TL;DR: A realisation of adaptive FEC subdued to a TCP-friendly rate control is described, to ensure efficiency of FEC, the source rate must be continuously controlled to avoid congestion.
Abstract: Real-time audio over the best effort Internet often suffers from packet loss. So far, Forward Error Correction (FEC) seems to be an efficient way to attenuate the impact of loss. Nevertheless to ensure efficiency of FEC, the source rate must be continuously controlled to avoid congestion. In this paper, we describe a realisation of adaptive FEC subdued to a TCP-friendly rate control.
TL;DR: This work develops a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly, and proposes an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality.
Abstract: In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm which optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation.
TL;DR: This thesis proposes models to evaluate the performance of real-time multimedia applications and proposes the first analytical model for this kind of protocols accounting for delay variability based on stochastic difference equations.
Abstract: In this thesis, we propose models to evaluate the performance of real-time multimedia applications. Besides, we propose a model for AIMD protocols. The first subject we study is a simple error correction (FEC) mechanism. We first model the network as an M/M/1/K queuing system. We assume a linear utility function relating the audio quality and the amount of redundancy. The redundancy of packet i is carried by packet i+f. Our analysis shows that, even for the case f->inf, this simple FEC scheme always leads to a quality deterioration. Next, we model the bottleneck router as an M/G/1/K queue. We consider two cases that may contribute to a quality improvement: (a) multiplexing the audio flow with an exogenous flow, and (b) considering non-linear utility functions. Under these assumptions, we show that this FEC scheme can lead to a quality improvement. The second subject investigated is about playout delay algorithms. We propose a set of moving average algorithms allowing to control the average loss rate in an audio session. We study and compare the performance of our algorithms by simulation with real packet traces. The third subject we study is about the performance of AIMD protocols. We propose, at the best of our knowledge, the first analytical model for this kind of protocols accounting for delay variability. The model is based on stochastic difference equations. It provides a closed-form expression for the throughput and for the window size in steady state. We show by analysis and simulation that an increase in delay variability improves the performance of AIMD protocols.
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.
Abstract: A resource reservation protocol (RSVP), a flexible and scalable receiver-oriented simplex protocol, is described. RSVP provides receiver-initiated reservations to accommodate heterogeneity among receivers as well as dynamic membership changes; separates the filters from the reservation, thus allowing channel changing behavior; supports a dynamic and robust multipoint-to-multipoint communication model by taking a soft-state approach in maintaining resource reservations; and decouples the reservation and routing functions. A simple network configuration with five hosts connected by seven point-to-point links and three switches is presented to illustrate how RSVP works. Related work and unresolved issues are discussed. >
TL;DR: A number of packet loss recovery techniques for streaming audio applications operating using IP multicast, and a series of recommendations for repair schemes to be used based on application requirements and network conditions are made.
Abstract: We survey a number of packet loss recovery techniques for streaming audio applications operating using IP multicast. We begin with a discussion of the loss and delay characteristics of an IP multicast channel, and from this show the need for packet loss recovery. Recovery techniques may be divided into two classes: sender- and receiver-based. We compare and contrast several sender-based recovery schemes: forward error correction (both media-specific and media-independent), interleaving, and retransmission. In addition, a number of error concealment schemes are discussed. We conclude with a series of recommendations for repair schemes to be used based on application requirements and network conditions.
TL;DR: This document describes a profile called "RTP/AVP" for the use of the real-time transport protocol (RTP) and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control.
Abstract: This document describes a profile called "RTP/AVP" for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings.
TL;DR: A simple algorithm is obtained that optimizes a subjective measure as opposed to an objective measure of quality, and incorporates the constraints of rate control and playout delay adjustment schemes, and it adapts to varying loss conditions in the network.
Abstract: Excessive packet loss rates can dramatically decrease the audio quality perceived by users of Internet telephony applications. Previous results suggest that error control schemes using forward error correction (FEC) are good candidates for decreasing the impact of packet loss on audio quality. However, the FEC scheme must be coupled to a rate control scheme. Furthermore, the amount of redundant information used at any given point in time should also depend on the characteristics of the loss process at that time (it would make no sense to send much redundant information when the channel is loss free), on the end to end delay constraints (destination typically have to wait longer to decode the FEC as more FEC information is used), on the quality of the redundant information, etc. However, it is not clear given all these constraints how to choose the "best" possible redundant information. We address this issue, and illustrate the approach using an FEC scheme for packet audio standardized in the IETF. We show that the problem of finding the best redundant information can be expressed mathematically as a constrained optimization problem for which we give explicit solutions. We obtain from these solutions a simple algorithm with very interesting features, namely (i) the algorithm optimizes a subjective measure (such as the audio quality perceived at a destination) as opposed to an objective measure of quality (such as the packet loss rate at a destination), (ii) it incorporates the constraints of rate control and playout delay adjustment schemes, and (iii) it adapts to varying loss conditions in the network (estimated online with RTCP feedback). We have been using the algorithm, together with a TCP-friendly rate control scheme and we have found it to provide very good audio quality even over paths with high and varying loss rates. We present simulation and experimental results to illustrate its performance.