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Many multimedia applications employ transmission control protocol (TCP) to send video packets. TCP ensures reliable transmission but experiences significant delays. Various techniques have been proposed to enhance the performances, for instance by controlling TCP window. This paper proposes a split-routed acknowledgement packet that is aimed to reduce acknowledgement delay. The simulation evaluations...
Since complex network environment makes the video data with network resources change quickly, video transmission must possess the ability to adapt to the change of network status. After studying the video stream congestion question as well as the network adaptive transmission control and the transmission characteristic of RTP (Real-time Transport Protocol) RTCP (Real-time Transport Control Protocol),...
Quality of Service (QoS) is imperative for multimedia real-time applications to ensure optimal use of the communication channels and deliver Video Conference (VC) with acceptable quality. Several algorithms have been proposed in the literature to optimize the use of the bandwidth when both VC and Internet browsing are operating at the same time. In this paper we apply three of these methods, Guaranteed...
Making the network programmable simplifies network management and enables network innovations. The Recursive Inter Network Architecture (RINA) is our solution to enable network programmability. ProtoRINA is a user-space prototype of RINA and provides users with a framework with common mechanisms so a user can program recursive-networking policies without implementing mechanisms from scratch. In this...
Avionics Full Duplex Ethernet (AFDX) is a well-established backbone network with wide usage in all current aircrafts, with its first usage onboard the Airbus A380. It was initially defined by Airbus and internationally standardized by ARINC (664 Part 7). The key features are 802.3 Ethernet compatibility, static routing based on a virtual link concept, reserved bandwidth with granted bandwidth allocation...
Over the past years, we have witnessed an explosive growth in the use of the multimedia applications such as audio streaming with mobile and static devices. Audio streaming applications demand new approaches to audio transmissions to meet the growing traffic volume, quantity and quality of audio traffic, and users' needs. This paper studies network coding which is a promising paradigm that has the...
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,...
When a mobile host transmitting real-time video traffic using DCCP CCID3 moves between two wireless LAN access points with different congestion situations, it is difficult to quickly change the transmission rate according to the changes of communication environment because controlling the sending rate depends on feedback information. To solve this problem, we recently proposed a predictive rate control...
As the escalating demand for bandwidth continues, 100-Gigabit networks mark the new frontier to be ventured by networking equipment vendors. Network processors (NPUs), fast emerging as key components in 10-Gigabit platforms, will also empower the 100-Gigabit platforms. However, packet processing at these line rates presents significant challenges to network processor designs. The most important driving...
On QoS of live broadcast, proposes an congestion control strategy based on multicast transmission for terminals, i.e. the sender adopts RTP/RTCP adaptive control strategy, and variable increase and decrease method is used to adjust sending rate, while the receiver sets the buffer to adjust the data to play order of playback and to maintain the matching rate and effective bandwidth at the same time...
Mobility support in IPv6 networks (MIPv6) and Fast Handovers for Mobile IPv6 (FMIPv6) are already released as RFCs for many years. While the former focuses on maintaining the network connections when a mobile node handovers across subnets, the latter is built based on the former concept, but focuses on reducing the packet loss and latency inherent to the handover process. Performance study of both...
Urban facility interconnection networks require a robustness and reliability usually found in public networks or private cable and radio networks in licensed bands. Both factors mean an increase in costs, which also increase as the size of the network increases. The recent rise in use of wireless technologies in open wavebands has attracted the interest of the industry in the spread of these types...
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...
High-bandwidth QoS sensitive services such as large scale video surveillance generally depend on provisioned capacity delivered by circuit-switched technology such as SONET/SDH. Yet development in layer 2 protocol sets and manageability extensions to Ethernet standards propose layer 2 packet switching technology as a viable, cheaper alternative to SONET/SDH. Layer 2 switched networks traditionally...
In this paper, we propose a new variant of TCP with an early retransmission scheme as an enhancement to make it more suitable for streaming media. We call this new protocol TCP-ER. We performed extensive ns2 simulations to show that: 1) The early retransmission scheme can considerably reduce the number of retransmission timeouts and packet delay jitter in a variety of network environments. 2) TCP-ER...
As part of the resource allocation mechanisms, each router must implement some queuing discipline that governs how packets are buffered while waiting to be transmitted. Various queuing disciplines can be used to control which packets get transmitted (bandwidth allocation) and which packets get dropped (buffer space). The queuing discipline also affects the latency experienced by a packet, by determining...
Real-time streaming over wireless networks is a challenging proposition due to the highly variable nature of wireless links and the resource-poor nature of mobile device. In such a context, transmission control schemes have to dynamically adapt both to the application requirements and to the channel conditions. In this paper, we propose an adaptive cross-layer quality-of-service (QoS) scheme for wireless...
Today, as the Real-time Transport Protocol (RTP) is used more and more widely in the streaming media services such as real-time video in the internet. The algorithm of RTP adaptive transmission control is an important factor to guarantee the quality of video transmission services. So, much more attention should be paid to the research on the algorithm. In this paper, on the base of researching on...
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...
We present in this paper a simulation-based comparison of one of the best known multicast congestion control schemes - TFMCC - against our proposed adaptive smooth multicast protocol (ASMP). ASMP consists of a single-rate multicast congestion control, which takes advantage of the RTCP sender (SR) and receiver reports (RR) in order to adjust the sender's transmission rate in respect of the network...
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