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The Stream Control Transmission Protocol (SCTP) supports multiple streams to minimize the performance degradation due to head-of-line blocking. Among several streams transmitted simultaneously, some stream may require smaller transmission time than the others. However, the current SCTP provides no facility for controlling the access frequency of respective streams. This paper proposes a new scheme...
Datagram Congestion Control Protocol (DCCP) is an unreliable transport layer protocol with a congestion control mechanism used for multimedia streaming. In order to use multiple network interfaces to transmit streaming data smoothly, a Multi-Path Datagram Congestion Control Protocol (MP-DCCP) is proposed and presented in this paper. Video streaming transmission through multi-path faces three problems:...
Taking into account the fact that uneven video streams, instability of the network bandwidth , it's difficult to match the transmission bandwidth and video bit rate in video delivery procedure. Based on the SIP / SDP protocol, this paper presents an adaptive video streaming transmission control algorithm, customs a RTP / RTCP control strategy. Considering the current network status, receiver-side...
Congestion control is vital in the streaming of a video sequence or clip, as network traffic varies unpredictably requiring constant adjustment of the transmission rate. Standard TCP-Friendly Rate Control (TFRC) wastes bandwidth and may react to congestion only when packet loss has already occurred. This paper presents a unicast transport protocol named RRB-SIMD for video streaming over the Internet,...
In this paper we introduce a reliable multicast concept for Device-to-Device (D2D) communication integrated into cellular network. In addition to the introduction of the basic concept, initial simulation results are presented as well. Clustering closely located devices which have local communication needs is a feasible and efficient way of solving the increasing data traffic requirements in the future...
The bandwidth in wireless networks is usually shared by a number of wireless nodes. The channel capacity depends on the number of accessing nodes, time, environmental conditions and location of nodes. Due to varying channel conditions, several issues are observed in the transmission of real time multimedia. The existing research on multimedia streaming can be classified into adaptation techniques...
Quality of Service (QoS) is still one of the major challenges for emerging services based on the Real-Time Transport Protocol (RTP) to deliver multimedia content. In the context of the Internet Protocol Television (IPTV) services, the synchronization of multimedia components plays an important role. In order to synchronize audio and video components of multimedia content, RTP timestamps have to be...
Real-time multimedia applications are time sensitive and require extra resources from the network, e.g. large bandwidth and big memory. However, wireless sensor networks (WSNs) suffer from limited resources such as computational, storage, and bandwidth capabilities. Therefore, sending real-time multimedia applications over WSNs can be very challenging. For this reason, we propose an adaptive multi-flows...
At present, video delivery over the wired or wireless packet-switched networks has been motivating a series of researches into video encoding, quality of service (QoS), multimedia communication, and service models of networks. However, most of researchers lack a suitable video delivery simulation platform closer to the real video delivery system. Different from other existing simulation mechanisms,...
New emerging technologies like Multimedia applications require timely delivery of information as compared to reliability. The most widely used protocols over transport layer are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). Both of these have some drawbacks while using in real time applications. TCP increases delay in transfer of data but achieving reliable transfer of data...
In network delivery of compressed video, packets may be lost if the channel is unreliable. But in some real time video applications such as video conferencing and push to talk systems, time delay and Quality of Service are both very important. This makes the use of error resilience and flow control techniques necessary. In this paper, we propose a new approach for real-time video stream transporting...
Nowadays, wireless networks have been deploying everywhere, with IEEE 802.11 as the most popular standard. However, wireless resources are scarce and wireless condition varies often. These limitations are crucial for applications with tight QoS requirements such as video or voice over IP. To cope with the problems, the standard has provided many features including multi-rate capability, which is the...
Video multicast over multirate wireless networks imposes great challenges in designing efficient rate adaptation schemes due to the multi-layer hierarchical video coding and heterogeneous fading across multicast receivers. We propose a cross-layer framework for rate selection in video multicast, where different video layers have diverse loss-rate QoS requirements and thus may employ different transmission...
In a traditional network stack, data from an application is transmitted in the order that it is received. An algorithm is proposed where information about the priority of packets and expiry times is used by the transport layer to reorder or discard packets at the time of transmission to optimise the use of available bandwidth. This can be used for video conferencing to prioritise important data. This...
This paper describes a methodology for building a reliable internet core router that considers the vulnerability of its electronic components to single event upset (SEU). It begins with a set of meaningful system level metrics that can be related to product reliability requirements. A specification is then defined that can be effectively used during the system architecture, silicon and software design...
This contribution addresses the case when live packet-switched video is used to enrich circuit-switched speech calls in mobile telephony. Circuit-switched and packet-switched transmissions generally operate on completely different transmission paths resulting in different QoS in terms of delay and loss rates. In this case, video and audio data recorded at the same time are not multiplexed together...
This paper explores the attributes, layering models and objective functions in cross layer designs for mobile ad hoc wireless networks by taking all the statistical characteristics and constraints from the physical (PHY) layer, media access control (MAC) layer and network (NET) layer into consideration. This paper reviews the entire network optimization across all the PHY/MAC/NET layers. At the PHY...
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