Koichi Nihei

Quality evaluation of voice over multiple TCP connections

Voice communications over the Internet are growing rapidly thanks to faster Internet and mobile networks. Although multimedia communication techniques such as voice over IP (VoIP) have been using UDP, TCP is gaining gradual popularity since it enables passage through network address translators (NATs) and firewalls. Many researches have been done on real-time communications using TCP, which indicates the possibility of achieving low delay communications over TCP. However, little mention has been made of user-perceived voice quality over TCP. This paper presents a VoIP system using multiple TCP connections. We conducted thorough voice quality evaluations over these connections by changing network and VoIP application parameters. Our evaluations show that voice communications can be achieved in 10% packet loss rate and 100 ms round-trip time environments if the number of TCP connections, the sending unit and the decoding buffer size are optimized. We also provide optimal parameter guidelines for achieving high quality voice communications over multiple TCP connections.

QoE maximizing bitrate control for live video streaming on a mobile uplink

Video streaming is now one of the main applications on mobile networks. Since the throughput fluctuates widely on a mobile network, adaptive bitrate control methods that control the video bitrate in accordance with this fluctuation have been developed. These methods decide the bitrate at a one-second or a few-second interval. In most of these methods, the output of a decision is a bitrate value, and the decided bitrate continues until the next decision. This paper models the behavior of video packets on a mobile uplink and proposes a novel bitrate control method. In the method, the output of a decision is the bitrate change over time for several seconds into the future, not a bitrate value, that maximizes an integrated value of the quality of experience (QoE) indicator for the model of behavior. This paper compares the proposed method with conventional methods by simulation. The proposed method improved the mean opinion score (MOS), which indicates the estimated QoE as a value between 1 and 5, by up to 0.2.

Robust available bandwidth estimation against dynamic behavior of packet scheduler in operational LTE networks

We shed new light on the mechanism behind how the dynamic behavior of a packet scheduler at the link layer in mobile networks degrades the accuracy of conventional available bandwidth (i.e., unused capacity of an end-to-end path) estimation methods that use a probing packet train (i.e., a set of multiple probing packets). Most of the conventional methods, which were originally designed for wired networks, estimate available bandwidth at the receiver by detecting changes of the observed queuing delays of probing packets. They utilize a microscopic approach in which they check the difference of the queuing delay of each packet on a packet-by-packet basis in order to detect the queuing delay changes. We found that the dynamic behavior of a packet scheduler at the link layer dramatically disturbs the queuing delays observed at the receiver. The disturbed queuing delays make it tremendously difficult for the conventional microscopic approach to detect changes of the delays, resulting in degraded estimation accuracy.

A QoE indicator and a transmission control method for VoIP on mobile networks considering delay spikes

Voice over IP (VoIP) services are now offered on many mobile networks. However, delay spikes, which are sudden large increases in packet delay, affect the quality of VoIP on mobile networks. This paper proposes a quality of experience (QoE) indicator considering delay spikes. The QoE indicator combines a voice quality indicator such as ITU-T P.862 and the delay part of ITU-T G.107, which is a quality indicator for VoIP. This paper also proposes a transmission control method that selects the codec type and the sending interval of voice packets so as to maximize the QoE indicator. The method improves the mean opinion score (MOS), which indicates QoE as a value between 1 and 5, by up to 0.4.

Study on mechanism and reduction approaches of delay spikes occurrence on mobile networks

Since smartphones have spread and the transmission rate of mobile networks has accelerated, a lot of VoIP services are provided on mobile networks. Voice quality degradation due to delay spikes (i.e. sudden large increases in the packet delay) is the key issue for VoIP on mobile networks. To resolve this issue, adaptive jitter buffer control methods have been proposed to deal with delay spikes. These methods control the length of the jitter buffer in order to reduce voice quality degradation due to delay spikes. For further improving the quality of voice communication, it is effective to reduce the number of delay spikes. This paper analyzes the mechanism of delay spikes occurrence and proposes delay spike reduction approaches by controlling size and interval of voice packets on the basis of radio quality. Simulation results on the network simulator (ns-3) show the proposed approaches reduce the number of delay spikes by more than 50%.

A dynamic rate switching method for live video streaming using a smart device

Live streaming services over HTTP used on smart devices via a mobile network have become very popular. However, playback interruption often occurs during a live streaming. This is because TCP throughput fluctuates due to radio strength variation and cross-traffic, and the playback buffer on the client side is small. To solve the playback interruption problem, adaptive rate control techniques are helpful. However, using a smart device with insufficient CPU power leads to control delays, which cause the rate to be controlled improperly. This paper proposes a novel rate control method that uses plural codecs and detection of throughput degradation, and shows that the proposed method reduces playback interruptions on smart devices. We evaluate our proposed method with a live streaming for 100 seconds as an experiment in conditions in which the TCP throughput is fluctuated by a network emulator. Playback interruption occurred three times and lasted 1.8 seconds in total with the conventional method, but did not occur with the proposed method. We also evaluate the method from the viewpoint of QoE defined in ITU-T Rec.P.1201/Amd.2. The proposed method is able to improve QoE(MOS) to 3.5 from 2.6 of conventional method.

Load reduction methods for DTMF decoders with adaptive omission of frequency analysis

Dual-Tone Multi-Frequency (DTMF) signaling is used in telecommunication systems for dialing telephone numbers and sending commands to equipment such as interactive voice response (IVR) systems. There are two types of signalings to transmit DTMF digits: in-band and out-of-band. Since many systems use in-band signaling, equipment that uses DTMF signaling needs to support in-band signaling. Media servers that provide IVR services must process a huge number of sessions in parallel. The load of decoding in-band DTMF digits needs to be reduced to increase in the number of sessions processed by media servers. This paper presents load reduction methods with adaptive omission of frequency analysis and simulation results showing that the presented methods reduced the load of DTMF decoding by more than 50%.