In this paper, we present the design of a new piggybacking algorithm for VoIP implemented using the G.722.2 codec. In piggybacking, multiple speech frames that include those transmitted in the past are encapsulated in a single packet. Because redundant copies of each frame are transmitted to the receiver, the receiver can recover those lost frames when one or more packets are lost or arrive late in their transmission. In this paper, we have enhanced the G.722.2 codec by removing further redundancies in the multiple frames encapsulated in piggybacking and by having only one set of LP coefficients for all the subframes encapsulated. We create multiple versions of the codec, each using a different frame size. Our new codec can encode the multiple frames with little degradation in PESQ, while having substantial bit savings. Its performance is evaluated against the original method of piggybacking over random losses, as well as that using packet traces collected in the PlanetLab.

In this paper, we study the statistical scheduling of offline subjective tests for evaluating alternative control schemes in real time multimedia applications. These applications are characterized by multiple counteracting objective quality metrics (such as delay and signal quality) that can be affected by various control schemes. However, the trade-offs among these metrics with respect to the subjective preferences of users are not defined. As a result, it is difficult to select the proper control schemes that lead to the best subjective quality at run time. Since subjective tests are expensive to conduct and the number of possible control schemes and run-time conditions is prohibitively large, it is important that a minimum number of such tests be conducted offline, and that the results learned can be generalized to unseen conditions with statistical confidence. To this end, we study in this paper efficient algorithms for scheduling a sequence of subjective tests, while leaving the generalization of limited offline subjective tests to guide the operation of the control schemes at run time to a future paper. Using Monte Carlo simulations, we verify the robustness of our algorithms in terms of their accuracy and efficiency.

In this paper, we propose four methods for equalizing the silence periods experienced by users in a multi-party VoIP conversation in order to improve their perceived conversational quality. To mitigate the unbalanced silence periods caused by delay disparities in Internet connections, the playout scheduler at the receiver of each client equalizes the silence periods experienced. Our limited subjective tests show that we can improve the perceptual quality when the network connections are lossy and have large delay disparities. Because it is impossible to conduct subjective tests under all possible conditions, we have developed a classifier that learns to select the best equalization algorithm using learning examples derived from subjective tests under limited network and conversational conditions. Our experimental results show that our classifier can consistently pick the best algorithm with the highest subjective conversational quality under unseen conditions, and that our system has better perceptual quality when compared to that of Skype (Version 3.6.0.244).

In this paper, we study the scheduling of off-line subjective tests for evaluating alternative parameter values of control schemes in real-time multimedia applications. These applications are characterized by multiple counteracting objective quality metrics (such as delay and signal quality) that can be affected by the control schemes. However, the tradeoffs among these metrics with respect to the subjective preferences of users are not defined. As a result, it is difficult to use the proper control value that leads to the best subjective quality without carrying out subjective tests. Since subjective tests are expensive to conduct and the number of possible control values and run-time conditions is prohibitively large, it is important that a minimum number of such tests be conducted, and that the results learned can be generalized to unseen conditions with statistical confidence. To this end, we study in this paper optimal algorithms for scheduling a sequence of subjective tests, while leaving the generalization of limited off-line subjective tests to guide the operations of the control schemes at run time to a future paper. Using an example application in the design of the play out scheduling (POS) algorithm for a two-party VoIP system, we study the accuracy and efficiency of conducting subjective tests simultaneously.

In this paper, we present the design of a VoIP conferencing system that enables the voice communication of multiple users in the Internet. After studying the conversational dynamics in multi-party conferencing, we identify user-observable metrics that affect the perception of conversational quality and their trade-offs. Based on the dynamics and the behavior on delays, jitters, and losses of Internet traces collected in the PlanetLab, we design the transmission topology and schemes for loss concealments and play-out scheduling. Last, we compare the performance of our system and Skype (version 3.5.0.214) using repeatable experiments that simulate human participants and network conditions in a multi-party conferencing scenario.

In this paper, we present new adaptive playout scheduling (POS) and loss concealment (LC) schemes for delivering high and consistent conversational voice communication quality (CVCQ) perceived by users in real-time VoIP systems. We first characterize the delay and loss conditions of an IP network and a human conversation in a VoIP system. We then identify the attributes that affect the human perception of CVCQ, which include listening-only speech quality (LOSQ), conversational interactivity (CI), and conversational efficiency (CE). We investigate their trade-offs with respect to system-controllable mouth-to-ear delays (MEDs) and the amount of redundant piggybacking. Finally, we evaluate our adaptive POS and redundancy-based LC schemes by packet traces collected in the PlanetLab.

In this paper, we evaluate die conversational voice communication quality (CVCQ) of VoIP systems, both from the user and the system perspectives. We first identify the metrics for CVCQ, which include listening-only speech quality (LOSQ). conversational interactivity (CI), and conversational efficiency (CE). These depend on the mouth-to-ear delays (MEDs) between the two clients. Based on packet traces collected in the PlanetLab and on the dynamics of human interactive speech, we study four popular VoIP client systems: Skype (v2.5), Google-Talk (Beta), Windows Live Messenger (v8.0). and Yahoo Messenger with Voice (v8.0). under various network and conversational conditions.

In this paper, we study Skype and Google Talk, two widely used VoIP systems, and compare their perceptual speech quality with that of our proposed system using UDP packet traces collected in the Planet Lab. Based on methods for speech coding, packetization, jitter control, estimation and feedback of network conditions, and loss concealments, our results show that Skype has noticeable quality degradations because it only uses a maximum of two-way redundancy for loss concealment, does not handle out-of-order arrivals, and applies a fixed jitter control of 60 ms relative to the expected arrival time. Its slow loss-adaptation time of more than one minute to change from one-way to two-way redundancy makes it susceptible to quality degradation under fast changing loss conditions. In contrast, Google Talk does not employ any loss adaptation and performs similar to Skype under low- to medium-loss scenarios. By addressing the shortcomings of Skype and Google Talk, we demonstrate improvements in speech quality in our proposed prototype

In this paper, we propose a speech-adaptive layered-coding (LC) scheme for the loss concealments of real-time CELP-coded speech transmitted over IP networks. Based on the ITU G.729 CS-ACELP codec operating at 8 Kbps, we design a loss-robust speech-adaptive codec at the same bit rate. Our scheme employs LC with redundant packetization in order to conceal losses and adapt to dynamic loss conditions characterized by the loss rate and the degree of burst, while maintaining an acceptable end-to-end delay. By protecting only the most important excitation parameters of each frame according to its speech type, our approach enables more efficient use of the bit budget. Our scheme delivers good-quality speech with a level of protection similar to full replication under medium loss rates, provides speech quality similar to the standard G.729 under very low loss rates, and outperforms both for low-to-medium loss rates.

In this paper, we propose a layered CELP speech coding (LC) scheme that adapts dynamically to the characteristics of the speech encoded and the network loss conditions in real time transmissions of voice over IP. Based on the ITU G.729 CS-ACELP codec operating at 8 Kbps, we design a variable bit-rate codec that is robust to losses and delays in IP networks. To cope with bursty losses while maintaining an acceptable end-to-end delay, our scheme employs LC with redundant piggybacking of perceptually important parameters in the base layer, with a degree of redundancy adjusted according to feedbacks from receivers. Under various delay constraints, we study trade-offs between the additional bit rate required for redundant piggybacking and the protection of perceptually important parameters. Experimental results show that our scheme works well and has quality comparable to full replication