Voice over IP: improving the quality over wireless LAN by adopting a booster mechanism: an experimental approach
25 Jul 2001-Vol. 4522, pp 157-168
TL;DR: In this paper, the authors describe the design and implementation of a speech property based booster that improves the quality of voice over wireless LANs, which uses characteristics of human speech production and features of modern audio codecs to distinguish packets regarding their importance for perceptual quality.
Abstract: The performance of unreliable voice transmission (Voice over IP) over wireless links is measured not by the throughput but by the perceptual speech quality. The speech quality is impaired by packet losses, which are common on wireless links, and by high transmission delays. In this paper, we describe the design and implementation of a novel Speech Property Based Booster that improves the quality of voice over wireless LANs. This booster is in compliance with existing standards and is transparent to other protocols. It uses characteristics of human speech production and features of modern audio codecs to distinguish packets regarding their importance for perceptual quality. Important packets are protected at the link layer by three mechanisms: selective packet loss recovery, redundant transmission and a hybrid solution. These mechanisms have been evaluated using an experimental set-up with commercial wireless LAN equipment. We made measurements of the objective audio quality and analyzed the effects of packet losses, both due to real wireless channels and late packet arrivals. Our experiments show that the booster increases the quality of voice best with the hybrid solution and that the performance of Voice over IP can be improved further.© (2001) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.
Citations
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01 Jan 2003
TL;DR: This paper investigates the impact of individual packet loss on the perceptual speech quality in Voice-over-IP systems using three popular coding types and receiver-side loss concealment algorithms and defines an appropriate quality metric.
Abstract: If highly compressed multimedia streams are transported over packet networks, losses of individual packets can impair the perceptual quality of the received stream in different degrees, depending on the content and context of the lost packet. In this paper, we investigate the impact of individual packet loss on the perceptual speech quality in Voice-over-IP systems using three popular coding types and receiver-side loss concealment algorithms. We set up a testing environment to measure the impairment of individual packet losses and define an appropriate quality metric. We evaluate published algorithms on packet loss quality prediction (DTX, Source-Driven Packet Marking and SPB-DiffMark) and identify their strengths and weaknesses. The quality of a VoIP telephone call can be enhanced significant, if a precise packet-loss quality model decides for each VoIP packet, how it should be forwarded throughout the network.
41 citations
Cites background from "Voice over IP: improving the qualit..."
...The quality of a VoIP telephone call can be enhanced significant, if a precise packet-loss quality model decides for each VoIP packet, how it should be forwarded throughout the network....
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Patent•
12 Jun 2009
TL;DR: In this article, a machine readable medium and a method are also disclosed for remote communication, where a local communication proxy module may be configured to conceal characteristics of the network from the local communication application module.
Abstract: A communication apparatus for remote communication may include a local communication proxy module configured to receive streams from a local communication application module. The streams may be in a form utilizing a first transmission protocol and destined to a remote destination. The local communication proxy module may be configured to facilitate sending, utilizing a second transmission protocol, the streams in real-time over a network to a remote communication proxy module. The local communication proxy module may be configured to conceal characteristics of the network from the local communication application module. The local communication proxy module may also be configured to be transparent to the local communication application module. A machine-readable medium and a method are also disclosed.
39 citations
Patent•
22 Mar 2010
TL;DR: In this article, the authors present a machine-readable medium and a method for remote communication with a transparent proxy module that intercepts a first stream destined to a remote destination and makes a first determination whether to accelerate communication associated with the first stream.
Abstract: A communication apparatus for remote communication may include a local transparent proxy module configured to intercept a first stream destined to a remote destination and configured to make a first determination whether to accelerate communication associated with the first stream. The communication apparatus may include a local proxy module configured to receive the first stream based on the first determination and configured to make a second determination whether a connection to a remote proxy module is established. If the connection is established, then the local proxy module may receive one or more additional streams and may direct the one or more additional streams to the remote proxy module utilizing an accelerated mode. If the connection is not established, then the local transparent proxy module may direct the first stream to the remote destination utilizing a non-accelerated mode. A machine-readable medium and a method are also disclosed.
35 citations
21 Mar 2004
TL;DR: Simulation results demonstrate that the network performance and the speech quality are substantially improved by modifying the MAC layer with SEC to suit a particular GSM speech compression standard, the narrow-band adaptive multirate (NB-AMR) coder operating at a rate of 7.95 kbps.
Abstract: Mobile ad hoc networks (MANET) have more severe operating conditions than traditional wireless networks. The MAC protocol of IEEE 802.11 mitigates collisions and ensures error-free packet transmissions at the cost of limiting capacity and increasing latency. For voice transmission over MANETs this cost should be minimized. We propose and examine selective error checking (SEC) at the MAC layer of 802.11 that takes advantage of the fact that many of the speech bits can tolerate errors while other bits must be protected for effective reconstruction of the speech. Simulation results demonstrate that the network performance and the speech quality are substantially improved by modifying the MAC layer with SEC to suit a particular GSM speech compression standard, the narrow-band adaptive multirate (NB-AMR) coder operating at a rate of 7.95 kbps.
30 citations
31 Mar 2006
TL;DR: Algorithms to enhance the efficiency of packetized, interactive speech communication over wireless networks are presented, which contain relevant innovations and novel algorithms, which have a high potential to influence future research and product development.
Abstract: This thesis presents algorithms to enhance the efficiency of packetized, interactive speech communication over wireless networks. The results achieved are the following: We present an improved approach to assess the quality of voice transmissions in IP-based communication networks. We combined the ITU E-Model, the ITU PESQ algorithm, and various codec and playout schedulers to analyse VoIP traces. Parts of this algorithm have been included in ITU standards. By using this assessment approach we derived design guidelines for application and data-link protocols. Also, we developed a quality model to parametrise adaptive VoIP applications. Later results received a best-paper award. If highly compressed packetized speech is transported over packet networks, losses of individual packets impair the perceptual quality of the received stream differently, depending on the content and context of the lost packets. We introduce the idea of the Importance of Individual Packets, which is defined by the impact of VoIP packet loss on speech quality. We present real-time and off-line algorithms to measure this importance. Using the concept of importance of packets we show that only a fraction of all speech packets needs to be transmitted if speech intelligibility is to be maintained. By applying this concept for Internet telephony over wireless links, significant transmission energy savings on wireless phones can be achieved, because fewer packets need to be transmitted. At the MAC layer we provided an open-source simulation model of IEEE 802.11e EDCA, which is used in many research projects and is often cited. Also, the ability of WLAN support voice traffic was studied qualitatively and quantitatively. Last, we proved that the distance between two WLAN nodes can be determined by packet round trip time measurements. This approach outperforms the previously used signal strength indications. Overall one can summarize, that this thesis contains relevant innovations and novel algorithms, which have a high potential to influence future research and product development.
25 citations
Cites background from "Voice over IP: improving the qualit..."
...Such enhancements have been studies in our work [84,85,183] and in related literature (refer to Sections 8....
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References
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01 Jul 2003
TL;DR: RTP provides end-to-end network transport functions suitable for applications transmitting real-time data over multicast or unicast network services and is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks.
Abstract: This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.
7,183 citations
TL;DR: The results show that a reliable link-layer protocol that is TCP-aware provides very good performance and it is possible to achieve good performance without splitting the end-to-end connection at the base station.
Abstract: Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly because of congestion. However, networks with wireless and other lossy links also suffer from significant losses due to bit errors and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to end performance in wireless and lossy systems. We compare several schemes designed to improve the performance of TCP in such networks. We classify these schemes into three broad categories: end-to-end protocols, where loss recovery is performed by the sender; link-layer protocols that provide local reliability; and split-connection protocols that break the end-to-end connection into two parts at the base station. We present the results of several experiments performed in both LAN and WAN environments, using throughput and goodput as the metrics for comparison. Our results show that a reliable link-layer protocol that is TCP-aware provides very good performance. Furthermore, it is possible to achieve good performance without splitting the end-to-end connection at the base station. We also demonstrate that selective acknowledgments and explicit loss notifications result in significant performance improvements.
1,325 citations
12 Jun 1994
TL;DR: The authors investigate the performance of four different algorithms for adaptively adjusting the playout delay of audio packets in an interactive packet-audio terminal application, and indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay can achieve a lower rate of lost packets.
Abstract: Recent interest in supporting packet-audio applications over wide area networks has been fueled by the availability of low-cost, toll-quality workstation audio and the demonstration that limited amounts of interactive audio can be supported by today's Internet. In such applications, received audio packets are buffered, and their playout delayed at the destination host in order to compensate for the variable network delays. The authors investigate the performance of four different algorithms for adaptively adjusting the playout delay of audio packets in an interactive packet-audio terminal application, in the face of such varying network delays. They evaluate the playout algorithms using experimentally-obtained delay measurements of audio traffic between several different Internet sites. Their results indicate that an adaptive algorithm which explicitly adjusts to the sharp, spike-like increases in packet delay which were observed in the traces can achieve a lower rate of lost packets for both a given average playout delay and a given maximum buffer size. >
567 citations
"Voice over IP: improving the qualit..." refers background in this paper
...During transmission they are adapted to the current delay conditions [24]....
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TL;DR: Find out more about the ANSI/IEEE C2 National Electrical Safety Code and recommended practice for sizing Lead-Acid Batteries for Stationary Applications.
Abstract: Id Number Title Year Organization Page 1 C2 National Electrical Safety Code(ANSI/IEEE) 2022 IEEE 0 2 16 IEEE Standard for Electrical and Electronic Control Apparatus on Rail Vehicles 2020 IEEE 3 48 Standard Test Procedures and Requirements for Alternating-Current Cable Terminations 2.5 kV Through 765 kV 2020 IEEE 4 308 Standard Criteria for Class 1E Power Systems for Nuclear Power Generating Stations 2020 IEEE 5 389 Recommended Practice for Testing Electronic Transformers and Inductors 2020 IEEE 6 485 Recommended Practice for Sizing Lead-Acid Batteries for Stationary Applications( ANSI/IEEE) 2020 IEEE 7 628 Standard Criteria for the Design, Installation, and Qualification of Raceway Systems for Class 1E Circuits for Nuclear Power Generating Stations 2020 IEEE
391 citations
29 Mar 1998
TL;DR: An analytical formula for the protocol capacity is derived and a distributed algorithm is proposed which enables each station to tune its backoff algorithm at run-time and indicates that the enhanced protocol is very close to the maximum theoretical efficiency.
Abstract: In WLAN the medium access control (MAC) protocol is the main element for determining the efficiency in sharing the limited communication bandwidth of the wireless channel. This paper focuses on the efficiency of the IEEE 802.11 standard for wireless LANs. Specifically, we derive an analytical formula for the protocol capacity. From this analysis we found (i) the theoretical upper bound of the IEEE 802.11 protocol capacity; (ii) that the standard can operate very far from the theoretical limits depending on the network configuration; (iii) that an appropriate tuning of the backoff algorithm can drive the IEEE 802.11 protocol close to its theoretical limits. Hence we propose a distributed algorithm which enables each station to tune its backoff algorithm at run-time. The performances of the IEEE 802.11 protocol, enhanced with our algorithm, are investigated via simulation. The results indicate that the enhanced protocol is very close to the maximum theoretical efficiency.
354 citations