With the popularity of and the advances in wireless networking technologies, wireless multimedia tra c has grown dramatically in recent years. Despite having many advantages, wireless multimedia services, particularly video services, still pose a number of challenges due to the time-varying, error-prone and bandwidth-fluctuating channels in the wireless networks. Therefore, provisioning end-to-end Quality of Service and Quality of Experience (QoS and QoE) of video transmission over wireless channels is of great importance. ☐ Video transmission is often described to be bursty as video is basically a sequence of frames transmitted at a particular frame rate. A video frame cannot be decoded or played out at the receiver side until all or most of its transmitted constituent packets are received in time. Depending on the application scenarios, video services may have different emphases in terms of QoE and QoS. While video streaming (e.g., Netflix and YouTube) allows for modest delay (on the order of a few seconds) at the beginning of the playout, video teleconferencing (e.g., FaceTime and WebRTC) is much more delay constrained (less than a few hundred milliseconds). This is because in real-time video systems, each frame must be delivered and decoded by its playback time, and any packet that is retransmitted due to loss in the last transmission or arriving late becomes useless when its stringent decoding and display deadline cannot be met. In this dissertation, we propose several optimization algorithms to improve the QoE and QoS for both video streaming (non real-time) and video teleconferencing (real-time) over wireless networks. ☐ In optimizing wireless video streaming, we focus on MPEG-DASH (ISO/IEC Standard 23009-1), the current standard for video streaming. We optimize video streaming by leveraging a technique called User Adaptive Video (UAV), which exploits the perceptual limits of the human visual system to modulate a video stream’s bit rate based on the viewing conditions, such as viewing distance and ambient illuminance, resulting in significant bandwidth saving without perceived loss of quality to the user. UAV presents an opportunity to significantly improve the e ciency of DASH by not requesting unnecessarily high bit rate videos. We design UAV-enabled DASH (UDASH) and evaluate its performance in Wi-Fi networks. Simulation results show that UDASH in a Wi-Fi network has the benefits of not only significantly improving the video streaming performance such as reducing the rebu ering probability, but also enhancing the performance of cross traffc. ☐ In addition, the MPEG-DASH standard uses TCP as the underlying transport layer protocol, and more importantly, TCP is one type of dominant tra c in the Internet. Therefore, we investigate how to improve TCP performance in wireless networks. We identify two issues of TCP performance degradation due to common channel errors via both analytical study and simulations in a typical Wi-Fi network. Motivated by these issues, a MAC layer optimization technique is proposed, which is based on the adaptation of the Retry Limit parameter after considering TCP traffic characteristics and throughput model. The evaluation results confirm that the proposed technique achieves higher performance gain. ☐ In optimizing video teleconferencing, we considerWebRTC, which is Google’s open source real-time communication framework. In wireless networks such as those based on IEEE 802.11, packet losses due to fading and interference are often misinterpreted as indications of congestion, causing unnecessary decrease in the data sending rate due to congestion control by the RTCP protocol working beneath WebRTC and above RTP. For delayconstrained applications such as video teleconferencing, packet losses may result in excessive artifacts or freeze in the decoded video. We propose a simple and yet effective mechanism to detect and reduce channel-caused packet losses by dynamically adjusting the retry limit parameter of the IEEE 802.11 protocol. Since the retry limit is left configurable in the IEEE 802.11 standard, and does not require cross-layer coordination, the proposed scheme can be easily implemented and incrementally deployed. We also propose to use a delay constrained retry limit adaptation algorithm to control transmission delays so that delay constraints required by different application scenarios can be met. Experimental results of applying the proposed scheme to aWebRTC based real-time video communication prototype show significant performance gain compared to the case where retry limit is configured statically. ☐ In addition to the optimization techniques proposed for the IEEE 802.11 protocol, we also propose a cross-layer approach to optimize video teleconferencing, termed early packet loss feedback (EPLF). In EPLF, if a packet loss is due to channel errors, the MAC layer directly feeds back the loss information to the RTP layer with a spoofed RTCP packet that carries a NACK message so that the RTP layer can retransmit the lost RTP packet. Since the whole feedback process takes place in the same device (the video sender), the latency is negligible in relation to the RTT, and hence the term ’early’ in EPLF. Theoretical analysis and prototype-based experimental results show that EPLF almost completely eliminates channelcaused video freezes in the decoded video while improving congestion control. ☐ Furthermore, we also apply the technique of UAV to video teleconferencing to further reduce bandwidth consumption, and build a prototype based on WebRTC and Licode (a video teleconferencing hub platform) to validate the bandwidth savings.