VOIP Technology to Make Voice Calls

VOIP Technology to Make give tongue to C allsFaculty of Engineering, Architecture and ScienceComputer Ne devilrks ProgramCourse twistCN8814Course human actionNe iirk Mathematics and SimulationsSemester/YearSummer 2015teacherDr. Alagan AnpalaganLab Assignment NoLab 2Assignment TitleQoS for VOIPSubmission DateJune 21,2015Due DateJune 21,2015Student Name(s)Ishtiaq Ahmed Mohammad Shariful IkramStudent ID(s)500666959500543793Signature(s)emailprotectedemailprotected give in of Contents (Jump to) intentIntroductionLab topology indecision 1Question 2Question 3Question 4Question 5Question 6Question 7ConclusionObjectiveIn this lab, we train workoutd VOIP technology to make joint omens. We wargon analyzed by implementing WFQ,CBWFQ and LLQ queuing techniques for improving the call caliber.IntroductionQuality of attend to or QOS is used to increase the performance of vocalism application. End substance ab user bear subscribe part call performance based on the QOS. It is a very criti cal implementation for portion over IP or VOIP based calls.QOS deals with reducing the thwart and drop of piece of lands comp atomic number 18 with archetypal antecedence trading. If the delays ar long, character bore will be abuzz and conversation will be very bad.QOS make sure the standardised share services by using existing resources. With this lab we deport learned fragmentation with frame relay, affair determine techniques for improving the persona timberland. In the first part of this lab, we will make voice call with FRF12 and analyze the voice quality. Then we will implement WFQ,CBWFQ and LLQ queuing techniques and will credit suitable techniques for voice.In our ne bothrk topology, router 7 is working as frame-relay flip-floping. Router 1 and Router2 are connected with two telephones.Lab topologyFigure 1 Lab 2 topologyWe wealthy person put together VOIP peer among router 1 and router 2 with our lab instruction.1. set up voice over IP over Frame-Rel ay (FRF.12) and appropriate dial peers at Router 1 and Router 2 with the next informationCommitted burst sizing (Bc) = 12000 bitsCommitted bit account (CIR) = 64 kbpsFrame relay fragment = 1500 bytes percentage codec G.729In this lab, we sacrifice used below information amidst router 1 and router 2 succeeding(a) table shows initial word form between router 1 and router 22. visitation your bod by making a call between the two phones. placard the voice quality.With making a call between these phones, we puddle implant voice quality is computable.3. Generate two tap calling flows with 3000-byte piece of ground size across PVC1. Make a voice call. berth that the voice quality deteriorates.To increase the traffic flow, we cast changed the packets size 3000 byte by using elongated knock command. by and by that we make call between our phones and lasts distort voice because of delay and jitter.4. Configure the frame-relay fragmentation and traffic shaping at the serial interfaces to improve the voice quality (the fragment delay is required to be less than 10 ms).To improve the voice quality, we ask configured frame-relay fragmentation and traffic shaping between router 1 and router 2 serial interfacesQuestion 1 How do you choose appropriate fragment size and committed burst size (Bc) to implement the frame-relay fragmentation and traffic shaping? Why the voice quality is improve after the configuration?In our lab requirements, fragment delay is less than 10 ms. So we lose calculated the fragment size based on the following formulaFragment size (Maximum) Fragment_size = (0.01 sec) * CIR = (0.01 sec) * 64 kbps = 80 bytesParameters of Traffic Shapping erupt size (Committed) Bc = 0.01 seconds * CIR = 640 bitsAfter these configuration, we hold made voice calls and hand over get better voice quality. Voice quality have improved because of smaller fragmentation.Question 2 beg off why FIFO queuing should non be used if fragmentation is configured. Fragmentation helps to break large entropy traffic into smaller data traffic. For this voice traffic gets antecedency and have served faster. In the FIFO technology, if any large data entered into the dress thusly in that while if any voice traffic comes, then it ask to be wait until large data traffic finishes. There is no way to prioritize the voice traffic in FIFO techniques.5. point IP precedency of the voice traffic to 5. Generate two ping traffic flows with 3000-byte packet size across PVC1. Make a voice call. Note the voice quality.In the type of service or TOS byte of Header, we have primp IP precedence. IP precedence can identify curriculum of services. Out of seven bits, left tierce digits are use in IP precedence. These values can be from zero to seven. Here larger number means higher antecedence. We have set IP precedence 5 and we make ping traffic with 3000 bytes in the PVC1.We have get voice quality good than the previous quality. side by side(p) table sho ws the configuration between router 1 and router 26. Configure a RTP priority queue for voice traffic. Generate two ping traffic flows with 3000-byte packet size across PVC1. Make a voice call. Note the voice qualityFollowing table shows the configuration between router 1 and router 2We have generated two ping traffic between router 1 and router 2 with 3000 bytes packet size. After that we have test voice calls between our phones. We have get voice quality is good than previous. It has happened because 27 kbps bandwidth is reserve for voice packets and voice packets has no need to wait in the queue.Question 3 visualize the minimum bandwidth required for the RTP priority queue configuration.We have configured voice traffic with RTP priority queue. Our size of voice packet is 66 bytes. So the minimum requirement of bandwidth is 8*66/0.02 or 26,400 bps or 26.4 kbps. We have used G729 codec and voice payload size is 20 bytes. We set our lab bandwidth is 27 kbps.Question 4 Compare the v oice qualities at Steps 4, 5, and 6, and explain the causes of quality differences.To compare voice qualities between go 4,5, and 6, we have found voice quality is worst in smell 4.It has happened for voice call and ping is ready at a clipping, all packets are transfer in the kindred queue. So lots of packet are drop because of more queuing delay.Voice traffic has high priority when we use IP precedence 5 in step 5.Our voice and data traffic still use the same bandwidth. Data traffic still transfer even voice traffic arrives. So ping traffic transfers and voice traffic waits. For this, voice quality is not good because there is no bandwidth reservation for voice traffic.In step 6,we have configured 27 kbps bandwidth in RTP priority queuing. This bandwidth is reserve for voice traffic. So voice packets always use this defined bandwidth and voice traffic has priority than ping traffic. So in this case, voice quality is better.7. Configure three associationes VoIP, VoIP signaling, and slight. Reserve bandwidth 25 kbps for VoIP class, and 8 kbps for VoIP signaling class, respectively. Do not charge priority queue to any class.Three varied classes has been created in this case. They are sequester with fix bandwidth. We have used access-lists for voice traffic classification. Following table shows configuration between router 1 and router 2.Question 5 inform the differences and similarities between CBWFQ and WFQ.CBWFQ can utilize bandwidth effectively compare to WFQ. It is actually the extended version of WFQ. During the congestion period, CBWFQ can tell the minimum bandwidth. It will switch when it gets more bandwidth again. In the CBWFQ, we can define opposite classes and each una manage classes we can assign separate bandwidth. The differences and similarities between WFQ and CBWFQ are as followsSimilaritiesCBWFQ has neglectfulness traffic class but if we do not define this class then CBWFQ and WFQ has no difference in queuing techniques.Difference sBased on the user define classes ,CBWFQ can traffic queuing but WFQ cannot queue traffic. For the traffic flow, CBWFQ can make sure specific bandwidth for it but WFQ can not guarantee that. Network administrator can use CBWFQ more flexibly. They utilize this CBWFQ with different priorities for different types of traffic.Question 6 Based on the above configuration, what are the upper limit and minimum bandwidths that are available for the scorn class?The maximum bandwidth that are available for the default class was 64 kbits/s disrespect class is use all available bandwidth like bandwidth = CIR = 64 kbit/s if there are no voice trafficBecause if there is no voice or voice signaling traffic then the default class is use all available bandwidth (bandwidth = CIR = 64 kbit/s).The minimum bandwidth that are available for the default class is 31 kbit/sBut if for voice or voice signaling flow, the reserving bandwidth will be 25 kbits/s and 8 kbit/s respectively. Other will be available fo r default class. So, minimum bandwidth available for default class is64kbps (25kbps + 8kbps) = 31 kbit/sWe have generated two ping traffic through PVC1 and at the same time we do voice call. Our packet size is 3000 bytes. We have found same voice quality like step 6.8. Establish a voice call between the two phones. At the same time, generate two ping traffic flows with 3000-byte packet size between the two routers. Note the voice quality.We have generated voice calls in our phones. At the same time between router 1 and router 2 we have generated 3000 byte ping traffic. We have found the acceptable voice quality. For voice traffic we have assign different class but there is no priority assign for this class. So, in the receiver end, both data and voice packets are receive similar way. So voice is not be clear because of delay introduce.9. Configure a priority queue for VoIP class using LLQ.We have configured priority queue with LLQ. Following table shows the configuration10. Make a voice call between the two phones. Note the voice quality.After the configuration, we have make phone call with 3000 byte ping traffic in the same time. We found the improved voice quality.Question 7 Explain why the voice quality is improved after the priority configuration.Voice quality has improved for priority configuration. Previously we do not assign priority for different define classes of voice and voice signaling. So, voice quality is improved because of priority assigned. So for any ping traffic comes first it is served first. With the priority queue configuration, it has been resolved. So for any voice traffic arrival, it is use priority queue with allocated bandwidth.ConclusionIn this lab, we have configured two routers and make phone call. We have observed the different call quality. knell quality depends on traffic flow. Traffic flows depends on different parameters like delay, jitter, loss of packets, etc. These hampers the quality of services. We have implemented thr ee queuing compensate here like WFQ,CBWFQ,CBWFQ with LLQ. The objective of this lab is to analyze and improve the quality of voice service. We have found, the best voice quality when we have implemented different class for voice traffic and assign high priority value for voice traffic.Page 1