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ITC/USA 2005 Conference Proceedings / The Forty-First Annual International Telemetering Conference and Technical Exhibition / October 24-27, 2005 / Riviera Hotel & Convention Center, Las Vegas, Nevada / To monitor ocean resources and environment, we develop an ocean stereo telemetry system built on a PC104 industrial control computer, which is carried by a buoy on the ocean. All monitoring instruments communicate with the computer by six serial ports in virtue of time division multiple access and are synchronized by GPS time to collect data. All monitoring data is archived and compressed in format of RINEX (Receiver Independent Exchange). The uploading data and downloading control command to and from monitoring center is transferred by Iridium communication in automatic retransmission request and broken-point continuing mechanism.

隨著網路的進步與發展，許多新興的資訊服務，如影音資訊，在網路上傳輸時並未要求封包都送達，不同的封包有不同的重要性，例如 MPEG的關鍵畫面(I -Frame)就比其他畫面重要。常用的傳輸層通訊協定中，UDP與TCP都對所有的封包一視同仁，前者不做任何保證，而後者雖可保證所有封包的送達，但效率較差。 本研究提出一個新的TCP，”Partial-Reliable TCP”，使用選擇性重傳機制，配合應用程式的需求，對指定的封包提供遞送保護。當封包遺失時，只重傳保護的封包，可減少額外的網路資源消耗，並提升服務的品質。 此外，我們提出Single-Side的版本，接收端可以使用一般的 TCP，在封包傳送時，讓接收端以為封包都是無誤傳達的，在server-client架構的網路服務中，只有伺服器端必須使用我們的Partial-Reliable TCP，大幅提高本通訊協定的可行性。 最後我們利用網路模擬工具NS-2來模擬實際網路環境，將我們的方法與現行的通訊協定在可解畫面封包數、PSNR值及額外耗用的網路資源三個參數做比較。我們使用兩個 Video 影像作為傳輸標的，在高遺失率的有線與無線網路的環境中進行實驗。當傳輸時間限制很短時，（例如影像會議的應用），在有線的環境中， Basic PR-TCP比TCP Reno、TFRC最少增加約18%的可解畫面封包數，比UDP、TFRC及TCP Reno的PSNR值最少高出約15%，比TCP Reno及TFRC最少節省了12%的頻寬資源，Single-Side PR-TCP比Basic PR-TCP的PSNR值約低了11%，額外耗用的頻寬約多出10%。在無線的環境中，Basic PR-TCP比TCP Reno、TFRC最少增加約19%的可解畫面封包數，比UDP、TFRC及TCP Reno的PSNR值最少高出約20%，比Single-Side PR-TCP、TCP Reno及TFRC最少節省了15%的頻寬資源，Single-Side PR-TCP比Basic PR-TCP的PSNR值約低了14%。當傳輸時間限制較充裕時 (例如VoD應用)，Basic PR-TCP雖然比TCP Reno及TFRC降低了約3%的PSNR值，但是最少能節省8%的頻寬耗費，Single-Side PR-TCP的PSNR值跟Basic PR-TCP相近，但是額外耗用約5%的頻寬資源。 / With the advance of computer and communication networks, many new information services over IP-based networks such as video streaming and VoIP (Voice over IP) are growing rapidly. These services can tolerate some packets lost in transmission without too much damage to their quality. The content carried in the packets of these services is not equally important in their replay processes. For example, key frames (e.g. I-Frames) of a video encoded in MPEG format are more important than others. The loss of I-frames may have a large impact to the quality of the transmitted video, while the loss of other types of frames may only have nominal damage. Unfortunately, the two most popular transport protocols, UDP and TCP, treat all packets equally without any discrimination. TCP guarantees the delivery of all packets, while UDP doesn't. TCP may waste too much resource to guarantee the delivery of unimportant packets, while UDP may fail to deliver too many important packets. This thesis proposes a new TCP protocol, named Partial-Reliable TCP (PR-TCP), which applies selective retransmission strategy to provide delivery guarantee to the selected packets designated by the application programs. In this way, we can save bandwidth consumption and reduce the average delivery time without significant quality degradation. In fact, if the delivery of an object requires a stringent delivery time, the reduction of average delivery time may also lead to the reduction of abandoned packets at the receiver end. We propose two different versions of PR-TCP, Basic PR-TCP and Single-Side PR-TCP. Basic PR-TCP requires both ends of a connection to adopt PR-TCP while Single-Side PR-TCP only requires the sender end to adopt it. It is much easier to deploy Single-Side PR-TCP on the client-server systems where only servers need to use PR-TCP. Finally, we use NS-2 network simulator to evaluate our PR-TCP against TCP Reno, TFRC and UDP. Two video stream samples are used for video sources. Three quality parameters are evaluated: wasted bandwidth consumption, PSNR, and the number of packets in decodable frames. Under heavy loaded wired network and short delay bound (<0.8 sec.), the simulation shows that Basic PR-TCP can outperform TCP Reno and TFRC in the number of packets in decodable frames by at least 18%. It can outperform TCP Reno, TFRC, and UDP in PSNR by at least 12%. The performance of Single-Side PR-TCP is less then Basic PR-TCP in terms of PSNR by 10%, and it consumes larger bandwidth by 8%. Under wireless environments where error rate is high, the simulation shows that Basic PR-TCP can outperform TCP Reno and TFRC in the number of packets in decodable frames by at least 19% as well as wasted transmission overhead by at least 15%. It can also outperform TCP Reno, TFRC, and UDP in PSNR by at least 20%. The performance of Single-Side PR-TCP is less then Basic PR-TCP in terms of PSNR by 14%, and it consumes larger bandwidth by 10%. Under large delay bound (>8 sec.), the quality (PSNR) of the video transmitted using Basic PR-TCP is downgraded by only 3%, while it can save network bandwidth by 8%. The performance of Single-Side PR-TCP is about the same as Basic PR-TCP in terms of PSNR, but it consumes slightly larger bandwidth by 5%.

"Packet loss can be detrimental to real-time interactive video over lossy networks because one lost video packet can propagate errors to many subsequent video frames due to the encoding dependency between frames. Feedback-based error control techniques use feedback information from the decoder to adjust coding parameters at the encoder or retransmit lost packets to reduce the error propagation due to data loss. Feedback-based error control techniques have been shown to be more effective than trying to conceal the error at the encoder or decoder alone since they allow the encoder and decoder to cooperate in the error control process. However, there has been no systematic exploration of the impact of video content and network conditions on the performance of feedback-based error control techniques. In particular, the impact of packet loss, round-trip delay, network capacity constraint, video motion and reference distance on the quality of videos using feedback-based error control techniques have not been systematically studied. This thesis presents analytical models for the major feedback-based error control techniques: Retransmission, Reference Picture Selection (both NACK and ACK modes) and Intra Update. These feedback-based error control techniques have been included in H.263/H.264 and MPEG4, the state of the art video in compression standards. Given a round-trip time, packet loss rate, network capacity constraint, our models can predict the quality for a streaming video with retransmission, Intra Update and RPS over a lossy network. In order to exploit our analytical models, a series of studies has been conducted to explore the effect of reference distance, capacity constraint and Intra coding on video quality. The accuracy of our analytical models in predicting the video quality under different network conditions is validated through simulations. These models are used to examine the behavior of feedback-based error control schemes under a variety of network conditions and video content through a series of analytic experiments. Analysis shows that the performance of feedback-based error control techniques is affected by a variety of factors including round-trip time, loss rate, video content and the Group of Pictures (GOP) length. In particular: 1) RPS NACK achieves the best performance when loss rate is low while RPS ACK outperforms other repair techniques when loss rate is high. However RPS ACK performs the worst when loss rate is low. Retransmission performs the worst when the loss rate is high; 2) for a given round-trip time, the loss rate where RPS NACK performs worse than RPS ACK is higher for low motion videos than it is for high motion videos; 3) Videos with RPS NACK always perform the same or better than videos without repair. However, when small GOP sizes are used, videos without repair perform better than videos with RPS ACK; 4) RPS NACK outperform Intra Update for low-motion videos. However, the performance gap between RPS NACK and Intra Update drops when the round-trip time or the intensity of video motion increases. 5) Although the above trends hold for both VQM and PSNR, when VQM is the video quality metric the performance results are much more sensitive to network loss. 6) Retransmission is effective only when the round-trip time is low. When the round-trip time is high, Partial Retransmission achieves almost the same performance as Full Retransmission. These insights derived from our models can help determine appropriate choices for feedback-based error control techniques under various network conditions and video content. "

Doctor of Philosopy (PhD) / Wireless communication always attracts extensive research interest, as it is a core part of modern communication technology. During my PhD study, I have focused on two research areas of wireless communication: IEEE 802.11 network performance analysis, and wireless cooperative retransmission. The first part of this thesis focuses on IEEE 802.11 network performance analysis. Since IEEE 802.11 technology is the most popular wireless access technology, IEEE 802.11 network performance analysis is always an important research area. In this area, my work includes the development of three analytical models for various aspects of IEEE 802.11 network performance analysis. First, a two-dimensional Markov chain model is proposed for analysing the performance of IEEE 802.11e EDCA (Enhanced Distributed Channel Access). With this analytical model, the saturated throughput is obtained. Compared with the existing analytical models of EDCA, the proposed model includes more correct details of EDCA, and accordingly its results are more accurate. This better accuracy is also proved by the simulation study. Second, another two-dimensional Markov chain model is proposed for analysing the coexistence performance of IEEE 802.11 DCF (Distributed Coordination Function) and IEEE 802.11e EDCA wireless devices. The saturated throughput is obtained with the proposed analytical model. The simulation study verifies the proposed analytical model, and it shows that the channel access priority of DCF is similar to that of the best effort access category in EDCA in the coexistence environment. The final work in this area is a hierarchical Markov chain model for investigating the impact of data-rate switching on the performance of IEEE 802.11 DCF. With this analytical model，the saturated throughput can be obtained. The simulation study verifies the accuracy of the model and shows the impact of the data-rate switching under different network conditions. A series of threshold values for the channel condition as well as the number of stations are obtained to decide whether the data-rate switching should be active or not. The second part of this thesis focuses on wireless cooperative retransmission. In this thesis, two uncoordinated distributed wireless cooperative retransmission strategies for single-hop connection are presented. In the proposed strategies, each uncoordinated cooperative neighbour randomly decide whether it should transmit to help the frame delivery depending on some pre-calculated optimal transmission probabilities. In Strategy 1, the source only transmits once in the first slot, and only the neighbours are involved in the retransmission attempts in the subsequent slots. In Strategy 2, both the source and the neighbours participate in the retransmission attempts. Both strategies are first analysed with a simple memoryless channel model, and the results show the superior performance of Strategy 2. With the elementary results for the memoryless channel model, a more realistic two-state Markov fading channel model is used to investigate the performance of Strategy 2. The simulation study verifies the accuracy of our analysis and indicates the superior performance of Strategy 2 compared with the simple retransmission strategy and the traditional two-hop strategy.

Traditional TCP when applied in wireless networks may encounter two limitations. The first limitation is the higher bit error rate (BER) due to noise, fading, and multipath interference. Because traditional TCP is designed for wired and reliable networks, packet loss is mainly caused by network congestions. As a result, TCP may decrease congestion window inappropriately upon detecting a packet loss. The second limitation is about the packet scheduling, which mostly does not consider wireless characteristics. In this Thesis, we propose a local retransmission mechanism to improve TCP throughput for wireless networks with higher BER. In addition, we measure the packet loss ratio (PLR) to adjust the queue weight such that the available bandwidth for each queue can be changed accordingly. In our mechanism, the queue length is used to determine whether there is a congestion in wireless networks. When the queue length exceeds a threshold, it indicates that the wireless networks may have congestion very likely. We not only propose the dynamic weight-adjustment mechanism, but also solve the packet out-of-sequence problem, which results form when a TCP flow changes to a new queue. For the purpose of demonstration, we implement the proposed weight-adjustment mechanisms on the Linux platform. Through the measurements and discussions, we have shown that the proposed mechanisms can effectively improve the TCP throughput in wireless networks.

Doctor of Philosopy (PhD) / Wireless communication always attracts extensive research interest, as it is a core part of modern communication technology. During my PhD study, I have focused on two research areas of wireless communication: IEEE 802.11 network performance analysis, and wireless cooperative retransmission. The first part of this thesis focuses on IEEE 802.11 network performance analysis. Since IEEE 802.11 technology is the most popular wireless access technology, IEEE 802.11 network performance analysis is always an important research area. In this area, my work includes the development of three analytical models for various aspects of IEEE 802.11 network performance analysis. First, a two-dimensional Markov chain model is proposed for analysing the performance of IEEE 802.11e EDCA (Enhanced Distributed Channel Access). With this analytical model, the saturated throughput is obtained. Compared with the existing analytical models of EDCA, the proposed model includes more correct details of EDCA, and accordingly its results are more accurate. This better accuracy is also proved by the simulation study. Second, another two-dimensional Markov chain model is proposed for analysing the coexistence performance of IEEE 802.11 DCF (Distributed Coordination Function) and IEEE 802.11e EDCA wireless devices. The saturated throughput is obtained with the proposed analytical model. The simulation study verifies the proposed analytical model, and it shows that the channel access priority of DCF is similar to that of the best effort access category in EDCA in the coexistence environment. The final work in this area is a hierarchical Markov chain model for investigating the impact of data-rate switching on the performance of IEEE 802.11 DCF. With this analytical model，the saturated throughput can be obtained. The simulation study verifies the accuracy of the model and shows the impact of the data-rate switching under different network conditions. A series of threshold values for the channel condition as well as the number of stations are obtained to decide whether the data-rate switching should be active or not. The second part of this thesis focuses on wireless cooperative retransmission. In this thesis, two uncoordinated distributed wireless cooperative retransmission strategies for single-hop connection are presented. In the proposed strategies, each uncoordinated cooperative neighbour randomly decide whether it should transmit to help the frame delivery depending on some pre-calculated optimal transmission probabilities. In Strategy 1, the source only transmits once in the first slot, and only the neighbours are involved in the retransmission attempts in the subsequent slots. In Strategy 2, both the source and the neighbours participate in the retransmission attempts. Both strategies are first analysed with a simple memoryless channel model, and the results show the superior performance of Strategy 2. With the elementary results for the memoryless channel model, a more realistic two-state Markov fading channel model is used to investigate the performance of Strategy 2. The simulation study verifies the accuracy of our analysis and indicates the superior performance of Strategy 2 compared with the simple retransmission strategy and the traditional two-hop strategy.

The profileration of multimedia applications, such as streaming video, teleconferencing, and interactive gaming has created a tremendous challenge for the traditional transport protocols of the Internet – UDP and TCP. Specifically, many multimedia applications are examples of soft real-time applications. They have often relatively stringent require- ments in terms of delay and delay jitter, but typically tolerate a limited packet loss rate. In recognition of the transport service requirements of soft real-time applications, this thesis studies the feasibility of using retransmission based, partially reliable trans- port protocols for these applications. The thesis studies ways of designing retransmis- sion based, partially reliable transport protocols that are congestion aware and TCP com- patible. Furthermore, the transport protocols should provide a service that, in terms of performance metrics such as throughput, delay, and delay jitter, are suitable for soft real- time applications. The thesis work comprises the design, analysis, and evaluation of two retransmission based, partially reliable transport protocols: PRTP and PRTP-ECN. Extensive simulations have been carried out on PRTP as well as PRTP-ECN. These sim- ulations have in part been complemented by some theoretical analysis. The results of the simulations and the analysis suggest that substantial reductions in delay jitter and improvements in throughput can indeed be obtained with both PRTP and PRTP-ECN as compared to TCP. While PRTP reacted too slowly to congestion to be TCP-friendly and altogether fair, PRTP-ECN was found to be both TCP-friendly and reasonably fair. The thesis work also comprises an extensive survey on retransmission based, par- tially reliable transport protocols. Based on this survey, we have proposed a taxonomy for these protocols. The taxonomy considers two dimensions of retransmission based, partially reliable transport protocols: the transport service, and the error control scheme.

IPTV (Internet Protocol Television), a new and modern concept of emerging technologies with focus on providing cutting edge high-resolution television, broadcast, and other fascinating services, is now easily available with only requirement of high-speed internet. Everytime a new technology is made local, it faces tremendous problems whether from technological point of view to enhance the performance or when it comes down to satisfy the customers. This cutting edge technology has provided researchers to embark and play with different tools to provide better quality while focusing on existing tools. Our target in dissertation is to provide a few interesting facets of IPTV and come up with a concept of introducing an imaginary cache that can re-collect the packets travelling from streaming server to the end user. In the access node this cache would be fixed and then on the basis of certain pre-assumed research work we can conclude how quick retransmission can take place when the end user responds back using RTCP protocol and asks for the retransmission of corrupted/lost packets. In the last section, we plot our scenario of streaming server on one side and client, end user on the other end and make assumption on the basis of throughput, response time and traffic.

Virginia Tech researchers have developed a broadband channel sounder that can measure channel quality while a wireless network is in operation. Channel measurements from the broadband sounder hold the promise of improving TCP/IP performance by trigging configuration changes in an adaptive data link layer protocol. We present an adaptive data link layer protocol that can use different levels of forward error correction (FEC) codes and link layer automatic retransmission request (ARQ) to improve network and transport layer performance. Using a simulation model developed in OPNET, we determine the effects of different data link layer protocol configurations on TCP/IP throughput and end-to-end delay using a Rayleigh fading channel model. Switching to higher levels of FEC encoding improves TCP/IP throughput for high bit error rates, but increases end-to-end delay of TCP/IP segments. Overall TCP/IP connections with link layer ARQ showed approximately 150 Kbps greater throughput than without ARQ, but lead to the highest end-to-end delay for high bit error rate channels. Based on the simulation results, we propose algorithms to maximize TCP/IP throughput and minimize end-to-end delay using the current bit error rate of the channel. We propose a metric, carrier-to-interference ratio (CIR) that is calculated from data retrieved from the broadband channel sounder. We propose algorithms using the carrier-to-interference ratio to control TCP/IP throughput and end-to-end delay. The thesis also describes a monitor program to use in the broadband wireless system. The monitor program displays data collected from the broadband sounder and controls the settings for the data link layer protocol and broadband sounder while the network is in operation. / Master of Science