• Proceedings of the KIEE Conference
• /
• 2000.07d
• /
• pp.3028-3030
• /
• 2000

In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

• Journal of Communications and Networks
• /
• v.7 no.1
• /
• pp.1-12
• /
• 2005

Ethernet is being deployed in metropolitan area networks (MANs) as a lower-cost alternative to SONET-based infrastructures. MANs are usually required to support common communication services, such as voice and frame relay, based on legacy synchronous TDM technology in addition to asynchronous packet data transport. This paper addresses the clock synchronization problem that arises when transporting synchronous services over an asynchronous packet infrastructure, such as Ethernet. A novel algorithm for clock synchronization is presented combining time-stamp methods used in the network time protocol (NTP) with signal processing techniques applied to measured packet interarrival times. The algorithm achieves the frequency accuracy, stability, low drift, holdover performance, and rapid convergence required for viable emulation of TDM circuit services over Ethernet.

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 2006.07a
• /
• pp.489-489
• /
• 2006

• Proceedings of the Korean Institute of Communication Sciences Conference
• /
• 1986.04a
• /
• pp.57-61
• /
• 1986

• 한국정보통신설비학회:학술대회논문집
• /
• 2007.08a
• /
• pp.354-358
• /
• 2007

This article presents several methods of mobile communication network to access technologies utilizing unlicensed spectrum interworking. Generic Access Network (GAN) technology was already specified in GERAN (GSM EDGE Radio Access Network) and Interworking WLAN (I-WLAN) was standardized for WCDMA system for WLAN user to access WCDMA packet based services through WLAN access point. Voice Call Continuity is not access network dependent technology but is a kind of domain change scheme for voice call from Circuit Switching (CS) network to IP Multimedia Subsystem (IMS) and vice versa.

• ETRI Journal
• /
• v.28 no.3
• /
• pp.311-319
• /
• 2006

This paper presents a terminal-assisted frame-based packet reservation multiple access (TAF-PRMA) protocol, which optimizes random access control between heterogeneous traffic aiming at more efficient voice/data integrated services in dynamic reservation TDMA-based broadband access networks. In order to achieve a differentiated quality-of-service (QoS) guarantee for individual service plus maximal system resource utilization, TAF-PRMA independently controls the random access parameters such as the lengths of the access regions dedicated to respective service traffic and the corresponding permission probabilities, on a frame-by-frame basis. In addition, we have adopted a terminal-assisted random access mechanism where the voice terminal readjusts a global permission probability from the central controller in order to handle the 'fair access' issue resulting from distributed queuing problems inherent in the access network. Our extensive simulation results indicate that TAF-PRMA achieves significant improvements in terms of voice capacity, delay, and fairness over most of the existing medium access control (MAC) schemes for integrated services.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.9B
• /
• pp.1272-1280
• /
• 2010

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in condition of no packet loss, and the WFQ guarantees those QoS requirements with the allocated resource. In a practice, however, most QoS-guaranteed services, specially the Voice of IP, allow a few percent of packet loss, so it is strongly desired that the RSVP and WFQ make the best use of this allowable packet loss. This paper enhances the WFQ to allow packet loss and investigates its performance. The performance evaluation showed that allowing the packet loss of 0.4% can improve the flow admission capability by around 40 percent.

• Proceedings of the KIEE Conference
• /
• 2001.07d
• /
• pp.1990-1992
• /
• 2001

In this paper, a NeuroFuzzy controller (NFC) with enhanced packet reservation multiple access (PRMA) protocol for QoS-guaranteed multimedia communication systems is proposed. The enhanced PRMA protocol adopts mini-slot technique for reducing contention cost, and these minislot are futher partitioned into multiple MAC regions for access requests coming from users with their respective QoS (quality-of-service) requirements. And NFC is designed to properly determine the MAC regions and access probability for enhancing the PRMA efficiency under QoS constraint. It mainly contains voice traffic estimator including the slot information estimator with recurrent neural networks (RNNs) using real-time recurrent learning (RTRL), and fuzzy logic controller with Mandani- and Sugeno-type of fuzzy rules. Simulation results show that the enhanced PRMA protocol with NFC can guarantee QoS requirements for all traffic loads and further achieves higher system utilization and less non real-time packet delay, compared to previously studied PRMA, IPRMA, SIR, HAR, and F2RAC.

• Journal of Internet Computing and Services
• /
• v.4 no.5
• /
• pp.61-68
• /
• 2003

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and Maximizing channel utilization. Data terminals with random data packets are not provided slot reservation with PRMA protocol. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme have to be performed for loss sensitive data packet, and contended their quality of service, Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. So the terminal which wants to request for ABR type service, is allocated a minimum bandwidth from system for the first time, If the system have some extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot, As a result, APRMA protocol can support the data service with loss sensitivity and maintain their channel utilization high.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.39 no.1
• /
• pp.20-33
• /
• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.