Real-time Transport Control Protocol Research Articles

Accelerating Information Security in Cloud Computing using a Novel Holomorphic Scheme

In the past decades, many security algorithms are invented and used in many applications for encryption and decryption of information or data. All the secure algorithms were uses the security key of size 8bit, 16bit, 32bit, 64bit and 128bits for encryption and decryption but literature survey says that higher the key size, higher the security. But the higher key size has number more bits and its requires more memory for storage and also to perform each and every bit through computational operations leads to more delay. To address these issues, the key and plain text are of 128bits and converted into integer. The converted integer values of key and plaintext are encrypted and decrypted using Holomorphic through Advanced Encryption Standard (AES). The research method is sophisticated and more protected through meaning fully lesser key in size and is accomplished for encryption in terms integer key and plaintext moderately than binary bits, therefore larger size of key and plaintext can be minimized and reduced the computational complexities. Finally the cipher text is uploaded into cloud through Real Time Transport Protocol (RTP) and Real Time Transport Control Protocol (RTCP) for storage. The main problem with continuous sharing of information into cloud is security attacks so in this research work, three different multimedia signals such as ECG, EEG and biomedical images are converted into integer and encrypted using AES. The stored data can access by the authorized users and can decode the information after decryption using AES.

A stealthy covert storage channel for asymmetric surveillance VoLTE endpoints

The widely adaptation of shared or public cloud computing raises an amount of concerns about privacy and information security. Establishing a covert channel is a feasible solution to the problem by hiding privacy data into the overt packets, so that network information eavesdropping will not detect the existence of user’s privacy data transmission. Enormous amount of VoLTE traffic makes it potentially available for stealthy message transfer. A novel covert storage channel based on RTCP (Realtime Transport Control Protocol) is proposed. Based on the analysis of the protocol and real VoLTE traffic data, the secret data is modulated by setting the jitter, or EHSNR(Extended Highest Sequence Number Received) and BLP (Bitmask of following Lost Packets) fields of the RTCP protocol, according to that the sender’s network packets are captured by the ward or not. The undetectability of the proposed covert channel is verified by the K-S test (Kolmogorov–Smirnov test) under adversary transmission model.

A Two-Way VoLTE Covert Channel With Feedback Adaptive to Mobile Network Environment

In the existing network covert channel research, the transmission of secret messages is one-way, lacking confirmation feedback on whether the secret message is successfully accepted. However, voice over LTE (VoLTE) has real-time interactive features, and the data packets between the sender and the receiver are transmitted in both directions, which facilitates the construction of a two-way covert channel with feedback. Therefore, we propose a two-way VoLTE covert channel adaptive to mobile network environment, which includes a sender-to-receiver covert timing channel that modulates covert message through actively dropping packets during the silence periods and a reverse covert storage channel that hides the acceptance of the covert message as feedback information into the feedback control information field of the real-time transport control protocol (RTCP) packet. The sender evaluates the current attack severity according to the feedback and adjusts the real-time parameters of the covert timing channel to weigh the robustness and other performance, thereby constructing a two-way covert channel adaptive to complex mobile network environments. Experimental results show that this solution can maintain voice quality which is an important performance indicator for VoLTE, while keeping the covert channel undetectable and robust.

Software Architecture for Streaming and Packet Loss Recovery Schemes in Road Monitoring Surveillance Camera Control Systems

Background/Objectives: This paper is about software architecture for security system design using surveillance control system. Various challenges are faced in surveillance control systems using monitoring cameras like packet loose, latency, quality of Closed-Circuit Television (CCTV) data, inconsistency and conflict among multiple sources etc. In this paper an attempt is made to solve some of the problems which are faced by surveillance control systems like road monitoring systems. Methods/Statistical analysis: Experimentation setup is proposed for the given software architecture and algorithm is developed for system. The Digital Video (DV) streaming system is a client-server model for the real-time delivery of DV streams over Internet Protocol (IP) networks based on Real-time Transport Protocol (RTP) / Real-time Transport Control Protocol (RTCP) protocol. The system includes two PCs, one at the server side and one at the client side, connected together with a Fast Ethernet LAN. Server has got a fire wire interface for capturing the data. Capturing device is connected to a DV converter (Canopus ADVC 1000) through a composite conversion cable. Converted DV stream is given to the server through IEEE 1394 interface. In the road monitoring systems such architecture is very much needed to recover from packet losses and it will help in reducing road accidents. Findings: In this paper attempt is made to propose the best possible architecture for arranging surveillance camera for getting the best streaming. Packet loss recovery schemes are proposed in the camera control systems which are used for monitoring roads and it can minimize the death rate using these monitoring camera systems. Applications/Improvements: The proposed software architecture is used in packet loss recovery systems of surveillance systems and very useful application for security systems which are used for monitoring road and it can reduce deaths caused due to road accidents.

Research of Network Camera Based on TMS320DM8168

The paper adopts the TMS320DM8168 processor to realize highly resolution network video surveillance system, and selects highly efficient video coding standard—H.264 compression algorithm to enhance the efficiency of coding data. The system based on embedded Linux system and the TMS320DM8168 processor is put forward which can be operated on 8-megapixel.At 60 frames per second in full frame rates, It is the video stream up to 2Mbps rate to provide resolution of 1920x1080 pixels. The Three video streams can be transferred in the 1000M Ethernet applied with the real time transport protocol (RTP) and real time transport control protocol (RTCP).

Technology of Vehicle Detail Parts Monitoring Based on CGI and IPv6

For the problem of the vehicle parts monitoring in the internet of vehicle, the service technology is proposed to access the detail components through the common gateway interface technology. The information transmission is managed using the common gateway interface and interface parts for docking, and the monitoring of detail parts is made by the bus integration monitoring. IPv6 virtual network address information transmission is used to avoid the hidden dangers of IPv4 address depletion, and the virtual and independent IPv6 address is transferred with the part number in the common gateway interface to achieve a wide geographical access. In order to achieve real-time information exchange and robustness, the real-time transport protocol and real-time transport control protocol is used to assure smaller transmission delay and less packet loss rate. The test results show that the average response time is no more than 85 nanoseconds and the accuracy of information transfer is not less than 98.2%, and the results meets the needs of the most environmental, and the program is viable in practice.

Real Time Communication over Modified UDP Protocol

The objective of this paper is to provide a uniform means of transmitting data (audio, video) to real time constraints over multicast or unicast network services and providing much enhanced experience of the Real-time communication over the existing systems. In this paper we will explore usage of the dominating technologies in the real time systems and shows how to design and develop communications within such systems. In particular, it proposes reliable protocol over UDP transport layer that gains less than one millisecond average round trip time (RTT) and processes dozens of thousands of messages every second at each and every port in network. This paper presents the current internet protocols: Real-time Transport Protocol (RTP), RTP Control Protocol (RTCP) and Real-time Streaming Protocol (RTSP). Together these protocols may provide controlled delivery of multimedia traffic over the Internet. The feedback mechanisms for the quality of service (QoS) monitoring, such as delay, jitter and packet loss calculations, are described in detail.

Periodic Feedback Control for Streaming 3D Videos in Last-Generation Cellular Networks

Transmitting stereoscopic 3D videos on cellular networks is a challenging issue because effective scheduling is necessary for reproducing frames continuously, without wasting bandwidth in frame retransmissions. We propose a scheduling algorithm that allows the user of interactive videos to perform VCR-like actions other than the usual playback (e.g., pause or fast forward at different speeds) with less problems. The algorithm exploits periodic feedback packets coming from the client to the server through the Real-time Transport Control Protocol and communicating the client buffer fill level. A P-controller uses this information to dynamically regulate the scheduled bitrate. The approach is compared with classical scheduling algorithms. Different 3D video formats and quality degrees are considered. Results show that the simple controller is very effective and outperforms the other schedulers in all the analyzed scenarios.

Development of a Network Adaptive H.264/AVC Medical Video Transmission System

Network high definition video applications such as video calls and videoconferences have been increasingly developed recent years. When congestions occur and network environment becomes unstable, the network applications have to deal with such issues like delay, packet loss and bandwidth constraints. In this paper, we describe a high definition H.264/AVC video transmission system with network adaptive function based on real-time transport protocol (RTP) and real-time transport control protocol (RTCP). The adaptive system is composed of three components: 1) transmission management (TM) which focuses on network state monitoring, bandwidth allocating and transmission rate control; 2) a feedback-based adaptive forward error correction (AFEC) scheme using Reed Solomon (RS) code; 3) adaptive video encoding (AVE) that adjusts the bitrate of video streaming according to the bandwidth constraint information offered by TM. In our paper, delay, packet loss and bandwidth constraints are generated by the network bottleneck and congestions. Hence, the main idea is to enhance the bitrate of video streaming adapt to the bandwidth constraints. In addition, we integrate AFEC with RTCP to feedback bandwidth status and make the adaptive video streaming more robust under packet loss environment. Here we call it interactive adaptive forward error correction (IAFEC). The test result shows IAFEC can be adaptive to strict bandwidth constraints and high packet loss rates.

Transmission control of Variable-Bit-Rate video streaming in UMTS networks

Variable-Bit-Rate (VBR) video transmission over UMTS networks is assuming an ever growing importance, then scheduling VBR data over wireless channels attracts great interest today. This paper proposes an on-line PI control algorithm for easily and suitably scheduling interactive multimedia transmission of VBR video streams. It dynamically adapts the transmitted bit rate to the user's actions that can suddenly and highly modify the client buffer level. The Real-time Transport Control Protocol (RTCP) packets periodically feed back the buffer level to the server for keeping the free buffer space at 50% of its capacity, so as to prevent both buffer overflow and underflow. Numerical results of different simulated scenarios show the effectiveness of the proposed controller in comparison with two well-established algorithms, enhanced with RTCP information.

MAC/PHY 정보를 이용한 실시간 멀티미디어 서비스의 QoS 제어 방식의 표준화를 위한 연구

본 논문에서는 모바일 환경에서 클라이언트의 MAC/PHY 파라미터를 이용하여 QoS(Quality of Service)를 제어하는 프로토콜과 그 효과에 대해 연구한다. 클라이언트는 자신의 MAC/PHY 채널 상황을 측정하여 서버측에 피드백하고 서버는 이를 이용하여 앞으로의 클라이언트의 상황을 예측하고 비트율 제어를 수행하는 방법을 제안한다. 제안 방법을 통해 기존의 RTCP(Real-time Transport Control Protocol)를 이용한 비트율 제어 방법보다 정확하게 가용 비트율을 예측할 수 있다. 정확한 가용 비트율 예측을 통해 비트율 낭비와 전송지연을 감소시키고, 비디오 전송 시 화질을 향상시킬 수 있다. WiMAX 환경에서 실제로 측정된 클라이언트의 CINR(Carrier to Interface Noise Ratio) 데이터를 활용하여 그 효과를 보여준다. In this paper, we study QoS(Quality of Service) control protocols and the effect using MAC/PHY parameters of client device in mobile network. We proposes the way of controling the bit-rate by estimating the channel condition of the client with measured MAC/PHY parameters which is sent from the client. With the proposed method, more accurate available bit-rate can be estimated compared to conventional protocol, RTCP(Real-time Transport Control Protocol). The accurate bit-rate estimation can decrease wasted bit-rate and transport delay. In the result of the advantages, the transported video quality can be enhanced. In this paper, we show the effects of enhancement using client's the field data measured in WiMAX.

Error Resistant Real-Time Transport Control Protocol

Problem statement: Real-time Transport Control Protocol (RTCP) protocol has been the subject of various criticisms due its problematic performance in large-scale networks. S-RTCP is a protocol with high potential as it has proved to be able to solve many problems of RTCP. It has numerous flaws on its own. This study aimed at dealing with flaws of S-RTCP and improving it in terms of stability and packet loss. Approach: A new proposed scheme was designed. Modifications included designing multi-manager scheme, improving parent-seeking procedures, reducing distribution of request packets, reforming the design to be independent from TTL, adding methods to check on sanity of manager nodes. This study considered packet loss ratio of below 2% as desirable. Results: ER-RTCP comparing to legacy RTCP in terms of packet loss using NS-2 in four different scenarios revealed improvements between 73 and 88% for various scenarios. It also kept packet loss rate below 2% for all scenarios. Comparison of ER-RTCP to S-RTCP showed that based on different α (stability of each single manager) values, ER-RTCP was more stable as it showed more resistance to entire scheme breakdown (β). ER-RTCP’s parent-seeking procedure, as modeled scenario revealed a packet generation reduction of 97%, compared to S-RTCP’s. In occurrence of parent AG leave or loss, ER-RTCP reduced request packet generation by 95%. Allowance of AG dismissing in ER-RTCP, avoided occurrence of packet loss, as sample scenario showed S-RTCP experiencing packet loss of 3.5% while ER-RTCP kept packet loss at zero in theory. Conclusion: Proposed design improved S-RTCP in terms of reduction of packet loss and stability.

Adaptive Video Streaming Using Bandwidth Estimation for 3.5G Mobile Network

Currently deployed mobile networks including High Speed Downlink Packet Access (HSDPA) offer only best-effort Quality of Service (QoS). In wireless best effort networks, the bandwidth variation is a critical problem, especially, for mobile devices with small buffers. This is because the bandwidth variation leads to packet losses caused by buffer overflow as well as picture freezing due to high transmission delay or buffer underflow. In this paper, in order to provide seamless video streaming over HSDPA, we propose an efficient real-time video streaming method that consists of the available bandwidth (AB) estimation for the HSDPA network and the transmission rate control to prevent buffer over-flows/underflows. In the proposed method, the client estimates the AB and the estimated AB is fed back to the server through real-time transport control protocol (RTCP) packets. Then, the server adaptively adjusts the transmission rate according to the estimated AB and the buffer state obtained from the RTCP feedback information. Experimental results show that the proposed method achieves seamless video streaming over the HSDPA network providing higher video quality and lower transmission delay.

Dynamic load balancing through real-time transport feedback control for VoIP QoS

The Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP) can be utilised for dynamic load balancing over routing diversity. We implemented and tested an RTP-VoIP based application between two internet nodes located in Egypt and the USA. The collected real-time statistics were then used to emulate multiple paths between the source and destination nodes. In this study, the source node utilises RTCP to distribute the voice load between the two paths used for routing diversity. The results show that routing diversity along with load-balancing provide QoS that is much higher than if a single bad path is used and close to what is provided by a path with less traffic impairment.

An MPEG-4 Compliant Interactive Multimedia Streaming Platform Using Overlay Networks

This paper presents a multimedia streaming platform for efficiently transmitting MPEG-4 content over IP networks. The platform includes an MPEG-4 compliant streaming server and client, supporting object-based representation of multimedia scenes, interactivity, and advanced encoding profiles defined by the ISO standard. For scalability purposes, we employ an application-layer multicast scheme for media transmission using overlay networks. The overlay network, governed by the central entity of the network distribution manager, is dynamically deployed according to a set of pre-defined criteria. The overlay network supports both broadcast delivery and video-ondemand content. The multimedia streaming platform is standards-compliant and utilizes widespread multimedia protocols such as MPEG-4, real-time transport protocol, real-time transport control protocol, and real-time streaming protocol. The design of the overlay network was architected with the goal of transparency to both the streaming server and the client. As a result, many commercial implementations that use industry-standard protocols can be plugged into the architecture relatively painlessly and can enjoy the benefits of the platform.

Control Based Mobile Ad-hoc Networks for Video Communications

This paper presents a feedback control scheme for transmission of video signals over mobile ad-hoc channels. The scheme is a combination of cross-layer (local) feedback and receiver feedback. The receiver feedback is based on the real time transport control protocol (RTCP), which is designed to provide an end-to-end feedback assessment of multihop transmission channels by tracking packet receptions in synchronization with the video frame rate. The control packet carries an overlapping bit-pattern in order to cope with losses on the reverse link. Assisting the receiver feedback is a local feedback, which aims at controlling the packet transmission flow with respect to the ad-hoc routing characteristics. The combined feedback scheme, together with a bitrate control and packet-loss compensation strategy, have been shown to be very effective in improving the ad-hoc network reliability for transmission of H.264/RTP/UDP/IP packets over multihop fading channels.

E-VLBI 구현을 위한 네트워크 모델 설계

e-VLBI was invented to enhance the efficiency of VLBI (Very-Long-Baseline Interferometry) system by transmitting the data via high speed network. Korean VLBI Network (KVN) has a plan to construct e-VLBI system named e-KVN. High speed backbone network and efficient network model are essential to implement successful e-VLBI system. This paper introduces a network model based on PC cluster technology. The present status of high speed backbone network in Korea is overviewed. We suggest that the network link via Korea Advanced Research Network (KOREN) is one of feasible way for e-KVN. We also describe the principles of e-VLBI and protocol for network transmission such as VSI-E (VLBI Standard Interface - Electronic), RTP (Real-Time Transport Protocol) and RTCP (Real-Time Transport Control protocol).

Intermedia synchronization for videoconference over IP

This paper deals with the problem of audio and video synchronization issues for real-time audio-visual communication over IP-based networks. Starting from the real-time transport protocol (RTP) specifications (Schulzrinne, 1995), it provides an accurate description on how to recover a reliable absolute time reference for audio and video signals from header information in RTP and RTP control protocol packets. Such temporal informations allow to synchronize both media within acceptable perceptual bounds for reconstruction at any receiver end, in a possibly multi-point videoconference. This may occur independently of the fact that all (audio/video) packets reach destination, or that multiple replications of such packets arrive at destination. The proposed solution does not take into account the possible different delays that may be introduced between the different communication points, thus resulting eventually in potential jitters in the reconstruction of the audio-visual streams between different receivers. Each receiver handles its reconstruction independently of any transmission/processing delay. In order to ensure a better quality of the reconstructed material, priority is given to audio information. If the audio stream anticipate the video stream, the receiver simply discards video packets. Conversely, when video is ahead of audio information, the video decoding stage is interrupted till audio information arrives. Experimental simulations over a LAN have demonstrated the validity of the proposed approach.