them are 4 6 1 Classification and Marking The first step in the implementation of the QoS is to classify the traffic Until the traffic is classified it is not possible to give the specific level of service and traffic is often classified by IP address source or destination or application gt After the classification the next step is to mark the classified traffic to the appropriate marking and the location where the traffic is marked is known as the trust boundary If the device that marked the traffic is trust then that marked traffic passes through the network and each device in the network gives it defined service level and if the device is not marked then some trusted device must re mark this traffic again gt Classification and marking should be done as close as possible to the
54. fficient and feasible for home users and industry Some of them are here 2 3 1 802 11a Officially this was released in September 1999 It uses the RF range of 5 0 GHz It uses the Orthogonal Frequency Division Multiplexing OFDM with 52 subcarriers to carry the data and gives the data rate of 54 Mbps theoretically with 8 Data Rates of 6 9 12 18 24 36 48 and 54 Mbps but it actually gives the throughput of 27 Mbps It is not cross compatible with 802 11b g but it has more Clean channels of 12 to 23 5 180 5200 5220 5240 5 260 5 280 5 300 5 320 MHz MHz MHz MHz MHz MHz MHz MHz Fig 2 4 802 11a channel 2 3 2 802 11b This was also officially released as of September 1999 It uses the RF range of 2 4 GHz and Direct Sequence Spread Spectrum DSSS to carry the data and gives the data rate of 11 Mbps theoretically with 4 Data Rates of 1 2 5 5 and 11 Mbps but it actually gives the throughput of 5 Mbps It is the most popular standard in 802 11 line up but it only gives 3 Clean channels The major problem is that 802 11b station cannot decode 802 11b radio signals H gt 2 3 3 802 11 This was released in June 2003 by IEEE It uses the same RF range of 2 4 GHz as used by 802 11b with Direct Sequence Spread Spectrum DSSS amp Orthogonal Frequency Division Multiplexing OFDM to carry the data and gives the data rate of upto 54 Mbps with 12 Data Rates but throughput of 22 Mbps It is totally backward compatible
55. h The latter can cause the poor quality VoIP calls or even calls can be dropped or packets delayed Signal strength is another problem that may be due to user roaming As the user moves away from the access point the signals become weaker and this degrades the quality of the VoIP call 7 3 Emergency 911 calls Emergency 911 services are also a hurdle for VoIP in the sense that all service providers are not offering the e911 service as standard Even the service providers that offer the e911 service are often unable to route the calls to the intended location of the user after hours of waiting Many users see it as a major problem when they compare it with PSTN phones However FCC Federal Communication Commission imposed the rule that all the service providers must provide the facility of e911 call service but this has not yet been implemented completely 7 4 Load Balancing Because of the asynchronous and untimed nature of wireless communication clients pc or handset need to be constantly monitoring the signal strength due to the fact that the transmitter and receiver must leave a dedicated channel open That leads towards the problem that open channel have a huge load on the access points W1 F1 is capable of handling of a specific amount of clients simultaneously on each access point although theoretically the value is much larger During peak communication demand the use of VoIP takes the number of real connections to hundred s
56. he definition of QoS is the ability to transmit and receive the voice continuously and clearly and without any disturbance 4 1 Why We Need QoS There are four major problems that force us to implement the QoS these are e Lack of Bandwidth e Delay e Jitter e Packet Loss i Bandwidth When a lot of applications are running then these multiple applications flows are using the same links which means that the available bandwidth for each application is even smaller and it 1s equal to the bandwidth of the smallest link divided by the number of application data flows If there is insufficient bandwidth then it will severely damage the time and delay sensitive application like VoIP and video streaming One way is to increase the bandwidth but this is expensive and an alternative way is to implement the QoS which ensures the bandwidth for sensitive applications gt ii Delay Delay is the time that a packet takes to reach its destination There are four types of delay that network traffic faces e Processing Delay e Queuing Delay e Serialization Delay e Propagation Delay ili Jitter When the packets reaches from source to the destination with variance delay then it is known as jitter and it can create a serious problem for audio and video traffic The major cause for the jitter is varying delay in the router s queues as well as the varying path between source and destination iv Packet Loss Issues Packets drop happens
57. hich it keeps the packets that are ready to put from transmit ring to media wire and that is always FIFO First In First Out Software Queue is a memory assigned to each interface where the traffic waits in case the transmit ring is full When the traffic is put into the queues this means that the network is congested and it is caused by the speed mismatch and link aggregation 1 4 6 2 1 Queuing Strategies There are some queuing strategies that manage the traffic during network congestion Some famous queuing strategies are FIFO Queuing Priority Queuing PQ Round Robin Queuing RRQ Weight Fair Queuing WFQ Class Based Weighted Fair Queuing CBWFQ Low Latency Queuing LLQ 4 6 3 Congestion Avoidance Congestion Avoidance is the strategy to implement some technique to avoid the network congestion To fulfill this purpose we used two techniques e Random Early Detection 20 e Weighted Random Early Detection 4 6 3 1 Random Early Detection RED This tries to avoid the congestion by random drops packets from TCP flows to minimize the synchronization Once the queue is filled above the threshold level the value at which the maximum packets can occupy by the queue it starts to drop packets randomly from the queue Dropping becomes more aggressive as the queues fill 1 4 6 3 2 Weighted Random Early Detection WRED Random Early Detection is not able to make the difference between different flows of traffic so it is not s
58. hnology used to establish this circuit is called circuit switching Fig 3 1 PSTN Network These are steps involved in a typical old telephone call 1 When we pick the phone it gives us the dial tone which is the indication that we are connected to local exchange 2 Then we dial the phone number of the other end where we wish to reach 3 Our call is taking route through the local exchange switch to the central office 4 The central office then connects our local exchange to the other end party local exchange 5 A connection has been established between us and other party that is taking a path through the several interconnected switches 6 The phone on the other end rings and other party answer the phone call 7 A dedicated circuit is opened between two parties 8 We talk for specific time period and then end the call 9 When we close the call the dedicated circuit is terminated between us and other party and releases our line as well as all the lines in between the path that it was using For the time that we talked on the phone a dedicated circuit was opened between us and other party and the call was forwarded by using a fixed rate of 64 Kbps in each direction a total of 128 Kbps in both directions 3 2 2 Packet Switching Packet Switching Network works in this manner that when one party is talking the other party is only listening to him so it only uses half of the bandwidth of the connection at any
59. ing instead of average They just give the overview of incoming voice quality 46 and outgoing while on the other hand the graphs in 9 2 3 give the detail information about these parameters in combined form Mos Mos 2 3 3 4 4 2 Poor Excellent Poor Excellent Outgoing call from Node A Incoming call to Node A Delay Mi4 ms Beles sms Lass Mes ae mii Fig 9 2 4 Incoming and Outgoing Call parameters 9 2 5 CPU Utilization Reset Counters Fig 9 2 5 CPU Utilization The average CPU utilization was at 10 before establishing the call After establishing the call we noted that the Speex creates a load of about 10 on the CPU utilization 9 3 SpeexFEC Then we enabled the SpeexFEC codec in Xlite softphone on both nodes and established between them and find some results which are as follows 9 3 1 Codec Details SpeexFEC Forward Error Correction is the enhancement to the speex codec in a way that SpeexFEC sends some error correction code in each packet to avoid the retransmission and detect the receiving system to sends some error correction code in each packet to improve the reliability of data by putting some known code into the data before transmission This feature enables the receiving system to detect and try to correct the errors due to corruption from the receiver This technique enables the decoder to correct the errors without retransmission of the original information CODEC Details Name SPEEM FEC Payload Type
60. ions to give preferences to one class over another WMM also defines the protocol that will be used between the AP and the QoS enabled client Tables 6 2 shows the AC Access Categories defined by the WMM Access Categories 802 1d Tags Highest Priority Video Priority Prioritize traffic over other best effort traffic Best Effort Priority Traffic from legacy or not QoS enabled devices Background Priority Table 6 2 WMM Access Categories Fig 6 3 and Fig 6 4 show the effect on throughput if we implement the WMM Fig 6 3 shows that the WMM gives the higher priority to the voice traffic over the other Both voice and other low priority have enough resources during the initial 10 seconds and the third data stream creates exceeded network capacity due to it transmission 28 demands But WMM gives the same priority to the voice traffic for its smooth transmission so that it will not affect it In the fig 6 4 all the streams have been given the same priority to access the wireless medium and the introduction of the third stream affects the transmission for all previous streams because without WMM all data streams have equal priority With WMM Wira baa poh ee edo iT Yh jy AA 70 Mbpsj Background prionty 14 Meos p z 10 15 z Tima s Fig 6 3 With WMM Without WMM new data stream affects voice stream m 16 ai 14 4 12 _ 2 ST OA OA an E 6 4 4 0 5 10 15 20 Time s Fig 6 4 Without WMM 6 4 WMM
61. is delay is of our main interest in this project and to check its impact on the quality of voice 2 Wireless Network A wireless network is any kind of computer network that is connected wirelessly meaning that the nodes are connected to each other or to the telecommunications network which connected them to the internet or backbone wired network without the need of wires Wireless networks use the electromagnetic waves commonly radio waves for carrying the signals and data between the nodes and it is implemented at the physical layer meant to replace the wires 2 1 Types of Wireless Network Some common types of wireless networks are as follows 1 Personal Area Network PAN 2 Wireless Local Area Network WLAN 3 Wireless Metropolitan Area Network WMAN 2 1 1 Personal Area Network PAN Wireless PAN connects the devices in a relatively small area which is generally a few meters using radio waves It can be used to communicate among the devices e g Bluetooth uses PAN to connect the wireless headset to a laptop or wireless mice to laptops or connects the device to the backbone network and Internet We can also create Wireless PAN with other network technologies like Z Wave Bluetooth and IrDA 2 1 2 Wireless Local Area Network WLAN Wireless Local Area Network WLAN is an alternative way to connect computers and devices in Local Area Network LAN by using radio waves while LAN technology uses Ethernet cable e g
62. itive traffic It only works with infrastructure mode in which AP acts as the coordinator controlling which station can transmit during any given period of time by sending beacon at regular intervals 0 1 second P 23 It also defines two periods Contention Free Period CFP and Contention Period CP In CPF the AP sends the Contention Free Pool CF Poll packets to all the nodes that are operating in PCF mode to give them the chance to send the traffic In CP it simply used DCF Thus PCF is a contention free protocol and it gives chance to the stations to transmit data frames with regular time delays between data frame transmissions and due to this it provides better management for QoS Unfortunately PCF has very limited support for QoS and it does not provide the support for classes of traffic To overcome these weaknesses the EEE defines the new wireless standard 802 1 le to provide the better QoS for time sensitive applications 5 4 IEEE 802 11e Wireless Standard 802 1 le is an amendment to the original IEEE 802 11 standard that introduced the QoS support for wireless networks by modifying the Layer 2 Media Access Control of OSI model It has critical importance for time and delay sensitive applications like VoIP video streaming IPTV etc 5 4 1 802 11e MAC Layer Operation 802 lle uses HCF Hybrid Coordination Function to support QoS HCF further defines the two medium access mechanisms e EDCA Enhance
63. jitter All the units are in millisecond In the delay graph it shows that the delay was gradually increase during the call while on the other hand the packet loss decrease which may conclude that the codec take more time for encoding and send few packets that is reason the loss graph gradually decrease 9 4 4 Incoming and Outgoing Call Quality MOS Mos 7 9 2 4 2 4 Foor Excellent Poor Er cellent Outgoing call from Node A Incoming call to Node A Delay Miso ms Delay M62 ms Loss Mi Loss Mas Fig 9 4 4 Incoming and Outgoing Call parameters These sub graphs represent the delay loss and mos separately for both incoming and outgoing instead of average They just give the overview of incoming voice quality and outgoing while on the other hand the graphs in 9 4 3 give the detail information about these parameters in combined form 9 4 5 CPU Utilization Reset Counters Fig 9 4 5 CPU Utilization The CPU utilization shows that the wideband did not create too much load on the CPU it created almost the same load as the Speex narrowband created on the CPU 52 9 5 Speex WidebandFEC In this section we show the result that we take after enabled the Speex WidebandFEC codec 9 5 1 Codec Details Speex WidebandFEC also uses the same 16 KHz sampling rate and it is also a sample based codec as indicated by the frame size field There is also field about the payload type which indicate that which kind of data the packet is containing
64. l into a digital signal that came from the phone and sends it to the VoIP network and also converts it back from the digital signal into analog when there is a response answer from the VoIP network 3 1 2 IP Phones These are special types of phones which are designed to work directly with the VoIP network or Internet gateway and without the need of ATA or an additional device It uses the RJ 45 Ethernet jack instead of RJ 11 jack that analog phones normally use but it has the entire feature that the old phones have and some additional features 3 1 3 Computer to Computer In this case we simply have to install a freeware mostly software with the existing speaker microphone and there is no need to pay any additional cost other than the monthly fee to the ISP In this case the long distance 1s also not a problem because the packets have to be carried totally in the Packet Switching Network between computers to computers H4 3 2 Methods of Transmitting Voice Before discussing that how voice is transmitted over the IP network we should look into the Circuit Switching and Packet Switching networks for better understanding 3 2 1 Circuit Switching Circuit Switching technology is used in legacy telephone networks We can say that it is the based of PSTN When we make a call between two phones we reserve the line path between the two phones that is only dedicated to these two and it is referred to as Circuit and the tec
65. l not be acknowledged and this avoids the retransmission of time sensitive data P 5 4 2 4 Direct Link Setup Direct Link Setup allows the direct transmission of frames from node to node within the Basic Service Set all the devices are associated local wireless LAN and this design is really useful where the node to node transfer is frequently used P 26 6 Implementation of QoS in Wireless Networks In this chapter we look at the implementation of QoS in wireless networks and it is definitely different to all the mechanisms we have talked about so far We shall look at the wireless standard that is built by the wireless association the Wi Fi Alliance and its partners Cisco amp Microsoft for the implementation of wireless QoS which 1s different instead of scheduling the packet it really scheduling time QoS becomes more important for the wireless access world as we get new devices that use wireless technology The idea of QoS is needed because the clients as we can see in Fig 6 1 such as W1 F1 phones laptops PDAs handheld computer run applications like the critical data service or VoIP that needs different levels of services than simply web surfing Wireless technology works on CSMA CA that is same as token ring technology in which a token moves around the ring and whatever node gets it can send packets The same situation is in wireless world whatever node gets access to the medium can send packets as long as it wants This
66. log signal 4 Analog signal is forwarded to the phone and then the phone converts the signal into audio and plays it on its speaker A typical voice packet contains all this information regardless what codec we are using Layer 2 Layer 3 Layer 4 RTP Voice Sample CRC Header Header Header Header Fig 3 3 A normal voice packet 3 4 Packetization As we discussed above the voice is an analog signal and the Internet is only capable of transmitting the digital signals in the form of bits Here we shall discuss how this 1s possible 3 4 1 How the Voice Signal becomes Packets bits There are a lot of factors involved in the conversion of voice signal into packets but we can summarize these factors into four steps 1 Take too many samples of the analog signal 2 Calculate the number representing each sample by using the Pulse Amplitude Modulation PAM which is called the quantization 3 Convert these numbers into binary 0 amp 1 4 Compress the signal by using the suitable codec scheme but this step is totally optional In the conversion of Signal to Packet one theory plays an important role and that is Nyquist Theorem 3 4 2 Nyquist Theorem If you sample a signal in regular intervals of at least twice the highest channel frequency the samples will contain enough information to accurately reconstruct the signal H Nyquist Theorem deals the frequency range of 300 4000 Hz or 300 4 KHz 3 5 Process of Quanti
67. means that the more nodes which are connected to the AP access point the more the bandwidth is saturated and the fewer time slots there are for the transmission of traffic 5 Broadband Canneaction Fig 6 1 A typical Wi Fi network today 6 1 Why is QoS Necessary Legacy wireless networks give equal access to all the devices that are connected to them and when the demand of traffic exceed from the available bandwidth throughput is reduced for all data streams regardless of type of traffic type This behavior is strongly affected by the application type A one or two second delay in sending a printing job from computer to a printer is not noticed by the user but even a one second delay can disturb the VoIP call It can even drop the call In the residential and industrial markets multimedia applications potentially create a new need for QoS 2i 6 2 QoS for Wi Fi Networks offered by WMM The Wi Fi Alliance played an active role in the development of QoS for multimedia applications by developing WMM wi fi multimedia The main advantages of WMM are 6 2 1 Relationship with IEEE 802 11le The 802 1le standard is approved but WMM is used still everywhere because it provide the base to 802 1 le standard Although 802 1le has additional features it uses the WMM as for its core functions E g The Wi Fi Alliance has already developed the test plan for the scheduled access capability 6 2 2 Industry Support WMM was develo
68. n in horrible environment When we established a call using G 729 Codec the quality was good but there was some echo At some stage during the call the echo problem even dominated and it also performed badly So it quality was also varying during call SpeexFEC codec was the one which performed really well although it create the load on CPU was about 21 but it has jitter value of zero many time and sometime 0 01 but on the average in has jitter value of zero which seems to be unbelievable When we establish the call using SpeexWideband with FEC forward error correction codec the thing was even worse We heard the voice of the caller after long time it was really annoying This codec was the only that perform really badly But still in this condition the call was established which proved one thing that Speex is a robust codec The highest CPU utilization was for SpeexFEC codec and lowest was for G 729 codec These values are not absolute when we change the environment the values will also change So all these values are dependent on system and environment in which the wireless network is operating 9 8 Conclusion From the results discussion and most importantly after hearing all the voice codec draw the conclusion that if we have enough bandwidth then G 711 is the best choice because of its less delay low packet loss and voice quality But is not robust codec and it cannot be able to adapt the change in the bandwidth and drop the call On the
69. o Node A Delay Ei ms Delay Mo ms Loss E3 Loss Be Fig 9 1 4 Incoming and Outgoing Call parameters 9 1 5 CPU Utilization Reset Counters Fig 9 1 5 CPU Utilization As in the beginning of this chapter we explained that the average CPU utilization was at 10 before establishing the call After establishing the call we noted that the G 711 only creates a load of 3 on the CPU 9 2 Speex Then we enabled the Speex codec in Xlite softphone on both nodes and established between them and find some results which are as follows 9 2 1 Codec Details Speex is free open source codec for audio compression for speech that originally designed for VoIP applications It is built on the speech coding algorithm named Code excited linear prediction It is a lossy format which means the quality is permanently degraded while reducing the file size Speex compress the voice at the bit rates ranging from 2 to 44Kbps It uses the sampling rate of 8 KHz Narrowband 16 KHz wideband and 32 KHz ultra wideband CODEC Details Name SPEEM Payload Type g7 Bits per sample Mot Aoplicabie Sampling Rate 5000 Hz Frame Size Mot 4Anoficable Packet Size 20 ms RTP Clock Rate S000 Hz No of Channels 1 Fig 9 2 1 Speex Codec Details The field of bits per sample is not applicable for this codec because it did not use exact bits per sample it uses variable bit rate F Speex uses the sampling rate of 8 KHz which means it uses the narrowband here Ther
70. overloaded during the peak flow of data The key issue arising with this is collisions that is when too many signals arrive same time at the access point and some get delayed The delays should not be more than 30 ms for true voice quality 7 8 Battery Life Battery life is an important factor in VoIP devices IP based communication 1s asynchronous which means that the VoIP device has to be active to properly handle the signal both in a call and out of a call as compared to a cellular device Cellular devices wake up at regular intervals of every 30 ms This 30 ms enables the device to save power briefly between receiving and transmitting information As a result the battery life increases VoIP handsets cannot preserve the power cyclic 7 7 9 Security VoIP that is transmitted on the wireless network requires three levels of security one for voice transmission one for its associated control signaling and configuration and one for WLAN channel on which voice traffic is transmitted s security creates hurdles for many companies in implementing 802 11 networks and the same issues exist for VoIP to transmit over wireless This is because the VoIP packets are carried by anc and it is easier for a hacker to hack into them while moving across the network 7 10 Three way War The wireless market was already established even before the VoIP came Cellular companies are much more active as compared to the VoIP service providers
71. p 5 Graph time 10 minute ae 30 E rs aa Lilla Li a 1134 Liqa 11154 11164 iira 1184 1 194 izla Fig 9 2 3 Voice Quality Parameters The figures above show the average delay mean opinion score MOS and packet loss The delay was 10 ms which is acceptable because the maximum delay permitted for VoIP call is 150 ms for one end but the packet loss is not acceptable for good voice quality it should not be more than 1 otherwise it create jerk in the voice call On the other hand the MOS is 2 8 on the scale of 5 which is not too good but just acceptable The jitter graph shows that the average jitter value that is acceptable and is 6 ms where the red bars represent the delay variation According to definition of jitter it is variation in end to end delay and red bars represent where these variations take place during the call X axis represents the call duration time and Y axis represents the random scale for jitter All the units are in millisecond In the delay graph it shows that the delay for voice is almost same because it did not show any change in the curve while in packets loss graph the curve goes down in the middle which shows that the interference gradually decrease after that point and mos graph the curve also goes straight which shows that the even the decrease in the packet loss does not affect on it 9 2 4 Incoming and Outgoing Call Quality These sub graphs represent the delay loss and mos separately for both incoming and outgo
72. ped by Wi Fi Alliance with their partners that are all the world leading industries so all the major industry player adopted the WMM 6 2 3 DiffServ Differentiated services WMM defines QoS classes structure that is based on the IETF DiffServ architecture which means that it is cross compatible with wired networks Individual packet is labeled with DSCP or 802 1d tags 6 2 4 Universal Plug and Play UPnP Compatibility DiffServ enables UPnP QoS to maintain WMM and allows administrators to develop network wide policies that can be applied to wired and wireless infrastructure 6 3 Access Categories The Wi Fi Alliance came up with a completely new standard called the WMM It was meant to replace the QoS mechanism used by the wired network Layer 2 switches use the 802 lp tag as a QoS mechanism which defines 8 classes of service by using 3 bits to transfer data across the trunk links The APs when they communicating wirelessly with the clients they do not have tags to use between the client and APs and we end up with degraded service when the AP is saturated WMM boils down the eight levels of services into four levels voice video best effort and background and the first vendor that came into action was Cisco who called these classes platinum gold silver and bronze WMM gives flexibility to the administrator to choose the appropriate policy for network wide use and also gives permiss
73. r There are two kinds of BSS Independent BSS and Infrastructure BSS Each BSS has its ID which is known as SSID 2 2 6 3 Extended Service Set amp Distribution System Extended Service Set is a collection of more than one BSS and Distribution System is used to connect the Access Points of BSS in Extended Service Set 2 2 7 Roaming in Wireless LAN There are two types of roaming in Wireless LAN 2 2 1 Layer 2 Roaming In layer 2 roaming mobile nodes migrate from one access point AP to other APs but within the same network The migration may be due to the fact that the node missed too many beacons data reaches maximum retry count or data rate is shift down Mobile node retains its previous open sessions using some software mechanism Roaning Domain Fig 2 2 Layer 2 Roaming 2 2 7 2 Layer 3 Roaming a k a Mobile IP In layer 3 Roaming mobile node left its home network and goes to the foreign network network other than the one the node belongs to There is some special method used to authenticate the mobile node in the foreign network The mobile node lost all of its previous open sessions foi Cera T Morri Dora E i Fig 2 3 Layer 3 Roaming 2 3 802 11 Standards The IEEE 802 11 is a collection of standards that are used to carry the data in Wireless LAN using the RF waves in the range of 2 4 GHz and 5 0 GHz A lot of amendments have been made in original 802 11 standards to make it more e
74. rance for varying signal levels and allowed divergence in the presence of destabilizing narrow band energy For VoIP the echo canceller can be put after an audio codec to minimize the echo from local hardware Which means more money we have to spend but if we look it in long run we can save a lot of bandwidth which mean save money We can increase the efficiency and save the bandwidth by using the RTP header compression By using this technique we compress the RTP UDP and IP headers from 40 bytes to in between 4 6 bytes and the bandwidth consumption reduced by 60 for compressed voice packets To sum up we can say this that each codec had it own unique properties and there is not even a single codec which can be 100 perfect for wireless network Although Speex was develop for VoIP but not for Wireless VoIP but still 1t perform really well 59 10 Reference 1 www fcc gov voip 2 Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network by Jianguo Cao and Mark Gregory 3 blogs techrepublic com com 4 www cambridge org us catalogue wireless 5 www wirelessnets com resources downloads wireless industry report 2007 6 oreilly com 7 CCNP Quick Reference by Denise Donohue Brent Stewart and Jerold Swan ISBN 13 978 1 58720 236 0 8 en wikipedia org wiki Wireless LAN 9 WIRELESS LAN SECURITY AND IEEE 802 111 by JYH CHENG CHEN MING CHIA JIANG AND YI WEN LIU IEEE Wireless
75. riment we checked the available bandwidth with the Cisco Speed Meter software which is the most authenticated and widely used software in the industry to check the wireless available bandwidth and the result is shown in fig 9 0 Internet download speed Normal 202 KB sec Internet upload speed Normal 179 KB sec Tested at 10 46 PM Router Fig 9 0 Available Bandwidth As from the above figure we can see that the available bandwidth 1s not too high it is really low but we did this by purpose because we want to check the performance in the terrible situation and the bandwidth was even less when we create more interference as we described before 40 9 1 G 711 Pulse code modulation PCM First we enabled the G 711 codec in Xlite softphone on both nodes and established calls between them and find some results which are as follows 9 1 1 Codec Details For the brief introduction G 711 digitizes the analog voice into 64000 bps or 64Kbps and it did not use any voice compression This is used as the default standard for the PBX vendors CODEC Details Mame PEMU Payload Type T Bits per sample 5 Sampling Rate anabe Frame Size Mot Aoplicabie Packet Size 0 ms RTP Clock Rate S000 Hz No of Channels 1 Fig 9 1 1 PCM Codec Details The formal name that use for G 711 codec is Pulse code modulation PCM of voice frequencies G 711 uses the logarithmic pulse code modulation PCM samples for the signals of voice frequencies and
76. rovide the specific level of service then the session 1s not allowed RSVP can work with any type of traffic but is mainly used for delay and time sensitive traffic like VoIP gt 4 2 3 DiffServ DiffServ is the most efficient and widely used QoS model It classifies the network traffic into classes and each class consists of the traffic that needs the same type of QoS e g VoIP traffic needs different QoS than email but email traffic can have the same QoS requirement as web traffic So we can put the email and web traffic into the same class The distinction between the classes is based on the certain bits value in the Layer 2 and Layer 3 header We can treat the traffic as we want each hop long the way to the destination and it is referred to as per hop behavior PHB gt 4 3 Applications that need QoS There are two types of applications one is called inelastic that has a specific bandwidth requirement and maximum latency to perform its function and other is elastic applications that can work well with little available bandwidth These are some important inelastic applications that require a certain amount of bandwidth VoIP Online games Multimedia streaming IPTV 18 4 4 Currently Problem with QoS Currently the Internet is not governed by any central authority It 1s administered by many independent authorities and currently the Internet only uses best effort for transmission and Internet2 a non profit advanced n
77. s CPU utilization with its described features it is not too much Reset Counters Fig 9 5 5 CPU Utilization 9 6 G 729 codec G 729 is mostly used these days for VoIP application It is proprietary codec we enabled it on our softphone on both sides and take some results that are as follows 9 6 1 Codec Details This operates at 8 Kbps and 8 KHz sampling frequency by using the Conjugate Structure Algebraic Code Excited Linear Prediction algorithm It also uses a human voice Codebook as a dictionary to work and Look Ahead of 5ms Special mathematical algorithms are used for voice synthesis The complexity lies at 15 because of its low bandwidth requirement It is the most favorable codec that is used in VoIP applications nowadays P gt CODEC Details Name S729 Payload Type 16 Bits per sample Not Annfica hie Sampling Rate 5000 Hz Frame Size 10 ms Packet Size 20 ms RTP Clock Rate 5000 Hz Mo of Channels 1 Fig 9 6 1 PCM Codec Details It requires 10 ms input frames as indicated by the frame size field and generates frames of 80 bits long Bits per sample are not implementing for this codec as it works 55 with frames It uses the payload type of 18 which is reserved for G 729 codec by the IANA Internet Assigned Numbers Authority The formula and table for the values that we used for calculating the bandwidth can be found in Appendix A with reference The bandwidth calculated using this formula is not absolute be
78. t acceptable and the reason behind this loss may be interference that affects the wireless link quality The MOS is 1 5 out of 5 which shows that it perform really bad and not acceptable The value of jitter is 24 ms and it reflects the variation in end to end delay X axis represents the call duration time and Y axis represents the random scale for jitter All the units are in millisecond In the delay graph it shows that the delay was almost constant during the call session and packet loss also remains the same during this time when the call session was established The reason may be the same that the link quality may be too bad that the delay time and packet loss rate did not improve during the call 9 5 4 Incoming and Outgoing Call Quality These sub graphs represent the delay loss and mos separately for both incoming and outgoing instead of average They just give the overview of incoming voice quality and outgoing while on the other hand the graphs in 9 5 3 give the detail information about these parameters in compact form 54 MOS MOS 1 8 1 3 2 4 1 2 4 Poor Excellent Poor Excellent Outgoing call from Node A Incoming call to Node A Delay M45 ms Delay E i ms Loss Loss Mos Fig 9 5 4 Incoming and Outgoing Call parameters 9 5 5 CPU Utilization The CPU utilization of Speex WidebandFEC is high when compare to other codecs and is about 15 when minus from the initial CPU utilization before establishing the call When we compare it
79. t is containing and the payload type 100 is from the dynamic range that defined by the IANA Internet Assigned Numbers Authority for the voice and video application 50 9 4 2 Initial Call Setup Fig 9 4 1 shows the SIP session that it establishes between the two softphones using the Speex Wideband codec Fig 9 4 2 SpeexWideband Call set between two Nodes 9 4 3 Voice Quality n Loss BW A i Time Min Max Awg Delay ms Mm ci Mss MM 56 Loss 5b Ma ms Mm z Mos Mea Mm 3 5 E 2 5 51 Hop Err PL IP DASMame Avg Min Max Cur Jtkr Graph 192 166 2 1 wl Belkin 14 l 60 a dlo 1 10 192 168 10 1 9 6 2 21 2 fo 194 47 15 130 9 netlogoni30 hh se 15 l 45 1 26 00 Round Trip 15 1 45 1 26 00 netlogon130 hh se 194 47 15 130 hop 3 Graph time 10 minutes Tr 30 z m i E 2i lba leilva 121184 1fi13a 12 i20a 1eizla leigea leiesa liga Wi2Sa Fig 9 4 3 Voice Quality Parameters The graphs above show the average delay mean opinion score MOS and packet loss The delay was 56 ms that is high when we compare to the other version of the Speex codec but still within the range of 150 ms The loss rate 1s about 3 which is not good but acceptable in the environment where there is lot of interference The MOS is 2 8 which are not good The value of jitter is 26 ms which reflect the delay X axis represents the call duration time and Y axis represents the random scale for
80. that are communicating with the AP and it leads towards the media access control difficulties This problem can however be solved by using RTS CTS messages but it is still taken into account when implement QoS 5 2 3 Multipath Propagation The signal may take different paths between the sender and the receiver due to diffraction scattering and reflection which causes various problems like time dispersion it can cause problems if high data rate digital modulation 1s employed and the signal 22 can reach at the receiver directly or phase shifted The distorted signal depends on the phase of the different parts 5 2 4 Handoff In wireless networks handoff is a mechanism whereby one AP gives the control of its associated node to another AP During handoff it has to re establish the route reservation that starts at the mobile node 5 2 5 Propagation Delay Propagation delay is another obstacle for implementing QoS in wireless networks because some wireless networks are spread over area of square Kilometers and propagation delay is an extra burden for the real time communication that requires guarantee on delay The problem exists mostly in MAN Metropolitan Area Networks Finally it is really difficult to maintain the guaranteed service in wireless networks because the nodes are mobile The scheme that involves the resource reservation did not work well with it because the new route had to establish to the destination 1f the s
81. tional streams if the resources are available It is mandatory for the AP while optional for the end station P 31 7 Shortcomings of Wireless VoIP Today The VoIP is deploying successfully on wireless these days but still there are a lot of challenges to be faced when it come to the wireless medium In this chapter we briefly describe some of them 7 1 Service Area Coverage area is a huge problem for VoIP over wireless Even latest versions of access points have ranges of only some yards or feet For the real wireless coverage for VoIP however the coverage needs to be extensive To overcome this problem we need to deploy a huge number of access points within a coverage area which also leads us to another problem namely that this solution 1s too expensive 7 2 Reliability Most of the users can tolerate delay with non real time applications For example we do not want to receive our emails instantly even if we get after some minutes it should not be too much of a problem When we talk about the voice the PSTN provides us with the reliable and high quality transmissions also the system is always on The reliability is the major reason for companies not converting their existing PSTN phones to VoIP quickly real time applications are so sensitive for packet loss as compared to other data applications as they do not bear the packet loss and delay Wireless adds another layer of problems like RF interference and signals strengt
82. uitable for real time traffic On the other hand Weighted RED drops the traffic on the value of its IP precedence or DSCP It combined with CBWFQ to implement the DiffServ s PHB in which each PHB has a unique WRED profile to identify a minimum threshold maximum threshold and MPD Mark Probability Denominator for that profile The decision of packets drop is taken on the value of IP precedence if the DSCP based value is not configured P gt 21 5 QoS for Wireless Networks Sure and guaranteed service is required for audio and video transmission Many service providers have been offering IP telephony and video services in addition to internet service to their customer and the motivation behind offering these services is the flexibility of packet switched networks But the challenges involved in the efficient audio or video transmission are those of compression methods that encode decode the streams at VBR variable bit rate and also assigning the highest bit rates to these streams 5 1 Challenges Involved in Wired QoS There are a lot of challenges involved in providing guaranteed services in a network and the main challenge in providing QoS is congestion in the network it increases the end to end delay because the packet has to stay for long in the queue at each hop in the network Packets loss also increases when the queue becomes full and it limits the throughput because we have to retransmit the lost packets Multi Path routing is also
83. zation Here is the Quantization process that 1s used to calculate the number that represents each sample by using PAM A phone system was designed to produce and capture the signals that have frequency less than 4 KHz and according to the Nyquist Theorem at least twice the highest channel frequency So Nyquist 4000 is higher frequency Sample Size 4000 x 2 8000 Thus we sample 8000 lines second Each 1 1 8000 of a second 12 KHZ Wotage PAM al ll a i E u Pa We hl al Fc ee en Were Hoge bens ly i i o Lt cH Time 2ec T 4o Fig 3 4 Quantization Process So each 1 contain 1 byte so 8000 x 8 64000 bps or 64 Kbps which is equal to one voice channel DSO 3 6 Codec A normal call that uses the VoIP infrastructure needs to go through two different stages The first stage is that takes the standard analog signal and converts it into digital to transmit on the internet The second stage is that when it reaches the other end it needs to be converted back to analog otherwise it is not recognizable There are two types of algorithms used for the coding decoding These are I Waveform Algorithm It encode everything e PCM Pulse Code Modulation e ADPCM Adaptive Differential Pulse Code Modulation II Source Algorithm It only encode the changes e CS ACELP Conjugate Structure Algebraic Code Excited Linear Prediction e LDCELP Low Delay Code Excited Linear Prediction Some of the famous code

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