The problem of error control and concealment in video communication is becoming increasingly important because of the growing interest in video delivery over unreliable channels such as wireless networks and the Internet. This paper reviews the techniques that have been developed for error control and concealment in the past ten to fifteen years. These techniques are described in three categories according to the roles that the encoder and decoder play in the underlying approaches. Forward error concealment includes methods that add redundancy at the source end to enhance error resilience of the coded bit streams. Error concealment by postprocessing refers to operations at the decoder to recover the damaged areas based on characteristics of image and video signals. Finally, interactive error concealment covers techniques that are dependent on a dialog between the source and destination. Both current research activities and practice in international standards are covered.

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Internet telephony offers the opportunity to design a global multimedia communications system that may eventually replace the existing telephony infrastructure. We describe the upper-layer protocol components that are specific to Internet telephony services: the Real-Time Transport Protocol (RTP) to carry voice and video data, and the Session Initiation Protocol (SIP) for signaling. We also mention some complementary protocols, including the Real Time Streaming Protocol (RTSP) for control of streaming media, and the Wide Area Service Discovery Protocol (WASRV) for location of telephony gateways. 1

With the success of the Internet and flexibility of MPEG-4, transporting MPEG-4 video over the Internet is expected to be an important component of many multimedia applications in the near future. Video applications typically have delay and loss requirements, which cannot be adequately supported by the current Internet. Thus, it is a challenging problem to design an efficient MPEG-4 video delivery system that can maximize the perceptual quality while achieving high resource utilization. This paper addresses this problem by presenting an end-to-end architecture for transporting MPEG-4 video over the Internet. We present a framework for transporting MPEG-4 video, which includes source rate adaptation, packetization, feedback control, and error control. The main contributions of this paper are: 1) a feedback control algorithm based on Real Time Protocol (RTP) and Real Time Control Protocol (RTCP); 2) an adaptive source-encoding algorithm for MPEG-4 video which is able to adjust the output rate of MPEG-4 video to the desired rate; and 3) an efficient and robust packetization algorithm for MPEG video bit-streams at the sync layer for Internet transport. Simulation results show that our end-to-end transport architecture achieves good perceptual picture quality for MPEG-4 video under low bit-rate and varying network conditions and efficiently utilizes network resources.

Rate-distortion (R-D) optimized mode selection is a fundamental problem for video communication over packet-switched networks. The classical R-D optimized mode selection only considers quantization distortion at the source. Such approach is unable to achieve global optimality under the error-prone environment since it does not consider the packetization behavior at the source, the transport path characteristics and receiver behavior. This paper presents an end-to-end approach to generalize the classical theory of R-D optimized mode selection for point-to-pointvideo communication. Weintroduce a notion of global distortion by taking into consideration both the path characteristics (i.e., packet loss) and the receiver behavior (i.e., the error concealmentscheme), in addition to the source behavior (i.e., quantization distortion and packetization). We derive, for the first time, a set of accurate global distortion metrics for any packetization scheme. Equipped with the global distortion metrics, we design an R-D optimized mode selection algorithm to provide the best trade-off between compression efficiency and error resilience. The theory developed in this paper is general and is applicable to many video coding standards, including H.26---ding and MPEG-1/2/4. As an application, weintegrate our theory with point-to-point MPEG-4 video conferencing over the Internet, where a feedback mechanism is employed to convey the path characteristics (estimated at the receiver) and receiver behavior (error concealmentscheme) to the source. Simulation results conclusively demonstrate that our end-to-end approach offers superior performance over the classical approachforInternet video conferencing.

this paper we evaluate the transmission of GSM encoded voice with ROHC over a wireless link. We first present a tutorial on voice quality evaluation. We introduce an evaluation methodology that combines elementary objective voice quality metrics with a novel frame synchronization mechanism. The methodology allows networking researchers to conduct e#ective and accurate quality evaluation of packet voice. Besides the impact of ROHC on the voice quality we consider the impact of ROHC on the consumed bandwidth and the delay jitter in the voice signal. We find that for a wide range of error probabilities on the wireless link, ROHC roughly cuts the bandwidth required for the transmission of GSM encoded voice in half. In addition, ROHC improves the voice quality compared to transmissions without ROHC, especially for large bit error probabilities on the wireless link. The improvement reaches 0.26 on the 5-point Mean Opinion Score for a bit error probability of 10 -3

IP Header compression mechanisms have always been an important topic for the research community to save bandwidth in the Internet. Due to the high license fees of 3G bands and the upcoming integration of IP based multimedia services, it is of particular importance to reduce the IP header overhead even in the wireless format. Reducing the IP overhead gives the network providers the possibility for a faster return of investment on their 3G networks and simultaneously enables real--time services by improving the latency of the IP packets over bandwidth limited links. Many compression methods exist already but they are either not designed for multimedia services, or not robust in the presence of error--prone links and therefore not suitable for wireless communication. For wireless environments robust header compression was introduced. Robust header compression was standardized by the Internet Engineering Task Force in RFC 3095 and will be an integral part of the 3GPP--UMTS specification. This compression scheme was designed to operate in error--prone environments by providing error detection and correction mechanisms in combination with robustness for IP based data streams. A connection oriented approach removing packet inter-- and intra--dependencies reduces the IP header significantly. This paper gives a solid performance evaluation for robust header compression showing both the bandwidth savings for the IP protocol stack and the quality of services at the application layer by the means video services.

This paper is a detailed survey of the four related protocols. Other Reports on Recent Advances in Networking Back to Raj Jain's Home Page Table of the Contents Multimedia Networking: Goals and Challenges The real-time challenge m Multimedia over Internet m The solution m l RSVP Development m How does RSVP work m RSVP features m RSVP interfaces m l RTP Development m How does RTP work m RTP fixed header fields m RTCP m RTP features m RTP Implementation Resources m l RTSP Development m RTSP operations and methods m l Multimedia Over IP: RSVP, RTP, RTCP, RTSP http://www.cis.ohio-state.edu/~jain/cis788-97/ip_multimedia/index.htm (1 of 24) [2/7/2000 12:37:09 PM] RTSP features m RTSP i

Syntax Notation 1 (ASN.1) [122] offer no functional advantage. Many parameters are textual, so that there is no significant penalty in terms of bytes transmitted. Unlike the ASN.1 Packed Encoding Rules (PER) [123] and Basic Encoding Rules (BER) [124], a SIP header is largely self-describing. Even if an extension has not been formally documented, as was the case for many common email headers, it is usually easy to reverse-engineer them. Since most values are textual, the space penalty is limited to the parameter names, usually at most a few tens of bytes per request. (Indeed, the ASN.1 PER-encoded H.323 signaling messages are larger than equivalent SIP messages.) Besides, extreme space efficiency is not a concern for signaling protocols. If not designed carefully, text-based protocols can be difficult to parse due to their irregular structure. SIP tries to avoid this by maintaining a common structure of all header fields, allowing a generic parser to be written. Unlike, say, HTTP and...

Applications that use real time video are becoming increasingly necessary for effective communication over the Internet. Their popularity is increasing in areas such as distance learning, distributed research and video conferencing. Since real time video has special Quality of Service (QoS) requirements for it to be acceptable to end users, administrators are faced with challenges that involve the end-to-end ability of participating systems to support real time video communication. Unfortunately, the tools that exist in the market today are either expensive and closed source, or are generic and do not explicitly consider the characteristics of real time video. This work proposes a framework for assessment of end-to-end Quality of Service capabil-ities for support of real-time Variable Bit Rate (VBR) compressed video communication. This framework evaluates the endpoints and their inter-connectivity and determines the ex-tent of their ability to support compressed video over User Datagram Protocol (UDP). The framework generates VBR traffic that mimics compressed low bit rate video, maintains and reports detailed and summary statistics of each test. User configurable options include specifying activity levels and bandwidth upper bounds. The framework itself is modular