• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.48-55
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• 2010

It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.

• ETRI Journal
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• v.31 no.4
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• pp.469-471
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• 2009

This letter proposes a weighted scheduling mechanism for Internet protocol television (IPTV) to improve the loss performance of multicast transmission over an Ethernet passive optical network (EPON). We propose a new weight policy from the number of multicast receivers to proportionally allocate the downstream bandwidth of IPTV traffic. The proposed mechanism is used in an optical line terminal to decrease lost packets of favorite IPTV services because the lost multicast packets are proportional to the number of receivers. The total proportion of lost multicast packets is reduced by up to 73% in an EPON.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.6
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• pp.9-17
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• 1994

ATM (Asynchronous Transfer Mode) has a fixed-length packet transport scheme. It is one of the promising proposals in B-ISDN.Since the packet length is fixed, it can be potentially to perform the various service to users. In this paper, a queueing model for an ATM multip`exer loaded by CBR and ON/OFF input sources is considered, and the two-queue system which each type of input sources has a queue with a finite capacity is analyzed. The cell loss probabilities for a performance measures of ATM multiplexer are derived, and are also evaluated through numerical examples. As a result, the cell loss probability of ON/OFF sources for the queue size is rapidly decreased when the multiplexed number and burstiness are increased. Since cells of the CBR source have lower priority than cells of the ON/OFF source, cell loss probabilities of CBR sources are accordingly high independently of CBR cell arrival rate when the number of CBR sources is large.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1599-1609
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• 2014

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm. Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.

• Journal of KIISE:Information Networking
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• v.35 no.2
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• pp.99-111
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• 2008

In recent years, the needs for WLANs(Wireless Local Area Networks) technology which can access to Internet anywhere have been dramatically increased particularly in SOHO(Small Office Home Office) and Hot Spot. However, unlike wired networks, there are some unique characteristics of wireless networks. These characteristics include the burst packet losses due to unreliable wireless channel. Note that burst packet losses, which occur when the distance between the wireless station and the AP(Access Point) increase or when obstacles move temporarily between the station and AP, are very frequent in 802.11 networks. Conversely, due to burst packet losses, the performance of 802.11 networks are not always as sufficient as the current application require, particularly when they use TCP at the transport layer. The high packet loss rate over wireless links can trigger unnecessary execution of TCP congestion control algorithm, resulting in performance degradation. In order to overcome the limitations of WLANs environment, MAC-layer LDA(Loss Differentiation Algorithm)has been proposed. MAC-layer LDA prevents TCP's timeout by increasing CRD(Consecutive Retry Duration) higher than burst packet loss duration. However, in the wireless channel with high packet loss rate, MAC-layer LDA does not work well because of two reason: (a) If the CRD is lower than burst packet loss duration due to the limited increase of retry limit, end-to-end performance is degraded. (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily by Reducing the drainage speed of the network buffer due to the increase of CRD. In this paper, we propose a new retransmission module based on Cross-layer approach, called BLD(Burst Loss Detection) module, to solve the limitation of previous link layer retransmission schemes. BLD module's algorithm is retransmission mechanism at IEEE 802.11 networks and performs retransmission based on the interaction between retransmission mechanisms of the MAC layer and TCP. From the simulation by using ns-2(Network Simulator), we could see more improved TCP throughput and energy efficiency with the proposed scheme than previous mechanisms.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5C
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• pp.490-496
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• 2006

H.264 is the most recent video coding standard, containing improved error resilience tools than previous video compression schemes. This paper shows an analysis of the dependency of error concealment (EC) performance on the expected number of correctly received neighboring macroblock(MB)s for a lost MB, applying error concealment schemes to the raster scan mode that is used in the previous video coding standard and the flexible macroblock ordering (FMO) which is one of error-resilience tools in H.264. We also present simulation results and performance evaluation with various packet loss rates. Simulation results show that the FMO mode provides better EC performances of $1{\sim}9dB$ PSNR improvements compared to the raster scan mode because of larger expected number of correctly received neighboring MBs. The PSNR improvement by FMO mode becomes higher as the intra-frame period is larger and the packet loss rate is higher.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2019.05a
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• pp.451-453
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• 2019

Recently, as the use of wireless communication devices has increased, the wireless network usage has increased, and a wired network and a wireless network have been mixed to form a network. Existing TCP algorithms are designed for wired networks. Therefore, in the modern network environment, packet loss can not be accurately distinguished and improper congestion control is performed, resulting in degradation of TCP performance. In this paper, we propose SLDA (Support Vector Machine based Loss Discrimination Algorithm) which can accurately classify the packet loss environment to improve TCP performance and evaluate its performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1151-1168
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• 2009

Improving packet loss does not necessarily coincide with the improvement in user perceivable QoS because each frame carries different degree of importance. We propose Significance-aware packet scheduling (SAPS) to maximize user perceivable QoS. SAPS carries out two fundamental issues of packet scheduling: "What to transmit" and "When to transmit?" To adapt to the available bandwidth, it is necessarily to transmit the subset of the data packets if the entire set of packets can not be transmitted. "Packet Significance" quantifies the importance of the frame by elaborately incorporating frames' dependency. Greedy approach is used in selecting packets and transmission schedule is determined based on the Packet Significance. The proposed scheme is tested using publicly available MPEG-4 video clips. Decoding engine is embedded in the simulation software and user perceivable QoS is exposeed in termstermiSNR. Throughout the simulation based experiment, the performance of the proposed scheme is compared two other schemes: Size-based packet scheduling and Bit-rate based best effort packet scheduling. SAPS successfully incorporates the semantics of a packet and improves user perceivable QoS significantly. It successfully provides unequal protection to more important packets.