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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: senderand 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. 1 Introduction The development of IP multicast and the Internet multicast backbone has led to be emergence of a new class of scalable audio/video conferencing applications. These are based on the lightweight sessions model [11, 17] and provide efficient multi-way communication which scales fr...

The current Internet provides a single class best effort service. From an application's point of view, this service amounts in practice to providing channels with time-varying characteristics such as delay and loss distributions. One way to support real time applications such as interactive audio given this service is to use control mechanisms that adapt the audio coding and decoding processes based on the characteristics of the channels, the goal be gin to maximize the quality of the audio delivered to the destinations. In this paper, we describe and analyze a set of such control mechanisms. They include a jitter control mechanism and a combined error and rate control mechanism. These mechanisms have been implemented and evaluated over the Internet and the MBone. Experiments indicate that they make it possible to establish and maintain reasonable quality audioconferences even across fairly congested connections.

Excessive packet loss rates can dramatically decrease the audio quality perceived by users of Internet telephony applications. Recent results suggest that error control schemes using forward error correction (FEC) are good candidates for decreasing the impact of packet loss on audio quality. With FEC schemes, redundant information is transmitted along with the original information so that the lost original data can be recovered at least in part from the redundant information. Clearly, sending additional redundancy increases the probability of recovering lost packets, but it also increases the bandwidth requirements and thus the loss rate of the audio stream. This means that 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 th...

Real-time audio and video data streamed over unreliable IP networks, such as the Internet, may encounter losses due to dropped packets or late arrivals. This paper reviews error-concealment schemes developed for streaming realtime audio and video data over the Internet. Based on their interactions with (video or audio) source coders, we classify existing techniques into source coder-independent schemes that treat underlying source coders as black boxes, and source coder-dependent schemes that exploit coder-specific characteristics to perform reconstruction. Last, we identify possible future research directions. 1. Introduction Increases in bandwidth and computational speed lead to growing interests in real-time audio and video transmissions over the Internet. In the Internet, packets carrying real-time data may be dropped or arrive too late to be useful because the Internet is a packet-switched, best-effort delivery service, with no guarantee on the quality of service (QoS). Traditi...

Recently we have seen research efforts on how to protect a real-time speech signal when transmitting over an unreliable packet-switched network like the Internet by open-loop error control. Research has covered the type of Foward Error Correction (generic or voice-specific), the protocol support needed and adaptivity to the current network congestion state. However, the sender does not take into account that some segments of the signal are essential to the speech quality, while others can be extrapolated at the receiver from data received earlier in the event of a packet loss. This is especially true for modern frame-based codecs like the G.729 and G.723.1 which contain an internal loss concealment algorithm. Thus, the sender consumes additional bandwidth and aggravates the congestion in the Internet by sending unnecessary redundancy. In this paper we first analyze the concealment performance of the G.729 decoder. We find that the loss of unvoiced frames can be concealed well. Also, the loss of voiced frames is concealed well once the decoder has obtained sufficient information on them. However the decoder fails to conceal the loss of voiced frames at an unvoiced/voiced transition because it extrapolates internal state (filter coefficients and excitation) for an unvoiced sound. Moreover, once the encoder has failed to build the appropriate linear prediction synthesis filter, it takes a long time for the decoder to resynchronize with the encoder. Using this result, we then develop a new FEC scheme to support frame-based codecs, which adjusts the amount of added redundancy adaptively to the properties of the speech signal. Objective quality measures (ITU P.861A and EMBSD) show that our speech property-based FEC (SPB-FEC) scheme achieves almost the same speech quality a...

The evolution of computers and networks leaves no doubt about the important role of interactive multimedia services. The quality of these new services will be a key issue for their wide deployment, and this quality is determined by the opinion of the users. The best Quality of Service (QoS) is not the highest, but the most suitable to the different users' needs. In order to provide a suitable level of QoS, an application needs to know which relevant network parameters have impact on the quality as it is perceived by the users. This paper presents a top-down approach to study the influence of network performance parameters on the user perceived quality, taking conversational speech as an example of a simple, essential and demanding application. Three types of network are considered: IP, ATM and IP over ATM. For each of these cases, the influence of the corresponding network performance parameters is studied. 1

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There are a number of packet-loss recovery techniques proposed for streaming audio applications recently. However, there are few works that are able to exploit the tradeoff between the recovery quality and the computational complexity. In this paper, we develop a recovery method, called DSPWR (Double Sided Pitch Waveform Replication) which is able to tolerate a much higher packet loss rate. In essence, DSPWR is composed of several procedures devised to improve the quality of the reconstructed speech. It is noted that a more sophisticated recovery scheme that can tolerate a higher degree of packet loss in general requires a larger computational cost. In view of this, we evaluate the quality of the reconstructed speech under different packet loss rates for various receiver-based recovery methods, and compare the computational complexity among these methods. Under the acceptable speech quality whose MOS (Mean Opinion Score) is above 3.5, we develop an adaptive mechanism that can select the recovery method with the minimal complexity in accordance with different packet loss rates encountered. To conduct real experiments in the networks, we implement these recovery methods and evaluate the performance of DSPWR devised and the adaptive recovery techniques empirically. As validated by our experimental results, the adaptive mechanism is able to strike a compromise between the computational overhead and the quality of the speech desired.

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