• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.1 no.2
• /
• pp.145-158
• /
• 1997

Method for synchronizing multimedia data is needed to support continuous transmission of multimedia data through a network in a bounded time and it also required for supporting continuous presentation of multimedia data with the required norminal playout rate in distributed network environments. This paper describes a new synchronization method for supporting delay-sensitive multimedia Presentation without degration of Quality of services of multimedia application. It mainly aims to support both intermedia and intermedia synchronization by absorbing network variations which may cause skew or jitter. In order to remove asynchonization problems, we make use of logical time system, dynamic buffer control method, and adjusting synchronization intervals based on the quality of services of a multimedia. It might be more suitable for working on distribute[1 multimedia systems where the network delay variation is changed from time to time and no global clock is supported. And it also can effectively reduce the amount of buffer requirements needed for transfering multimedia data between source and destination system by adjusting synchronization intervals with acceptable packet delay limits and packet loss rates.

• The KIPS Transactions:PartC
• /
• v.12C no.1 s.97
• /
• pp.53-62
• /
• 2005

The NG-SDH system requires signal synchronization to synchronize incoming ethernet signal with GFP frame. The foreign nation research completes a chipset development until now and it secures a relation technique, but it does not secure a relation technique from domestic. Therefore, in this paper, we presented with signal synchronization method of Ethernet signal through GFP frame. We knew that the synchronized method of Ethernet signal through GFP-F must apply ingress & egress buffer and GFP Idle. We understood that the synchronized method of Ethernet signal through GFP-T must apply GFP Idle and $65B{\_}PAD$, and require maximum 3-bit addition & deletion of idle. Also we showed signal synchronization realization through simulation and obtained MTIE/TDEV characteristics and peak to peak jitter in egress output.

• The Journal of The Korea Institute of Intelligent Transport Systems
• /
• v.11 no.3
• /
• pp.48-54
• /
• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• The Journal of the Korea Contents Association
• /
• v.6 no.11
• /
• pp.295-301
• /
• 2006

The scope ID field of IPv6 multicast address indicates the zone of the destination for which a multicast traffic is intended. Without any further examination on the header field, the scope ID enables a router to determine whether the traffic will be forwarded to a subnet or not. For the graceful migration from IPv4 networks to IPv6 networks, various IPv6 applications working through IPv4 networks are indispensable during the migration period. This paper describes development of an interactive remote lecture system providing service integration on voice, image, and data of teaching materials. Access right to the network for dialog among multicast group members is controlled via additional TCP (Transmission Control Protocol) session. A jitter buffer algorithm was implemented to improve the voice communication jitters.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.9 no.1
• /
• pp.45-57
• /
• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.1 no.2
• /
• pp.102-109
• /
• 2006

In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.

• Journal of Korea Multimedia Society
• /
• v.5 no.6
• /
• pp.697-703
• /
• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.2
• /
• pp.93-101
• /
• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.6 no.2
• /
• pp.236-247
• /
• 2002

We propose a cell－multiplexing scheme for the real－time communication service in ATM network and a new service discipline guarantee end-to-end delay based on pseudo-isochronous cell switching. The proposed scheme consists of two level frame hierarchy, upper and lower frame, which is used to assign the bandwidth and to guarantee the requested delay bound, respectively. Since the Proposed algorithm employs two level frame hierarchy, it can overcome the coupling problem which is inherent to the framing strategy It can be comparatively reduce the complexity, and still guarantee the diverse delay bounds of end-to-end. Besides, it consists of two components, traffic controller and scheduller, as the imput traffic description model and regulates the input traffic specification. The function of the traffic controller is to shape real -time traffic to have the same input pattern at every switch along the path. The end-to-end delay is bounded by the scheduller which can limit the delay variation without using per-session jitter controllers, and therefore it can decrease the required buffer size. The proposed algorithm can support the QoS's of non-real time traffic as well as those of real time traffic.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.5C
• /
• pp.445-453
• /
• 2002

We propose a cell-multiplexing scheme for the real-time communication service in ATM network and a new service discipline guarantee end-to-end delay based on pseudo-isochronous cell switching. The proposed scheme consists of two level frame hierarchy, upper and lower frame, which is used to assign the bandwidth and to guarantee the requested delay bound, respectively. Since the proposed algorithm employs two level frame hierarchy, it can overcome the coupling problem which is inherent to the framing strategy. The proposed scheme consists of two components, traffic controller and scheduller, as the imput traffic description model and regulates the input traffic specification. The function of the traffic controller is to shape real-time traffic to have the same input pattern at every switch along the path. The end-to-end delay is bounded by the scheduller which can limit the delay variation without using per-session jitter controllers, and therefore it can decrease the required buffer size. The proposed algorithm can support the QoS's of non-real time traffic as well as those of real time traffic