Deployment plan for a VoIP solution

Deployment plan for a VoIP solution





Problem statement

There have been problems with quality of service offered by Airtel communication Company through voice over internet protocol (VoIP). The service quality of voice communication over this network has continued to deteriorate as the traffic also continues to vary at different times. This service quality compromise has also been attributed to related issues of performance metrics for different codecs employed within the system of VoIP in the communication course. Some of the performance metrics include the average throughput, end to end delay, mean opinion score, average delay and average jitter.

The quantity of data received after every second over the network has reduced considerably, signifying lower average throughput that compromises the quality of service over the Airtel VoIP network. More time is also being taken to deliver voice messages from the source to destination point of Airtel VoIP network. This more time increases end to end delay time which in turn lowers the quality of service. The low service quality can also be attributed to mean opinion score that is poor or below average. The numerical values used in the case of mean opinion score ranges from 1 to 5 with 1 being the poorest while 5 being the highest. The high average delay also tend to suggest more time taken by the VoIP network in the relay of data bits to the receive from the respective sender. This high average delay is also a contributing factor towards the poor quality of service across the communication network. The quality of service is also affected by jitter parameters in communication. The normal average jitter expected for normal communication rages between 20ms to 30ms. Therefore the presences of poor quality service imply that the amount of average jitter is outside this rage. It is changes in the time amid transmitted packets which reaches the destination which in turn comes due to the congestion of the network, drift in time and/or changes in the respective routes taken by the information packets.

The purpose of this project is to ensure that the overall quality of service in the VoIP network of Airtel is enhanced by using the best means possible to harmonize the above mentioned performance metrics that affects quality of performance or service. This will be achieved by carrying out a number of project activities. The activities in the process will involve the investigation of the existing VoIP system, requirements/information gathering about the existing system in relation to the expected system, feasibility study, analysis of the requirements, designing of the new or expected VoIP system and finally implementation of the system. Therefore, the following objectives will guarantee that the result of implementation delivers the expected VoIP system.

The activities are to ensure that all the relevant data requirements are collected that reveals the weaknesses of the current system.

The process should also ensure that the precise feasibility study is done that can prove the cost effectiveness of implementing the new system.

The activities are also to ensure that all the requirements are accurately analyzed in order to come up with the exact requirement specifications.

The process is to ensure that the design of the system is done to reflect all the specifications requirements.

The activities are also to ensure that the implementation of the final system reflects the design specification that will be significant in achieving the purpose/goal of VoIP system which is high quality of service.

In order to ensure that the expected system has been achieved, some kind of criteria will be employed to measure the result. Some of the criteria will comprise of response time of the total VoIP system in communication, efficiency of the system, reliability of the system in terms of data transmission errors and the network throughput. Other criteria will also include delay time in transmission which comprise of average delay time as well as source terminal to end terminal delay time.


The main problem to be solved by this project is the poor service quality offered by the current VoIP system. This is determined by the types of voice codecs used in the VoIP system. The main function of the voice codecs is to convert the analogue patterns of voice that are in the form of waves at the subscriber or source terminal in to digital pulses as well as to reconvert the digital pulses at the receiving terminal in to the previous analogue equivalent of voice wave patterns. The whole of this conversion process and the transmission in between is what contributes to various performance metrics that compromises the quality of service at different circumstances or scenarios (Mohamed, Zaki & Elfeki, 2012).

There are many types of codecs used for signal conversion between analogue and digital within the VoIP system. The causes of the problems witnessed in the current system can be discussed in the context of the strengths and weaknesses of each of these codecs. The lists of codecs which can be considered in this case include G.711, G.729, G.726, GSM speech codec among others. These codecs can be open source, proprietary or patented one. For instance, G.11 and G.729 are patented codecs while G.729 can be patented or proprietary (Gurung & Singh, 2013, Rattal, Badri & Moughit, 2013).

Some of the causes of low service quality problem include average delay, average jitter, mean opinion score, end to end delay and average throughput which are all used as performance metrics for open source, patented and proprietary codecs. These performance metrics are also looked at in relation to other parameters such as bit rate in kilobits per second(Kbps) also called bandwidth requirements, sample size measured in bytes, number of packets received per second, and payload size in bytes (Gurung & Singh, 2013).

The G.11 codec has a bandwidth requirement of 64 kbps, sample size of 80 bytes transfer speed of 50 packets per second and payload of 160 bytes. The patented G.729 has a bandwidth requirement of 8kbps, sample size of 10 bytes, with the same packet transfer speed as that of G.11 codec. The first version of proprietary G.729 has a bandwidth requirement of 6.3kbps, sample size of 24 bytes, packet transfer speed of 33.3 packets per second with a payload size of 24. The second version of proprietary G.729 has a bandwidth requirement of 5.3 kbps, sample size of 20bytes, packet transfer speed similar to that of the first proprietary version of G.729 with the payload of 20 bytes. The open source G.726 has a bandwidth requirement of 24 kbps, sample size of 15 bytes, packet transfer speed of 50 packets per second with the payload of 60 bytes (Gurung & Singh, 2013).

Let us consider, for example, the performance analysis of various codec for voice over internet protocol with vehicular ad hoc network that is planned with universal mobile telecommunication system and with the help of H.322 protocol in distinct scenarios such as city or high way surroundings with changing situations of traffic. In this case, research from dependable sources shows that G.711 can perform best in both sparse and city condition in terms of throughput. However, it is only the performance of G.726ar32 codec that becomes high as the network becomes busy with very high traffic (Gurung & Singh, 2013).

In considering the metric of end to end delay, G.711 can illustrate the maximum performance in city and highway sparse situations in addition to highway high traffic conditions of the network. The G.723.1ar5.3 has less delay on average in both the city and highway conditions of high network traffic. The G.723.1ar5.3 and G.728ar16 codecs also illustrates varying performance with reference to mean opinion score numerical values. The G.723.1ar5.3 alone in the sparse highway network offers satisfactory quality. On the other hand, every codec has poor quality in the high traffic dense network. In the sparse network of city areas, G.728ar16 has good quality while G.723.1ar5.3 illustrate better mean opinion score numerical value in dense traffic network as compared to other types of codec. The G.7231ar5.3 is appropriate for jitter in both low and dense scenarios of the highway while G.726ar32 and G.711 are suitable for city sparse and city dense scenarios respectively for jitter performance metric (Gurung & Singh, 2013, Rattal, Badri & Moughit, 2013).

In summary, the codecs G.723.1ar5.3 has better performance in all city scenarios and in sparse highway scenario in terms of average delay with G.723.1ar having better performance in dense highway scenario for the same average delay metric. The G.711 has better performance in sparse situations of both city and highway scenarios for the average throughput. The codecs G.726ar32 on the other hand has better average throughput performance in the dense high traffic network of both city and highway scenarios. The average jitter performance is such that the G.726ar24 performs best in both scenarios of highway traffic. The G.726ar32 has best performance of average jitter in the sparse city scenario with G.11 codec having the best performance of average jitter in the dense city scenario (Gurung & Singh, 2013).

The performance of hybrid codecs G.711 and G.729 can also be evaluated with the use of both SIP and H.323 protocols. If the computation of quality of service performance by means of Opnet is carried out under the same metrics such as jitter, average delay and others shows clearly that, it will be apparent that the most suitable codec combination and signaling protocol is the H.323 and G.729. This mixture of codec and protocol provides the lowest delay with a satisfactory value of jitter for the best quality of call across the Airtel VoIP communication network. It should also be noted that the quality of rating for voice quality varies against different numerical values of the mean opinion score. For instance, the best quality of voice is rated at numerical values ranging from 4.34 and 4.5 while the poorest voice quality at the numerical values ranging from 4.58 and 3.0. High quality of voice is rated between 4.03 and 4.34 numerical values, medium quality between 3.6 and 4.03 while the low voice quality at 3.1 and 3.6 numerical values of mean opinion score (Rattal, Badri & Moughit, 2013).

The codecs having high level of compression are also characterized with les bandwidth requirements and thus have better performance in the situations of high congestion for the network. It is therefore advisable to choose the most appropriate codec for the purpose of getting the best quality of voice with the smallest requirement of bandwidth. The G.711 codec does not use any compression and has a sampling rate of 8 kHz. The G.711 codec requires an audio bandwidth of 64 Kbps and offers extremely good quality level of performance in terms of service quality. The G.729 codec has high complexity of computation but nevertheless offers significant bandwidth savings. It has a ratio of compression of 8 to 1 and requires an audio bandwidth of 8kbps (Rattal, Badri & Moughit, 2013).

In comparing for instance, G.711 and G.729 codecs, the G.711 has a higher bit rate of 64 kbps as compared to G.729 which has low bit rate of 8 Kbps. The G.711 better link utilization of 87.2 Kbps with G.729 having lower link utilization of 31.2 Kbps. The G.711 and G.729 codecs also have delays of 0.125 ms and 15ms respectively. This implies that G.711 has better performance since it has lower delay of packet transmission. However, the loss of packet transmission is higher in G.711 as compared with G.729. This loss ranges between 7 percent and 10 percent for G.711 and less that 2 percent in G.729 (Rattal, Badri & Moughit, 2013).

The performance parameter thus affects the quality of VOIP call at different levels. The end to end delay together with the entire probability of loss impacts the call quality in the voice over internet protocol, which is the R-score. The delay effect in the system of VOIP, the mouth to ear delay comprises of components such as codec delay network delay in addition to play out delay. The codec delay illustrates both packetization and the algorithm delay connected to the codec and which changes from one type of codec to the other. The G729.a codec for example, sets up a delay for the steams of data packets of voice that arrives at the receiving destination. The network delay on the other hand is the one way delay passage through the internet protocol network transport that takes place from one internet gateway to the a different gateway that links two or more dissimilar types of networks. This means that the total delay becomes the sum total of codec delay payout delay and the network delay. The delay parameter becomes the transmission impairments that rely on one way delay of mouth to ear. This also determines the voice communication interaction. The effect of this delay on the quality of voice relies on critical value of time (177.3ms). This time value forms the total delay budget of delay for streams of VOIP across the communication network. Every metric parameter thus behaves differently in affecting the quality of service in the transmission of VOIP streams. As a result, the remedy or solution against each transmission impairment parameter call for different solution or strategy to deal with the problem in an attempt to compensate for the poor service quality that may be brought about by the combination of different transmission impairments (Mohamed, Zaki & Elfeki, 2012).

Recommended Solution and Implementation

Recommended solution

It is clear from the analysis above that each and every codec used in the system of VOIP has at least one weakness and strength. In other words, there is no completely perfect type of codec can cam perform satisfactorily in all the scenarios and conditions of traffic. Because of this a combination of two or more codecs should be used as the solution measure by the Airtel communication company with an aim after taking advantage of every coded while at the same time compensating for the limitation of one codec with the strength of the other.

As an example for the solution, a combination of G.711, G.726ar32, G.729, G.726ar24, G.723.1ar5.3 and G.723.1ar can solve the problem by eliminating the negative effects of low average throughput. High average delay time, low mean opinion score high average jitters and other unfavorable effects of other metrics. From the analysis, it is apparent that the employment of G.11 and G.726ar32 will solve the entire problem to with low average throughput in all scenarios of both sparse and dense network situations. The addition of G.726ar24 to the list will also ensure that the high effect of jitter which is the result in variations of packet delivery is solved. Adding G.723.1ar5.3 and G.723.1ar ensures that the effects of high delay is either eliminated completely or reduced to an acceptable level. Nevertheless, the combination of these codecs will be based on the prevailing causes. This is because not all the transmission impairments to with the metrics in this case may cause the poor service quality. Therefore, the exact and specific number of performance metrics responsible for the causes of the problem has to be known in order to guide on what type of codecs to be combined for the design of the new solution


The process of implementation will comprise of several activities aimed at finding out the exact causes of the poor service quality. This will begin with the investigation of the current Airtel VOPI system within some period of time. This will be followed by the r5equirements determination which consists of defining the problem, feasibility study, requirements acquisition and requirements analysis. The feasibility part of it will involve the economic operational and technical feasibilities. The requirements acquisition will comprise the use of methodologies like scenarios, questionnaires and even interviews where possible. The requirements determination phase is to be followed by the specification phase which will involve the use of requirements specification from the previous analysis stage to specify precisely what the fresh VOIP system is needed to carry out.

The design phase will then follow where by the specifications from the previous phase will be used to come up with the design models which transforms the the analysis information model into structures that are required for the implementation of the new system of VOIP. The design will be used as the architectural framework to represent the fresh system of communication. Therefore the design in total is to address the system architecture, the system structure, interface and its components. The common activities will include approaches like structured system analysis and design method and the rest, modular decomposition and others.

The implementation stage is to comprise of translating the comprehensive VOIP system design in to code by means of by using the most suitable programming language. This phase will thus comprise of the choice of programming languages and tools to be used, the employment of good practices of programming in addition to documentation. Next is the integration and testing of the new VOIP system where the testing will be done at various stages.