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Nowadays mobile streaming service through cell phone is becoming the highlight of new value-added mobile services. Based on the present CDMA1x wireless data network and Binary Runtime Environment for Wireless (BREW) platform, adop...
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Nowadays mobile streaming service through cell phone is becoming the highlight of new value-added mobile services. Based on the present CDMA1x wireless data network and Binary Runtime Environment for Wireless (BREW) platform, adopting compression technologies of H.264 and QCP, a set of streaming media players are designed and implemented, and the principle, structure, key technologies and performance analysis of this system are introduced. This player works well in practice.
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If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The g...
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If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.
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IPTV video services are increasingly being considered for delivery to mobile devices over broadband wireless access networks. The IPTV streams or channels are multiplexed together for transport across an IP core network prior to d...
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IPTV video services are increasingly being considered for delivery to mobile devices over broadband wireless access networks. The IPTV streams or channels are multiplexed together for transport across an IP core network prior to distribution across the access network. According to the type of access network, prior bandwidth constraints exist that restrict the multiplex data-rate. This paper presents a bandwidth allocation scheme based on content complexity to equalize the overall video quality of the IPTV sub-streams, in effect a form of statistical multiplexing. Bandwidth adaptation is achieved through a bank of bit-rate transcoders. Complexity metrics serve to estimate the appropriate bandwidth share for each stream, prior to distribution over a wireless or ADSL access network. These metrics are derived after entropy decoding of the input compressed bit-streams, without the delay resulting from a full decode. Fuzzy-logic control serves to adjust the balance between spatial and temporal coding complexity. The paper examines constant and varying bandwidth scenarios. Experimental results show a significant overall gain in video quality in comparison to a fixed bandwidth allocation.
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On chip interconnect plays a dominant role on the circuit performance in both analog and digital domains. Interconnects can no longer be treated as mere delays or lumped RC networks. Crosstalk, ringing and reflections are just som...
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On chip interconnect plays a dominant role on the circuit performance in both analog and digital domains. Interconnects can no longer be treated as mere delays or lumped RC networks. Crosstalk, ringing and reflections are just some of the issues that need to be addressed for the efficient design of high speed VLSI circuits. In order to accurately model these high frequency effects, inductance had been taken into consideration. Within this frequency range, the most accurate simulation model for on-chip VLSI interconnects was the distributed RLC model. Unfortunately, this model has many limitations at much higher of operating frequency used in today’s VLSI design. This can lead to inaccurate simulations if not modeled properly. At even higher frequency the conductance metrics has become a dominant factor and has to be taken into consideration for accurate modeling of the different on-chip performance parameters. The traditional analysis of crosstalk in a transmission line begins with a lossless LC representation, yielding a wave equation governing the system response. With the increase in frequency and interconnection length due to the increase in the number of on-chip devices, the lossy components are prevailing than the lossless components. With the reduction of pitch between the adjacent wires in deep sub-micron technologies, coupling capacitances are becoming significant. This increase in capacitances results the introduction of noise which is capable of propagating a logical fault. An inaccurate estimation of the crosstalk could be the origin of the malfunction of the circuit. Cross talk can be analyzed by computing the signal linkage between aggressor and victim nets. The aggressor net carries a signal that couples to the victim net through the parasitic capacitances [13]. To determine the effects that this cross talk will have on circuit operation, the resulting delays and logic levels for the victim nets must be computed. This paper proposes a difference model approach to derive crosstalk in the transform domain. A closed form solution for crosstalk is obtained by incorporating initial conditions using difference model approach for distributed RLCG interconnects. We have proposed an explicit expression for the estimation of cross-talk noise. Our model considers both lossless components (i.e. L, C) and lossy components (i.e. R, G). The SPICE simulation justifies the accuracy of our proposed approach.
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Voice over Internet Protocol (VoIP) has launched 20 years ago. After its launching, VoIP has become one of the most popular and powerful technologies of the 20th and 21st century [9]. With millions of users from business phones to...
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Voice over Internet Protocol (VoIP) has launched 20 years ago. After its launching, VoIP has become one of the most popular and powerful technologies of the 20th and 21st century [9]. With millions of users from business phones to social networking apps, VoIP has become an underlying technology that has power the way we people connect to each other. Developing at 6% compound annual growth rate (CAGR), VoIP is expected to have a total market of $82.7 billion by 2017 [9]. On top of this, as indicated by the site Telecom Reseller, VoIP is one of the top performing businesses of this decade, alongside biotechnology and e-commerce. Whereas the wired communication is one of the worst. In this paper call recording solution is developed by using the latest technology such as SIP. This solution is scalable and can be implemented by telecom service provider companies which works on VoIP.
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The scalable extension of H.264/AVC, known as scalable video coding or SVC, is currently the main focus of the Joint Video Team's work. In its present working draft, the higher level syntax of SVC follows the design principles of ...
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The scalable extension of H.264/AVC, known as scalable video coding or SVC, is currently the main focus of the Joint Video Team's work. In its present working draft, the higher level syntax of SVC follows the design principles of H.264/AVC. Self-contained network abstraction layer units (NAL units) form natural entities for packetization. The SVC specification is by no means finalized yet, but nevertheless the work towards an optimized RTP payload format has already started. RFC 3984, the RTP payload specification for H.264/AVC has been taken as a starting point, but it became quickly clear that the scalable features of SVC require adaptation in at least the areas of capability/operation point signaling and documentation of the extended NAL unit header. This paper first gives an overview of the history of scalable video coding, and then reviews the video coding layer (VCL) and NAL of the latest SVC draft specification. Finally, it discusses different aspects of the draft SVC RTP payload format, including the design criteria, use cases, signaling and payload structure.
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As the VoIP steganographic methods provide a low capacity covert channel for data transmission, an efficient and real-time data transmission protocol over this channel is required which provides reliability with minimum bandwidth ...
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As the VoIP steganographic methods provide a low capacity covert channel for data transmission, an efficient and real-time data transmission protocol over this channel is required which provides reliability with minimum bandwidth usage. This paper proposes a micro-protocol for data embedding over covert storage channels or covert hybrid channels developed by steganographic methods where real-time transport protocol (RTP) is their underlying protocol. This micro-protocol applies an improved Go-Back-N mechanism which exploits some RTP header fields and error correction codes to retain maximum covert channel capacity while providing reliability. The bandwidth usage and the performance of the proposed micro-protocol are analyzed. The analyses indicate that the performance depends on the network conditions, the underlying steganographic method, the error correction code and the adjustable parameters of the micro-protocol. Therefore, a genetic algorithm is devised to obtain the optimal values of the adjustable micro-protocol parameters. The impact of network conditions, the underlying steganographic method and the error correction code on the performance are assessed through simulations. The performance of this micro-protocol is compared to an existing method named ReLACK where this micro-protocol outperforms its counterpart.
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There are two main contributions in this paper. First, the port number assignment mechanism for each type of NAT has been probed. Second, an approach combining the use of STUN and the port assignment prediction mechanism is presen...
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There are two main contributions in this paper. First, the port number assignment mechanism for each type of NAT has been probed. Second, an approach combining the use of STUN and the port assignment prediction mechanism is presented as an effective way to improve the NAT traversal success rate, that is, the bandwidth cost of a relay server for P2P communication can be further reduced. As a preliminary step of this work, 50 commercially available NATs were categorized according to the mapping and filtering rules employed in port number assignment. With the 50 NATs as testing objects, the NAT traversal success rate was measured for various combinations of caller and callee types. The performance was compared among the STUN, multi-hole punching, and proposed approaches in terms of the success rate. The proposed approach had a success rate of 94.36%, outperforming the counterparts, and is expected to be widely applied to P2P communication apps, such as those in V~2oIP, IoT, and many more. NAT traversal remains a key issue for P2P communication in the future.
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Recently, we proposed an ad hoc, cluster-based, multihop network architecture for video communications using IEEE 802.11 FHSS (Frequency Hopping Spread Spectrum) wireless LAN (WLAN) technology with 2GFSK (2-level Gaussian Frequenc...
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Recently, we proposed an ad hoc, cluster-based, multihop network architecture for video communications using IEEE 802.11 FHSS (Frequency Hopping Spread Spectrum) wireless LAN (WLAN) technology with 2GFSK (2-level Gaussian Frequency Shift Keying) modulation capable of 1 Mb s{sup}(-1) transmission. To increase the transmission rate to 2 Mb s{sup}(-1) for higher-quality video communications, we evaluated the performance of the IEEE 802.11 FHSS system when a 4GFSK modulation option is selected. Since the 2 Mb s{sup}(-1) system utilizing 4GFSK modulation is not very efficient in terms of Radio Frequency (RF) range, to improve its performance for multihop applications a combination of diversity and noncoherent Viterbi equalizer are considered. For video transmission, we employed a bitstream-splitting technique together with a packet-based error-protection strategy to combat packet drops under multipath fading conditions. The real-time transport protocol (RTP), user datagram protocol (UDP), and Internet protocol (IP) are used for video streaming. This includes an RTP packetization scheme to control the packet size and to improve the error-resilient decoding of the partitioned video signal. The paper includes the simulation results showing the effects of the receiver design and diversity on the quality of the received video signals.
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With the advancement of digital imaging technology and broadband internet technology, the digital video surveillance industry has matured. There have been rapid changes in people's lifestyle and society, therefore personal safety ...
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With the advancement of digital imaging technology and broadband internet technology, the digital video surveillance industry has matured. There have been rapid changes in people's lifestyle and society, therefore personal safety needs are of great importance. People or enterprise builds video surveillance systems in the home or office area to provide police reference and evidence for criminal investigations. This paper is to investigate the use of video phones for home monitoring services present an innovative framework and the necessary intermediary system research and development of such devices, and introduce related systems and network operation. Furthermore, this paper hopes to examine a new type of home security monitoring service architecture that can solve the problem of high cost of imaging equipment and storage security.
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