Quality of Service Management for Voice Over IP Networks

1. Quality of Service Management for Voice Over IP Networks Team Members: Prashant Anantha Krishnan Sunil Kumar Derasriya

2. Objectives and Goals • To analyze and study the parameters that affect the Quality of Service in Voice Over IP. • Using a testing tool to estimate these parameters under all possible network connections. • Calculate a Mean Opinion Score(MOS) based on an algorithm.

3. Challenges and Difficulties • VoIP is a very new and sophisticated concept so a lot of study had to be done to understand these concepts • Finding a testing tool or a simulation tool that would help us estimating the parameters, which are required to calculate the Mean Opinion Score(MOS).

4. What is VoIP • It is a technology for transmitting voice calls over the Internet using packet linked routes. Also known as IP telephony. • It enables the people to use the Internet as a transmission medium for sending voice data in packets using IP rather than using traditional circuit transmission of the PSTN.

5. Gateways allow PCs to also reach phones Public Switched Telephone Network Initially, PC to PC voice calls over the Internet Gateway Multimedia PC Multimedia PC IP Network Gateway PSTN (NY) PSTN (DC) …or phones to reach phones Contd……

6. Advantages of VoIP • Greater Efficiency • Lower Cost • Higher Reliability • Supporting Innovation

7. Quality of Service • The ultimate objective of VoIP is reliable, high-quality voice service, the kind that users expect from the PSTN. • It is hard to achieve the same level of QoS as in PSTN. The main QoS issues are speech quality, service availability and usability. Voice requires lower delay, jitter and packet loss where as Ordinary Data transfer can be delayed without affecting much to the client’s requirement. • To withstand to such needs a minimum level of QoS mechanism must be maintained.

8. Quality of Service Parameters • Delay : The amount of time taken by a packet to reach from the source to the destination. • Issues with Delay • Echo • Talk Overlap • Types of Delay in a VoIP Call: • Processing Delay • Network Delay

9. Jitter • Jitter is the variation in the time between packets arriving, caused by network congestion or route changes. • Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence which causes additional delay.

10. Packet Loss • Packet Loss is losing packets along the data path which further degrades the VoIP Applications. • Voice packets are time-sensitive unlike Data packets. Therefore, retransmission is not a solution to this problem.

11. Latency • When the packet is being sent, there is a “latent” time till the computer that sent the packet waits for a confirmation that the packet has been received. • Latency causes packets to be lost. If a packet does not arrive in time to be replayed at the receiving end, the packet is dropped. • Latency does not distort the voice signal but delay can be very annoying, making normal conversation difficult for the speakers. The parties may start to talk at the same time or interrupt each other. As a result, the conversational quality is perceived as being poor.

12. Solutions for QoS Issues • For Voice communications over IP to become acceptable to the user, the delay needs to be less than a threshold value. • To ensure good quality of service, we can use Echo Cancellation and Packet Prioritization. • Use of service quality models that gives an estimate of perceptual quality rating using the networking parameters. Mean Opinion Score (MOS) is one of the quality rating on a scale of 1 (bad) to 5 (excellent)

13. Perceptual Assessment Model • PESQ is a model for perceptual evaluation of speech quality. • One novel feature of PESQ is the identification of transmission delays. • First PESQ adjusts the degraded version to be time aligned. Then it assesses the distortion between original and degraded sample. • Constant delays are not considered in the calculation of the MOS value, but delay variations change the rating of the speech quality. One should note that PESQ can only be applied for distortions which have been known before its development. • In PESQ the original and the degraded signals are mapped onto an internal representation using a perceptual model. The difference in this representation is used by a cognitive model to predict the perceived speech quality of the degraded signal. This perceived listening quality is expressed in terms of a mean opinion score (MOS), an average quality score over a large set of subjects.

14. Perceptual Evaluation Of Speech Quality Model

15. E-Model • “Mouth to ear” transmission quality measurement • Produces an “R” factor typically in the range 50 (bad) -95 (good) • R factor can be related to MOS score, Terminate Early (TME) etc. • ITU G.107/ G.108 and ETSI ETR250

16. Contd… R = Ro - Is - Id - Ie + A Base R value - Noise level Advantage factor Equipment Impairment Factor - CODEC - multiplexing effects Impairments that occur simultaneously with speech - received speech level - sidetone level - quantization noise Impairments that are delayed with respect to speech - talker echo - listener echo - round trip delay

17. Contd.. Packet Loss Loss Model Ie R Factor Jitter Model E Model Jitter Codec type Codec Model Delay, measured using RTCP

18. R Factor vs MOS R Factor MOS 4.5 4.0 3.0

19. How does Rider Works? • Rider is an entry-level network performance measure program that measures the network response time, bandwidth, and Voice Over IP parameters between any two computers on your network. • There are four basic tests performed by Rider when sending test data between pairs of computers: • Bandwidth testing.   How long will it take to copy a big file across the network? • Response time testing.   How long will it take for a packet of data to travel from one end of the network and back? • Voice over IP testing.   If you were to use a new VoIP phone, how good or bad would the packet loss and jitter be?  Dropped packets hurt the sound quality.  Jitter refers to the variation in packet arrival time.  Packets that arrive too late or out of order (yes, this happens) can't be used.  • Stream testing.  This is just the general case of Voice over IP testing.  If we wanted to run a movie stream, or some other application, would it work?

20. How to Calculate MOS? • For the codec used pick a corresponding default R value.  The R-values for the most popular codec has been used. These are R=93 for the G.711 codec, R=80 for the G.729a codec and R=86 for iLBC codec. • From the Rider streaming test we can calculate the Jitter and Packet Loss. • From the Rider response time test, we can measure the network latency between the control and remote locations. • Add 10 ms if you are using G.729a and 5 ms for iLBC codec to account for computation time. • Add step no. 2, 3 and 4 to calculate the effective latency. (Latency plus jitter plus computation time.) • Adjust the R value down based on effective latency.  Deduct 5 for a delay of 150 ms, 20 for a delay of 250 ms, 30 for a delay of 350 ms.

21. Contd.. • From the packet loss test in step no. 2, deduct the R value from consecutive packet losses using the table given below.

22. DEMONSTRATION

23. Conclusion At last, we conclude that Mean Opinion Score is one of the better and reliable ways to estimate the quality of service for a VoIP Network. The future Implementations of our project is to bind our application with a simulation tool.