Playout Buffering for Conversational Voice over IP

In Voice over IP, the quality of interactive conversation is important to users. Major factors affecting perceived quality are delay, delay jitter, and missing packets. For conversational VoIP, a conversational delay also plays an important role for perceived quality. Large conversational delay can result in double talk, echo or even the termination of the conversation. In practice, a playout buffer is introduced at the receiver’s side to remove delay jitter, so that the voice information carried on packets can be available at regular intervals for decoding. A longer buffer reduces the possibility of late packet loss at the expense of increasing conversational delays. Since the time delay of a playout buffer is a major addition to conversational delay, to keep conversational interactivity, it is desirable to design a playout buffer to be short but capable of protecting packets against late packet loss. In this thesis, we will explore playout buffering algorithms with improved conversational quality. We propose a quality-based adaptive playout buffering algorithm with improved voice quality and reduced conversational delays. We use the E-Model R factor as the cost index to obtain playout delays which adapt for each talkspurt. Special steps are taken to reduce conversational delay : (1) immediately play out stretched speech carried on the first packet of a talkspurt when received (stretching provides additional buffer delay for following packets) ; (2) compress the speech segment carried on the packets in the playout buffer at the end of a talkspurt (compression reduces the playout delay for the packets). As other quality-based algorithms, our scheme is subject to burst losses. To improve perceived quality further, we use sender-driven repair algorithms, in which a sender sends redundancy information, to mitigate the impact of the missing packets due to network (lost packets) and buffer underflow (late packets) without increasing buffer delays. In this thesis, we develop a new adaptive forward error correction (FEC) scheme to provide redundancy without additional delay and apply it to our adaptive playout buffering algorithm for improved perceived quality. As an alternative sender-based technique to send redundancy information, a path diversity scheme uses multiple paths (here we consider two paths). Redundant information is sent on a second path. We consider four different path diversity schemes (two of them are proposed based on E-model in this work), and design corresponding playout buffering algorithms based on conversational quality including both calling quality and interactivity.