不論從寬頻時代的來臨，乃至於無線網路的普及，影音串流的傳輸都被視為非常重要的應用，應用的類型從主從式架構的視訊隨選系統，更擴展到像視訊交談系統的端對端架構。以現在最普遍的802.11b無線網路而言，面臨的最大問題為：(1)頻寬的不足，以及(2)不可靠的傳輸媒介，這些問題使得無線網路上的串流控制異於有線網路，由於無線網路與有線網路的傳輸特性不同，有線網路上常見的封包遺失是由於網路擁塞，但無線網路通常是傳輸的錯誤使然，在主從式架構下串流的傳輸是由有線端流向無線端，而在端對端架構下串流是從無線端經由有線端到達另一個無線端，發送端的無線網路會使得中間的有線網路特性無法呈現，因此在端對端的架構下更突顯出串流控制的困難度。此外，在影音串流壓縮技術方面，MPEG-4是現在最熱門的影音串流壓縮技術，擁有高壓縮比、可變解析度、可調位元率、容錯性編碼技術等優點，非常適合用於網路傳輸，因此本論文的目的在實作一套適用於無線網路環境的端對端MPEG-4串流系統，為了解決兩端之無線網路對串流傳輸的影響，我們提出來的控制機制利用無線網路通訊協定中MAC層所提供的資訊，透過分析完成判斷封包遺失類型和估算可用頻寬的功能，並藉由這些結果來調整串流的資料和控制串流資料的傳輸，我們提出來的控制機制包含下面四個方向：(1)解決在無線發送端可能會因為無線傳輸發生錯誤而產生的封包遺失；(2)視訊串流使用分層機制，根據當時的網路狀況決定傳輸的視訊層；(3)避免因為I-VOP遺失時造成之後的VOP解碼錯誤，所以改良重傳機制，降低對使用者的影響；(4)控制傳輸時的速率，一方面避免傳輸速率過快超出網路路徑的負載而產生擁塞現象，一方面可用頻寬太小時，可以動態的決定P-VOP是否要丟棄。系統實作時，有關於取得無線網路通訊協定中MAC層的資訊是透過NDIS介面進行操作。

　　No matter boardband networks had been coming, and even wireless networks popularize, audio and video streaming is considered a very important application. The original application category is extended from client/server architecture like VOD system to end-to-end architecture like video conference system. For general wireless network based on 802.11b standard, the most serious problems are: (1) smaller bandwidth and (2) unreliable transmission media, these problems cause the control schemes for wireless networks different from wired networks. Due to the characteristics of transmission over wireless networks and wired networks are different; packet loss in wired networks is mainly caused by network congestion, but transmission error in wireless networks is the main cause of packet loss. In the client/server architecture, streaming data transmitted from wired networks to wireless networks, but in the end-to-end architecture streaming data transmitted from wireless networks to another wireless networks through wired networks. The characteristics of wired networks may not appear caused by sender side wireless networks, thus streaming control schemes in the end-to-end architecture are more difficult. Besides, in streaming data compression technology, now MPEG-4 is the most popular compression standard, the advantages include high compression ratio, variable resolution, adaptive bitrate, error resilience coding, etc. For these reasons, it suits for network transmission.
　　The goal of this thesis is to implement an end-to-end MPEG-4 streaming system over wireless networks. In order to resolve effects of these two wireless networks upon the transmission, we have purposed the control schemes use the information provided from MAC layer of wireless standard to archive the functionality that can determine packet loss type and estimate the available bandwidth. According the result, we can adjust the streaming data and control the transmission. The control schemes include four: (1) resolve the wireless loss in the sender side; (2) adopt video layering, adjust according current network situation; (3) improve the retransmission scheme to avoid other VOP decode error after the lost I-VOP; (4) control the transmission rate, on the one hand we can avoid network congestion caused by sending too fast, and on the other we can dynamic drop P-VOP when the available bandwidth is too slow. In the implementation, we use NDIS to obtain the MAC layer information.

[1] Hai Le Vu, and Hanly S., “A study of TCP performance and buffer occupancy over a fading wireless link”, GLOBECOM '01. IEEE , Volume: 6, pp. 3478-3482, 2001

[2] Ratnam K., and Matta I., “WTCP: an efficient mechanism for improving TCP performance over wireless links”, Proceedings of the 3rd IEEE Symposium on Computers and Communications, pp. 74-78, 1998

[3] S. Cen, C. P.C., and Voelker G.M., “End-to-end differentiation of congestion and wireless losses”, submitted to EEE/ACM Transactions on Networking, pp. 703-717, 2002.

[4] Jain M. and Dovrolis C., “End-to-end available bandwidth: measurement methodology, dynamics, and relation with TCP throughput”, submitted to IEEE/ACM Transactions on Networking, pp. 537-549, 2003.

[5] S. Saroiu, P.K. Gummadi, and S. D. Gribble, “SProbe: A Fast Technique for Measuring Bottleneck Bandwidth in Uncooperative Environments”, INFOCOM 2002, New York, 2002.

[6] Bruce A. Mah., Pchar, http://www.employees.org/ bmah/Software/pchar.

[7] Pathchar, http://www.caida.org/tools/utilities/others/pathchar.

[8] C. Dovrolis., http://www.pathrate.org

[9] Jiri Navratil., “ABwE: A Practical Approach to Available Bandwidth Estimation”, PAM 2003, 2003

[10] Biaz S., and Vaidya N.H., “Discriminating congestion losses from wireless losses using inter-arrival times at the receiver”, Proceedings of 1999 IEEE Symposium on ASSET '99., pp. 10-17, 1999.

[11] L. Rizzo, "Effective Erasure Codes for Reliable Computer Communication Protocols", ACM Computer Communication Review, VOL. 27, NO. 2, pp. 24-36, 1997

[12] Rhee, V. Ozdemir, and Y. Yi. TEAR, "TCP Emulation at Receivers-Flow Control for Multimedia Streaming", Technical Report, Department of Computer Science, NCSU, 2000

[13] S. Cen, C. Pu, and J. Walpole, "Flow and Congestion Control for Internet Media Streaming Applications", Preceedings of Multidedia Computing and Networking, pp. 250-264, 1998

[14] Rejaie R., Handley M., and Estrin, D., “RAP: An end-to-end rate-based congestion control mechanism for realtime streams in the Internet”, Proceedings of IEEE INFOCOM '99. Vol.3, pp. 1337-1345, 1999.

[15] J. Mahdavi and S. Floyd, "TCP-Friendly Unicast Rate-Based Flow Control", Note sent to end2end-interest mailing list, 1997.

[16] D. Sisalem and H. Schulzrinne, "The Loss-Delay Adjustment Algorithm: a TCP-friendly Adaptation Scheme", Proceedings of Workshop on Network and Operrating System Support for Digital Audio and Video, pp. 215-226, 1998.

[17] Information technology-Coding of audio-visual objects-Part1: System, ISO/IEC 14496-1:1999.

[18] Information technology-Coding of audio-visual objects-Part2: Visual, ISO/IEC 14496-2:1999.

[19] Information technology-Coding of audio-visual objects-Part3: Audio, ISO/IEC 14496-3:1999.

[20] Information technology-Coding of audio-visual objects-Part6: Delivery Multimedia Integration Framework, DMIF, ISO/IEC 14496-6:1999.