• Journal of Korea Multimedia Society
• /
• v.7 no.4
• /
• pp.518-531
• /
• 2004

This paper discusses a testbed architecture for implementing scalable service differentiation in the Internet. The differentiated services (DiffServ) testbed architecture is based on a model in which a bandwidth broker (BB) can control network resources, and the ALTQ can reserve resources in a router to guarantee a Quality of Service (QoS) for incoming traffic to the testbed. The reservation and releasemessage for the ALTQ is contingent upon a decision message in the BE. The BB has all the information in advance, which is required for a decision message, in the form of PIB. A signaling protocol between the BB and the routers is the COPS protocol proposed at the IETF. In terms of service differentiation, a user should make an SLA in advance, and reserve required bandwidth through an RAR procedure. The SLA and RAR message between a user and the BB has implemented with the COPS extension which was used between a router and the BB. We evaluates the service differentiation for the video streaming in that the EF class traffic shows superb performance than the BE class traffic where is a network congestion. We also present the differentiated service showing a better packet receiving rate, low packet loss, and low delay for the EF class video service.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.6B
• /
• pp.595-604
• /
• 2009

The HCCA reserves the channel resources based on the mean data rate in IEEE 802.11e. It may cause either the waste of channel resource or the increase of transmission delay at MAC layer if the frame size is rapidly varied when a compressed mode video codec such as MPEG video is used. To solve these problems, it is developed that the packet scheduler allocates the wireless resource adaptation by according to the packet size. However, it is difficult to perform the admission control because of the difficulty with calculating the available resources. In this paper, we propose a CAC mechanism to solve the problem that may not satisfy the QoS by increasing traffic load in case of using EDCA. Especially, the proposed CAC mechanism calculates the EB of TSs using the traffic information transmitted by the application layer and the number of average transmission according to the wireless channel environment, and then determines the admission of the TS based on the EB. According to the simulation results of the proposed CAC mechanism, it admitted the TSs under the loads which are satisfied within the delay bound. Therefore, the proposed mechanism guarantees QoS of streaming services effectively.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.3B
• /
• pp.380-389
• /
• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.44 no.6
• /
• pp.100-110
• /
• 2007

When compressed video data is transmitted over error-prone network such as wireless channel, data is likely to be lost, so the quality of reconstructed picture is severely decreased. It is specially so in case that important information such as motion vector or macroblock mode is lost. H.264/AVC standard includes DP as error resilient technique for protecting important information from error in which data is labeled according to its relative importance. But DP technique requires a network that supports different reliabilities of transmitted data. In general, the benefits of UEP is sought by sending multiple times of same packets corresponding to important information. In this paper, we propose MDC technique based on data partitioning technique. The proposed method encodes motion vector of H.264/AVC standard into multiple parts using MDC and transmits each part as independent packet. Even if partial packet is lost, the proposed scheme can decode the compressed bitstream by using estimated motion vector with partial packets correctly transmitted, so that achieving improved performance of error concealment with minimal effect of channel error. Also in decoding process, the proposed multiple description matching increases the accuracy of estimated lost motion vector and quality of reconstructed video.

• The Transactions of the Korea Information Processing Society
• /
• v.5 no.12
• /
• pp.3047-3062
• /
• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of KIISE:Information Networking
• /
• v.37 no.5
• /
• pp.317-326
• /
• 2010

Nowadays, transmission bandwidth for network traffic is increasing and the type is varied such as peer-to-peer (PZP), real-time video, and so on, because distributed computing environment is spread and various network-based applications are developed. However, as PZP traffic occupies much volume among Internet backbone traffics, transmission bandwidth and quality of service(QoS) of other network applications such as web, ftp, and real-time video cannot be guaranteed. In previous research, the port-based technique which checks well-known port number and the Deep Packet Inspection(DPI) technique which checks the payload of packets were suggested for solving the problem of the P2P traffics, however there were difficulties to apply those methods to detection of P2P traffics because P2P applications are not used well-known port number and payload of packets may be encrypted. A proposed algorithm for identifying P2P heavy traffics based on flow transport parameters and behavioral characteristics can solve the problem of the port-based technique and the DPI technique. The focus of this paper is to identify P2P heavy traffic flows rather than all P2P traffics. P2P traffics are consist of two steps i)searching the opposite peer which have some contents ii) downloading the contents from one or more peers. We define P2P flow patterns on these P2P applications' features and then implement the system to classify P2P heavy traffics.

• Journal of the Korea Society of Computer and Information
• /
• v.19 no.3
• /
• pp.63-71
• /
• 2014

We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.

• Journal of Broadcast Engineering
• /
• v.15 no.2
• /
• pp.205-216
• /
• 2010

In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.6
• /
• pp.630-640
• /
• 2002

The compressed video bitstream is very sensitive to transmission errors. If we lost packet or received with errors during the transmission, not only the current frame will be corrupted, but also errors will propagate to succeeding frames. Error concealment is a data recovery technique that enables the decoder to conceal effects of transmission errors by predicting the lost or corrupted video data from the previously reconstructed error free information. Motion vection recovery and motion compensation with the estimated motion vector is a good approach to conceal the corrupted macroblock data. In this paper, we prove that it is reasonable to use the estimated motion vector to conceal the lost macroblock by providing macroblock distortion models. After we propose a new motion vector recovery algorithm based on optical flow fields, we compare its performance to those of conventional error concealment methods. The proposed algorithm has smaller computational complexity than those of conventional algorithms.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.12 no.2
• /
• pp.860-872
• /
• 2018

In this Paper, we propose 8K UHD (Ultra High Definition) encapsulation method for efficient transmission and reception based on MMT (MPEG Media Transport). Broadcasting services for 8K UHD allow users to feel the maximized reality. However, present technology is difficult to provide 8K UHD in broadcasting networks, because the 8K UHD bitrate is too high to be transmitted in the current broadcasting networks. Research for transmitting 8K UHD is underway. In some researches, a receiver is implemented with four 4K UHD display instead of a 8K UHD display. In order to transmit 8K UHD within the limited transmission bitrate of broadcasting network, 8K UHD contents encoded by SHVC (Scalable High Efficiency Video Coding) and then transmitted over heterogeneous network. For using the broadcasting and communication networks, MMT standard is used. MMT is IP based transmission protocol as the next generation transmission protocol. According to the MMT standard, video stream encapsulated and transmitted in MMTP (MMT Protocol) packet. IP-based broadcasting and communication networks can be used to transmit simultaneously, and the receiver can synchronize and play it. We propose an encapsulation method that can efficiently transmit and receive 8K UHD. The proposed method increases a payload rate and decreases an initial delay at the receiver. We show that the efficiency of the proposed method is verified by experimental tests.