Channelized voice over digital subscriber line

Computer Networks, 2004

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATMs built-in Quality of Service (QoS) mechanisms. IP

This paper presents a baseline test for the listening quality (LQ) using voice over digital subscriber line (VoDSL) access technology by using voice/listening quality(V/LQ) transmission with voice compression while countinously downloading file. The design of an experimental VoDSL network architecture is presented. We identify the efficiency of the LQ based on the following digital subscriber line (DSL) service levels, 640K/640K, 1.5M/384K, and 3.0M/512K for each American National Standards Institute (ANSI) loops. office home office (SOHO) customers. This paper contains results that can be used in evaluating VoDSL solutions that offers multiple voice connections simultaneously with data onto the high speed digital line offered by ADSL line. The results are based on the V/LQ of VoDSL. In Section 2, we present description of the network architecture for V/LQ test setup. In section 3, we have a discussion on the subscriber loop plant noise. Section 4 is the result of the tests performe...

IEEE Communications Magazine, 2004

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VolP. This article summarizes much of what we have learned in this work, and we Kope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VolP network performance.

2008

During the recent 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 telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls.

2008

One of major technical problems that hinder the promotion of voice communication over Asymmetric Digital Subscriber Lines is the unacceptable voice traffic performance in the presence of coexisting data traffic from other applications. In this paper, we study the delay, caused by transmission of long non-real-time data packets over the bottleneck of low upstream bit-rate of Asymmetric Digital Subscriber Line. In order to retain the delay bellow the acceptable limit, we propose using lower values for Maximum Transmission Unit for the traffic deploying the link in addition to the standard IP Quality of Service mechanisms. We investigated this approach through simulations and shown that performance of voice traffic is improved with respect to other data traffic even without increasing upstream bit-rate of the Asymmetric Digital Subscriber Line.

Nowadays, voice transmission over data networks has become of wide spread. However data networks always suffer from packet loss, delay, and delay jitter that greatly affect the quality of the perceived voice quality. Delay jitter can be cured through using playout buffers. However delay can cause further loss for voice packets. Many recovery techniques were developed to recover lost voice packets and are broadly divided to sender-based and receiver-based recovery techniques. Sender-based techniques require more transmission band width and add delay which is undesirable. Receiver-based techniques depend on received packets and voice properties making it attractive as they remedy the sender-based recovery problems. Examples of it are, Silence Substitution, Packet repetition, Waveform Substitution, Sample interpolation, Pitch waveform Replication and Double Sided Pitch waveform Replication. It is noted that the recovery quality increases as the recovery technique complexity increases. ...

While voice-over-IP (VoIP) products have been deployed in the market for over seven years, recent announcements by service providers such as Vonage, AT&T, Sprint and others have created a flurry of activity by consumer equipment manufacturers racing to roll out residential VoIP gateway products. These low-cost devices are usually standalone boxes that provide VoIP functionality for POTS (plain old telephone system) via a broadband modem (usually cable or DSL). They serve as a bridge between the TMD/analog POTS world, and the IP-centric, packet-based world of the Internet.

IEEE Transactions on Multimedia, 2006

The cost savings and novel features associated with Voice over IP (VoIP) are driving its adoption by service providers. Unfortunately, the Internet's best effort service model provides no quality of service guarantees. Because low latency and jitter is the key requirement for supporting high quality interactive conversations, VoIP applications use UDP to transfer data, thereby subjecting themselves to quality degradations caused by packet loss and network failures. In this paper we describe an architecture to improve the performance of such VoIP applications. Two protocols are used for localized packet loss recovery and rapid rerouting in the event of network failures. The protocols are deployed on the nodes of an application-level overlay network and require no changes to the underlying infrastructure. Experimental results indicate that the architecture and protocols can be combined to yield voice quality on par with the PSTN.