Codec Selection Research Articles

Analysis of Video CODEC Performance Using Different Softphone Applications

In a video call there are several components, such as IP phone or softphone, CODEC, and server. Selection of softphone and CODEC is a consideration in building a video communication network because it will affect the quality of video call. This research compare the quality of video calls based on softphone application and CODEC combination. The quality measured by QoS, PSNR, and MOS parameters.Softphone applications examined in this research are Blink, Zoiper, MicroSIP, PortGo, Linphone, and X-Lite. CODEC examined in this research are H.264, VP8, H.263+, and H.263. Each softphone application will be combined with a CODEC that is native to the softphone. There are nine combinations of softphone application and CODEC.Based on the research results, CODEC H.264 has the best performance when paired with the Blink softphone application. CODEC VP8 has the best performance when paired with the Zoiper softphone application. The H.263+ CODEC has the best performance when paired with the PortGo softphone application. The H.263 CODEC and X-Lite softphone applications have the worst test results but still get “good” grades when tested using QoS, PSNR, and MOS parameters.

SEAMLESS VOIP SERVICE USING CENTRALIZED CODEC ADJUSTMENT: A FRAMEWORK IN WIRELESS NETWORK

The evolution of wireless technology leads VoIP application to be a good preferred choice among mobile users. Customers can having a VoIP service while being mobile regardless of location, terminal, and it is independent of network access. It is believed to be an alternative to the PSTN and its service is expected to be as good as PSTN. In essence, QoS plays a crucial part in order to maintain the seamless service during the VoIP session. As wireless networks are prone to channel error and VoIP application is known to be a delay-sensitive application, network traffic congestion could easily affect the voice quality of VoIP. Therefore, the codec selection within the VoIP application must be improved so that the codec can operate at various bit rates in order to cater for different traffic conditions. In this paper, we explain how centralize codec switching can help user to communicate seamlessly. We use WiFi network traffic as an example scenario.

Influence of speech codecs selection on transcoding steganography

The typical approach to steganography is to compress the covert data in order to limit its size, which is reasonable in the context of a limited steganographic bandwidth. Trancoding steganography (TranSteg) is a new IP telephony steganographic method that was recently proposed that offers high steganographic bandwidth while retaining good voice quality. In TranSteg, compression of the overt data is used to make space for the steganogram. In this paper we focus on analyzing the influence of the selection of speech codecs on hidden transmission performance, that is, which codecs would be the most advantageous ones for TranSteg. Therefore, by considering the codecs which are currently most popular for IP telephony we aim to find out which codecs should be chosen for transcoding to minimize the negative influence on voice quality while maximizing the obtained steganographic bandwidth.

Non-intrusive speech quality assessment using several combinations of auditory features

Quality estimation of speech is essential for monitoring and maintenance of the quality of service at different nodes of modern telecommunication networks. It is also required in the selection of codecs in speech communication systems. There is no requirement of the original clean speech signal as a reference in non-intrusive speech quality evaluation, and thus it is of importance in evaluating the quality of speech at any node of the communication network. In this paper, non-intrusive speech quality assessment of narrowband speech is done by Gaussian Mixture Model (GMM) training using several combinations of auditory perception and speech production features, which include principal components of Lyon's auditory model features, MFCC, LSF and their first and second differences. Results are obtained and compared for several combinations of auditory features for three sets of databases. The results are also compared with ITU-T Recommendation P.563 for non-intrusive speech quality assessment. It is found that many combinations of these feature sets outperform the ITU-T P.563 Recommendation under the test conditions.

A comparative analysis of video codecs for multihop wireless video sensor networks

Wireless video sensor networks (WVSNs) have drawn significant attention in recent years due to the advent of low-cost miniaturized cameras, which makes it feasible to realize large-scale WVSNs for a variety of applications including security surveillance, environmental tracking, and health monitoring. However, the conventional video coding paradigms are not suitable for WVSNs due to resource constraints such as limited computation power, battery energy, and network bandwidth. In this paper, we evaluated and analyzed the performance of video codecs based on emerging video coding paradigms such as distributed video coding and distributed compressive video sensing for multihop WVSNs. The main objective of this work was to provide an insight into the computational (encoding/decoding) complexity, energy consumption, node and network lifetime, processing and memory requirements, and the quality of reconstruction of these video codecs. Based on the findings, this paper also provides some guidelines for the selection of appropriate video codecs for a given WVSN application.

Mastering the Integrity of Technical Information in Tapeless Production and Long-Term Archives

The main challenge for information technology teams is to ensure that the media created today will not become obsolete tomorrow. Standards must remain high, while all types of information continue to be accessible. In workflow architecture, key decisions revolve around choosing the right audio/video codecs to guarantee maximum interoperability. However, although various layers such as vertical ancillary (VANC), vertical blanking interval (VBI), SMPTE 328M, and video auxiliary (VAUX) contain strategic data (time codes, multichannel audio descriptions, closed captioning, aspect ratio, etc.), they are not sufficiently taken into account in operation processes. This valuable information often comes embedded in the input source video stream and must be directed toward the Material Exchange Format (MXF) essences to ensure a genuine integrity of metadata in a multilayered bark. This document provides practical case studies on codec selection, time code and subtitle carriage, as well as the number of audio channels, including those encoded in Dolby-E.

Proposal for Robust and Adaptive Forward Error Correction for real-time Audio-Video streaming solutions

In this paper, we present a robust and adaptive forward error correction technique for high end Audio-Video Streaming solutions. This solution targets all kinds of network loss conditions and can be used in any embedded platforms. This technique uses optimum solution for packet loss recovery that involves switching between adaptive FEC based packet loss recovery and retransmission management. Whenever network loss increases the permissible limit of adaptive FEC loss recovery, then it switches to re-transmission of media. Adaptive FEC based packet loss recovery works on best of adaptive matrix size selection, loss recovery method selection (on-the-fly or store and compute) and FEC codec selection (Parity 1D, 2D, Raptor, Read Solomon and Hamming etc.) This technique also incorporates support for embedded systems which do not have much computation power. For such embedded systems, the adaptive FEC recovery technique switches to fixed matrix of having store and save approach for single and burst packet loss recovery. This technique provides inter-operability w.r.t FEC codec selection which also takes care of embedded system capability. Keywords - Adaptive FEC, Adaptive matrix size, FEC coding (Parity 1D, 2D, Raptor, read-soloman, hamming etc.), , FEC SDP negotiation, packet loss calculation techniques

Best Approach for Video Codec Selection Over VoIP Conversation Using Wireless Local Area Network

This study evaluates video codec performance over VoIP using a campus wireless network. Today, the deployment of VoIP occurs in various platforms, including VoIP over LAN, VoIP over WAN and VoIP over VPN. Therefore, this study defines which video codec provides good video quality over VoIP transmission. The soft phone is used as a medium for communication between two parties. A network management system is used to evaluate and capture the video quality performance over VoIP. The quality of video codec is based on MOS, jitter, delay and packet loss. The experimental scope is limited to G.722 with MP4V-ES, G.726 (16) with H.261 and G.726 (24) with H.264. The results show that audio codec G.722 with MP4V-ES generates good video quality over VoIP using wireless local area network. Whereas audio codec G.726 (16) with H.261 generates low rate video and voice quality performance. Therefore, using the appropriate video and audio, the codec selection increases video quality over VoIP transmission.

Voice transmission over ad hoc network adapting optimum approaches to maximize the performance

Although a lot of research and implementations have been performed for voice transmission over regular network and internet, a little is done for mobile ad hoc network (MANET). Like other computer networks, in MANET voice transmission is also very much demanding and necessary. Since the wireless links in an ad hoc network are highly error prone and may go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure, it is very difficult and challenging to implement voice transmission over MANET. In this research, to maximize the performance of MANET during voice transmission we choose some parameters and methodological approaches e.g. Method of media access, selection of audio codec, selection of routing protocol etc in efficient and optimum way. Since voice applications consume more energy than typical applications, we use an energy aware routing protocol known as WEAC for the study. During the selection process of routing protocol we have a performance comparison study between Improved DSR and WEAC, where WEAC performs better consuming less energy in case of larger network. We also have a comparative study among several audio codecs (G.711, G.729 and G.723.1) where by simulation we show that the G.729 codec is more suitable to use for voice transmission over MANET in terms of latency. Finally, we show that it is possible to launch voice transmission with acceptable quality and throughput over MANET using G.729 and WEAC protocol.

A transmission capacity enhancement of the CDM‐based satellite DMB system

AbstractAfter the launch of Satellite Digital Multimedia Broadcasting (S‐DMB) service in Korea, there was a requirement to have more channels to accommodate more bandwidth‐requiring applications. We designed a new architecture to provide the system with more channel capacity while maintaining the current performance as much as possible. We took a hierarchical modulation technology used for the DVB‐S2 standard. A number of combinations of modulation parameters and FEC modes have been simulated for the evaluation of technical feasibility of the proposed structure. Also investigations were carried out to assess the impact to current service performance. It was found that a 16200‐bit LDPC with hierarchical modulation could give additional channel capacity while keeping the current quality of service for selected CODEC and code rates (Int. J. Satellite Commun. Networks 2004; 22). Though 1ldB degradation of BER to the existing service was resulted by this approach, but no perceptional service degradation was expected since current link margin is larger than 3ldB. We found that about 36% of capacity enhancement can be achievable by the hierarchical modulation with proper CODEC selection. Additional degradation, however, may be encountered in mobile environment due to the multipath channel in spite of its performance immunity to the Doppler effect. Copyright © 2008 John Wiley & Sons, Ltd.

How to tune VoIP codec selection in WLANs?

In IEEE 802.11e-based WLANs, link adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bit-rates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. A VoIP codec selection algorithm has been proposed as a solution to this issue, which is triggered both by channel rate changes as well as in combination with a call admission control mechanism. The results show that an important improvement in terms of hotspot capacity for VoIP calls can be achieved by choosing the VoIP codec adaptively in a multi-rate scenario. By defining a new grade of service-related parameter, Qmacr, which captures the tradeoff between dropping and blocking probabilities and perceived speech quality, the codec selection algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements.

Objective Assessment of Speech and Audio Quality—Technology and Applications

In the past few years, objective quality assessment models have become increasingly used for assessing or monitoring speech and audio quality. By measuring perceived quality on an easily-understood subjective scale, such as listening quality (excellent, good, fair, poor, bad), these methods provide a quick and repeatable way to estimate customer experience. Typical applications include audio quality evaluation, selection of codecs or other equipment, and measuring the quality of telephone networks. To introduce this special issue, this paper provides an overview of the field, outlining the main approaches to intrusive, nonintrusive and parametric models and discussing some of their limitations and areas of future work

Using Movie Clips in PowerPoint Presentations: Part 1, Compatibility Issues

The purpose of this article is to discuss common compatibility issues relating to the use of movie clips in PowerPoint presentations. Our investigations found that successful display of movie clips in PowerPoint depends on the following three factors: movie file format, codec selection, and cross-platform compatibility. These factors primarily result from the rapid change in today's computer operating systems. Based on a compatibility study of common movie clips used in radiology presentations, a general guideline for preparing movie clips is presented. Also, a simple method is described for converting movie clips into compatible movie formats and reducing large movie clips to manageable file sizes to ensure smooth presentation.

A framework for SIP session policies

Proxy servers are important network elements that play some key roles in the establishment of sessions in the Session Initiation Protocol (SIP). In particular, they serve to register the contact address (the registrar) and provide call rendezvous and routing, among other services. Their requirements can have a strong impact on policies on the signaling level. However, there is no standard means by which network elements can influence policies on the session level (i.e., policies affecting the sessions and media streams). Examples of session policies include restricting the set of allowed codecs (e.g., to prevent the use of high-bandwidth codecs in wireless networks) or routing media streams through an intermediary (e.g., a firewall). Today, non-standard techniques are deployed to realize session policies. In this paper, we propose a standards compliant framework that enables network elements to request session policies from user agents. We discuss the key aspects of the framework design and present a prototype implementation for codec selection during the establishment of a session.

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls.