Packet aggregation for voice over internet protocol on wireless mesh networks

>Magister Scientiae - MScThis thesis validates that packet aggregation is a viable technique to increase call ca-pacity for Voice over Internet Protocol over wireless mesh networks. Wireless mesh networks are attractive ways to provide voice services to rural communities. Due to the ad-hoc routing nature of mesh networks, packet loss and delay can reduce voice quality.Even on non-mesh networks, voice quality is reduced by high overhead, associated with the transmission of multiple small packets. Packet aggregation techniques are proven to increase VoIP performance and thus can be deployed in wireless mesh networks. Kernel level packet aggregation was initially implemented and tested on a small mesh network of PCs running Linux, and standard baseline vs. aggregation tests were conducted with a realistic voice tra c pro le in hop-to-hop mode. Modi cations of the kernel were then transferred to either end of a nine node 'mesh potato' network and those tests were conducted with only the end nodes modi ed to perform aggregation duties. Packet ag- gregation increased call capacity expectedly, while quality of service was maintained in both instances, and hop-to-hop aggregation outperformed the end-to-end con guration. However, implementing hop-to-hop in a scalable fashion is prohibitive, due to the extensive kernel level debugging that must be done to achieve the call capacity increase.Therefore, end-to-end call capacity increase is an acceptable compromise for eventual scalable deployment of voice over wireless mesh networks

A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications

This article surveys the various techniques adopted for optimising bandwidth for VoIP services over the period 1999-2014. The improvement of bandwidth can be realized through; silence suppression measure of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm; by so doing, the transmission rate during the inactive periods of speech is reduced, and thus, the mean transmission rate can be reduced. A second measure is packet header reduction which defines a process of multiplexing and de-multiplexing packet headers to curb excesses. Voice/ Packet Header compression is considered the most productive of all the techniques, offering a scheme where VoIP packets are compressed from the 40 bytes of size to a smaller byte size of 2 bytes. When combined with aggregation, compression potentially yields a compressed size of up to 1 byte. In either case, bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of enhancing results favourably in Voice over broadband networksComment: 8 pages, 7 figures. ISSN (Print): 1694-0814 | ISSN (Online): 1694-078

VoIP Packet Delay Techniques: A Survey

The continuous development in the field of communication have paved the way for Voice over Internet Protocol (VoIP). VoIP is a group of hardware and software that facilitates people to utilize the Internet as the transmission medium for telephone calls by transmitting voice data in packets using IP instead of using conventional circuit transmissions of the Public Switched Telephone Network (PSTN). At present, VoIP is becoming an important tool for quick communication across the world. There are several Internet telephony applications existing at present. The major disadvantage in VoIP is that the packet delay. In VoIP, the terminology jitter is used to refer the type of packet delay where the delay has a huge setback in the quality of the voice conversation. Several packet delay techniques were proposed in recent years. Some of the important packet delay techniques are discussed in the literature. This survey would definitely help the researchers to carry out their research for providing better communication in VoIP without any delay

Call capacity for voice over Internet Protocol on wireless mesh networks

This paper describes work in progress on call capacity optimization for voice over Internet Protocol on wireless mesh networks. In a developing country such as South Africa, evidence has shown that rural inhabitants find it difficult to afford the voice services offered by cellular networks. Voice over Internet Protocol is known for its affordability relative to cellular voice services, therefore deploying such services for rural communities will not only benefit rural inhabitants but also offer economic advantages to service providers. We are interested in the provision of voice services with rural wireless mesh networks. Unfortunately voice on mesh networks can experience packet loss and delays that cause reduction in voice quality. Transmission of small voice packets over wireless mesh networks imposes high overhead that leads to a tremendous decrease in call capacity. Therefore, we aim to study the performance of voice over 802.11 wireless mesh networks and evaluate packet aggregation mechanisms that merge small voice packets into a single large packet, in order to preserve voice quality with more calls. We will implement and evaluate packet aggregations mechanisms on a 'mesh potato' network with iterative cycles of laboratory experiments using a network simulator to collect data for performance evaluation.Telkom, Cisco, THRIPDepartment of HE and Training approved lis

A study of mobile VoIP performance in wireless broadband networks

Voice service is to date still the killer mobile service and the main source for operator revenue for years to come. Additionally, voice service will evolve from circuit switched technologies towards packet based Voice over IP (VoIP). However, using VoIP over wireless networks different from 3GPP cellular technologies makes it also a disruptive technology in the traditional telecommunication sector. The focus of this dissertation is on determining mobile VoIP performance in different wireless broadband systems with current state of the art networks, as well as the potential disruption to cellular operators when mobile VoIP is deployed over different access networks. The research method is based on an empirical model. The model and experiments are well documented and based on industry standards for voice quality evaluation. The evaluation provides results from both experiments in a controlled laboratory setup as well as from live scenarios. The research scope is first, evaluate each network technology independently; second, investigate vertical handover mobility cases; third, determine other aspects directly affecting end user experience (e.g., call setup delay and battery lifetime). The main contribution of this work is a systematic examination of mobile VoIP performance and end user experience. The research results point out the main challenges for achieving call toll quality, and how derive the required changes and technological performance roadmap for improved VoIP service. That is, investigate how the performance and usability of mobile VoIP can eventually be improved to be a suitable substitute for circuit switched voice. In addition, we evaluate the potential disruption to cellular operators that mobile VoIP brings when deployed over other access networks. This research extends the available knowledge from simulations and provides an insight into actual end user experience, as well as the challenges of using embedded clients in handheld devices. In addition, we find several issues that are not visible or accounted for in simulations in regard to network parameters, required retransmissions and decreased battery lifetime. The conclusion is that although the network performance of several wireless networks is good enough for near toll quality voice in static scenarios, there are still a number of problems which make it currently unfeasible to use as a primary voice service. Moreover, under mobility scenarios performance is degraded. Finally, there are other issues apart from network performance such as energy consumption, hardware limitations and lack of supporting business models (e.g., for WiFi mesh) that further limit the possibility of rolling out mobile VoIP services

Packet aggregation for voice over internet protocol on wireless mesh networks

>Magister Scientiae - MScThis thesis validates that packet aggregation is a viable technique to increase call capacity for Voice over Internet Protocol over wireless mesh networks. Wireless mesh networks are attractive ways to provide voice services to rural communities. Due to the ad-hoc routing nature of mesh networks, packet loss and delay can reduce voice quality. Even on non-mesh networks, voice quality is reduced by high overhead, associated with the transmission of multiple small packets. Packet aggregation techniques are proven to increase VoIP performance and thus can be deployed in wireless mesh networks. Kernel level packet aggregation was initially implemented and tested on a small mesh network of PCs running Linux, and standard baseline vs. aggregation tests were conducted with a realistic voice traffic profile in hop-to-hop mode. Modifications of the kernel were then transferred to either end of a nine node 'mesh potato' network and those tests were conducted with only the end nodes modified to perform aggregation duties. Packet aggregation increased call capacity expectedly, while quality of service was maintained in both instances, and hop-to-hop aggregation outperformed the end-to-end configuration 4:1. However, implementing hop-to-hop in a scalable fashion is prohibitive, due to the extensive kernel level debugging that must be done to achieve the call capacity increase. Therefore, end-to-end call capacity increase is an acceptable compromise for eventual scalable deployment of voice over wireless mesh networks