JVT/H.

26L VIDEO TRANSMISSION IN 3G WIRELESS ENVIRONMENTS

THOMAS STOCKHAMMER, TOBIAS OELBAUM
Institute for Communications Engineering (LNT) Munich University of Technology (TUM) D-80290 Munich, Germany
AbstractVideo transmission in 3G wireless environments is a challenging task calling for high compression efficiency as well as a network friendly design. These are the main goals of the new ITU-T H.26L standardization effort addressing conversational (i.e., video telephony) and nonconversational (i.e., storage, broadcast, or streaming) applications. The video compression performance of H.26Ls Video Coding Layer typically yields a factor of two or more in bit-rate savings when comparing against all previous international video coding standards and therefore provides a significant improvement towards this end. The networkfriendly design goal of the H.26L project is addressed via the Network Adaptation Layer that is being developed to transport the coded video data over existing and future networks such as circuit-switched wired networks, IP networks with RTP packetization, and 3G wireless systems. The error resilience features and appropriate test model extensions at encoder and decoder are introduced in this paper. Additionally, selected results showing the potentials of these features are presented.

THOMAS WIEGAND
Image Processing Department Heinrich Hertz Institute (HHI) D-10587 Berlin, Germany avoiding any excessive quantity of optional features or profile configurations. This paper is organized as follows. We will briefly introduce video transmission in 3G networks and present the test conditions used in the JVT Coding project to evaluate rate-distortion performance. Then we will provide a short overview of JVT Coding with primary focus on the error resilience features. Encoder and decoder test model extensions are discussed that allow the use of the provided features in an optimized way. Selected simulation results are presented and discussed and some concluding remarks are provided. II. VIDEO TRANSMISSION IN 3G NETWORKS A. Overview Video transmission for mobile terminals will be a major application in the upcoming 3G systems and may be a key factor for their success. The display of video on mobile devices paves the road to several new applications. Three major service categories are identified in the JVT standardization process [5]: 1) conversational services for video telephony and video conferencing, 2) live or prerecorded video streaming services, and 3) video in multimedia messaging services (MMS). In general, mobile devices are hand-held and constrained in processing power and storage capacity. Therefore, a mobile video codec design must minimize terminal complexity while remaining consistent with the efficiency and robustness goals of the design. In addition, the mobile environment is characterized by harsh transmission conditions in terms of fading and multi-user interference, which results in time- and location-varying channel conditions. Many highly sophisticated radio link features like broadband access, diversity techniques, fast power control, interleaving, forward error correction by Turbo codes, etc., are used in 3G systems to reduce the channel variance and, therefore, the bit error rate and radio block loss rate. Entirely error-free transmission of radio blocks is a generally unrealistic assumption although with RLC retransmission methods, delay insensitive applications like MMS can be delivered error-free to the mobile user. In contrast, conversational and streaming services with real-time delay and jitter constraints allow for only a very limited number of retransmissions, if any. In addition, in a cellular multi-user environment the transmission capacity within each cell is limited. Therefore, if new users enter

I. INTRODUCTION H.26L [1] is the current project of the ITU-T Video Coding Experts Group (VCEG) a group officially chartered as ITU-T Study Group 16 Question 6. Just recently, a Joint Video Team (JVT) was formed consisting of VCEG and MPEG (ISO/IEC JTC 1/SC 29/WG 11: Moving Picture Experts Group). The charter of the JVT is to finalize the H.26L project of VCEG as technically aligned ITU-T Recommendation and ISO Standard called JVT Coding. The primary goals of the JVT project are improved coding efficiency, improved network adaptation and simple syntax specification. The syntax of JVT Coding should permit an average reduction in bit rate by 50% compared to all previous standards for a similar degree of encoder optimization. Recent results show that this performance is almost achieved [3]. This makes JVT Coding an attractive candidate for wireless video transmission, as the resource bit-rate is extremely costly in mobile environments. However, to allow transmission in mobile environments in addition to coding efficiency, a network adaptation layer and error resilience features are very important. Relating issues examined seriously for the first time in the H.263 and MPEG-4 projects [2], [4] are being taken further in JVT Coding. The scenarios emphasized are primarily for Internet, LAN, and third-generation mobile wireless channels. Finally, the design of JVT Coding is strongly intended to lead to a simple and clean solution

the cell, active users must share resources with new users and if users exit the cell, the resources can be re-allocated to the remaining users. The well-designed 3G air interfaces allow data rate switching in a very flexible way by assigning appropriate scrambling or channel coding rates. This results in a need for video codecs to be capable of (for example) doubling or halving video data rate every 520 seconds. Hence, due to the time-varying nature of the mobile channel, the video application must be capable of reacting to variable bit-rate (VBR) channels as well as to residual packet losses. Finally, the prioritization and quality of service design for mobile links is an ongoing standardization and research activity. Systems supporting prioritized transmission show improved performance if video standards allow generating data with different priorities. B. Common Test Conditions for 3G Mobile Video In the JVT standardization process the importance of mobile video transmission has been recognized by adopting appropriate common test conditions for 3G mobile transmission for circuit switched conversational services based on H.324M [6] and for packet switched conversational and streaming services [7]. These test conditions allow for selecting appropriate coding features, to test and evaluate error resilience features and to produce meaningful anchor streams. In the following we will focus on packetswitched applications and the corresponding common test conditions, as this seems to be more important in nowadays IP-based world. Additionally, the packetization scheme as well as the performance for H.324M based video transmission is comparable to IP-based schemes. The common test conditions define video test sequences with the appropriate temporal and spatial resolution. Additionally, a simplified offline 3GPP/3GPP2 simulation software [8] is available in combination with appropriate parameter settings. For simulating radio channel conditions, bit-error patterns are used that were captured in different real or emulated mobile radio channels. The bit-error patterns are measured above the physical layer and below the RLC/RLP layer, such that in practice they act as the physical layer simulation.
IP UDP PPP RLP frame RTP RoHC RLP Physical frame LTU . frame CRC NAL packet NAL packet RLP frame CRC RTP/UDP/IP Framing, ROHC Link layer Physical layer

very similar. After Robust Header Compression (RoHC), the IP/UDP/RTP packet is encapsulated into one PDCP/PPP packet that will become an RLC-SDU. Video packets are in general of varying length, so RLC-SDUs will be of varying length as well. In the case that an RLCSDU is larger than an RLC-PDU, the SDU is segmented into several PDUs. The flow of variable size RLC-SDUs is continuous to avoid padding bits. RLC-SDUs with one or more RLC-PDUs that contain part of the RLC-SDU have not been received correctly are discarded. The RLC/RLP layer can perform re-transmissions. The retransmission scheme may be set up with different levels of persistency. The common test conditions specify 12 anchors with different video sequences, radio bit error patterns, transmission bit-rates and retransmission modes. III. JVT CODING STANDARD A. Overview Coding Algorithm Although the design of the JVT codec basically follows the design of prior video coding standards as MPEG2, H.263, and MPEG-4, it contains many new features that enable it to achieve a significant improvement in terms of compression efficiency. We will briefly highlight those. For more details we refer to [1] and [3]. In JVT Coding, blocks of 4x4 samples are used for transform coding, and thus a MB (MB) consists of 16 luminance and 4 blocks for each chrominance component. Conventional picture types known as I- and P-pictures are supported. Furthermore, JVT Coding supports multi-frame motioncompensated prediction. That is, more than one prior coded picture can be used as reference for the motion compensation. Encoder and decoder have to store already coded pictures in a multi-frame buffer. A generalized frame-buffering concept has been adopted allowing motion-compensated prediction not just from previous frames but also from future frames. For that, a flexible and efficient signaling method has been adopted. In addition, JVT Coding permits so-called multi-hypothesis (MH) pictures, which similar to B-Pictures allow two prediction signals per block but reference more than one picture. Therefore, the simple B-picture functionality is included with MH-pictures. A MB can always be coded in one of several INTRAmodes. There are two classes of INTRA coding modes, one which basically allows to code flat regions with low frequency components and one which allows to code details in a very efficient way utilizing prediction in the spatial domain by referring neighboring samples of already coded blocks. In addition to the INTRA-modes, various efficient INTER-modes are specified in JVT Coding. In addition to the SKIP-mode that means just copying the content from the same position from the previous picture,

Physical frame LTU

Figure 1 Packetization through 3GPP2 protocol stack According to Figure 1 the software simulator assumes a JVT Network Adaptation Layer Packet (NALP) to be encapsulated in an IP/UDP/RTP packet at the input. NALP usually contain a single slice packet (SSP) or a data partition. For more details we refer to [11]. In the following we briefly examine the user plane protocol stack for 3GPP2 CDMA-2000. The 3GPP UMTS stack is

seven motion-compensated coding modes are available for MBs in P-pictures. Each motion-compensated mode corresponds to a specific partition of the MB into fixed size blocks used for motion description. Currently, blocks with sizes of 16x16, 16x8, 8x16, 8x8, 8x4, 4x8, and 4x4 samples are supported by the syntax, and thus up to 16 motion vectors maybe transmitted for a MB. The JVT Coding syntax supports quarter- and eighthsample accurate motion compensation. The motion vector components are differentially coded using either median or directional prediction from neighboring blocks. The chosen prediction depends on the block shape and the position inside the MB. JVT Coding is basically similar to other prior coding standards in that it utilizes transform coding of the prediction error signal. However, in JVT Coding the transformation is applied to 4x4 blocks and, instead of the DCT, JVT Coding uses a separable integer transform with basically the same properties as a 4x4 DCT. Since the inverse transform is defined by exact integer operations, inverse-transform mismatches are avoided. Appropriate transforms are used to the four DCcoefficients of each chrominance component (2x2 transform) and the INTRA16x16-mode (repeated 4x4). For the quantization of transform coefficients, JVT Coding uses scalar quantization. The quantizers are arranged in a way that there is an increase of approximately 12.5% from one quantization parameter (QP) to the next. The quantized transform coefficients are scanned in a zigzag fashion and converted into coding symbols by runlength coding (RLC). All syntax elements of a MB including the coding symbols obtained after RLC are conveyed by entropy coding methods. JVT Coding supports two method of entropy coding. The first one called Universal Variable Length Coding (UVLC) uses one single infinite-extend codeword set. Instead of designing a different VLC table for each syntax element, only the mapping to the single UVLC table is customized according to the data statistics. The efficiency of entropy coding is improved if Context-Adaptive Binary Arithmetic Coding (CABAC) is used that allows the assignment of non-integer numbers of bits to each symbol of an alphabet. Additionally, the usage of adaptive codes permits the adjustment to non-stationary symbol statistic and context modeling allows for exploiting statistical dependencies between symbols. For removing block-edge artifacts, the JVT Coding design includes a deblocking filter. The JVT Coding block edge filter is applied inside the motion prediction loop. The filtering strength is adaptively controlled by the values of several syntax elements.

B. Error Resilience Features For enhanced error resilience, the test model allows interrupting spatial, temporal and syntactical predictive coding on a MB basis. The principles of each of the adopted features are reasonably well known from prior video coding work, particularly from the H.263+, H.263++, and MPEG-4 projects. However, these features are taken a bit further in the JVT Coding design. Temporal resynchronization within a JVT video bitstream can be accomplished by use of intra picture refresh (stopping all prediction of data from one picture to another), whereas spatial resynchronization is supported by slice structured coding (providing spatially-distinct resynchronization points within the video data for a single picture). In addition, the usage of intra MB refresh and multiple reference frames allows the encoder to introduce well-selected intra updates reference frame selection. Additionally, the packet length can be adapted by appropriate grouping of MBs. Fast rate adaptation can be accomplished by switching the quantization fidelity on a MB basis such that a realtime encoder can react immediately to varying bit rate. For streaming of pre-coded sequences, well-designed buffering can deal reasonably well with varying bit-rate conditions. Still, buffer overflows in VBR environments may not be completely avoidable. For this reason JVT Coding defines new picture types, SP-frame [10] and SIframes, to allow switching between versions of a stream without introducing the efficiency loss associated with an I-frame. Additionally, a syntax-based data partitioning scheme with at most 3 partitions per slice was introduced in [11] allowing less important information to be dropped in the event of a buffer overflow or to be used in conjunction with network prioritization or unequal error protection to support quality of service concepts in networks. IV. TEST MODEL EXTENSIONS A. Overview In the previous section, the coding and error resilience tools of JVT Coding have been presented. In general, the standard defines only the appropriate syntax for each included feature. The selection of appropriate options for each application is up to the implementer. However, in order to judge adopted and proposed normative features, appropriate non-normative mechanisms in the encoder and decoder test model are also included. The JVT Coding project has carefully adopted different non-normative features in the test model software to improve performance in terms of coding efficiency and error resilience. The approach of rate-distortion optimized encoding is used in the encoder test model to select the appropriate coding options. This method is well known to improve the coding efficiency significantly in case of encoders provid-

with Dm being the distortion in the current MB when selecting MB mode m and Rm being the corresponding rate, i.e. the number of bits. The distortion Dm is computed as

with fi being the original pixel value at position i within being the reconstructed pixel value at the MB and position i for coding MB mode m in the simulated channel-decoder pair n. The decoder in the encoder applies simple previous frame EC and therefore serves as an upper bound on the expected MB distortion. According to [14] the parameter should depend on the quantization parameter q as =5exp(0.1q)(q+5)/(34-q). Obviously for large N the encoder has a good estimate of the average decoder distortion. However, with increasing N a linear increase of storage and computational complexity in the encoder is obvious. Therefore, this method might not be practical in real-time encoding processes. Less complex algorithms with similar performance are known [15]. However, the applications of these algorithms are not straightforward due to sub-pel motion estimation, loopfiltering and intra MB prediction in JVT Coding. Additionally, for the purpose of standardization, this simple solution provides flexibility as new features in the decoder are just copies to the encoder. In addition, any other error resilience tools can be tested and an estimation of the expected decoder distortion can be obtained in the encoder easily. C. Packet Length Selection Another critical parameter is the selection of the packet length. the JVT Coding standard allows to group any number of MBs into one slice or NAL packet. No spatial prediction over slice boundaries is allowed. Therefore, a new slice allows resynchronization within one frame. In mobile environments, the probability that shorter packets are hit by a bit error is typically smaller than for larger packets. In addition, shorter packets provide more resynchronization possibilities and, therefore, are favorable in terms of error resilience. But smaller packets also result in efficiency loss due to the restricted spatial prediction in smaller slices and the introduced slice header and network overhead for each packet and due to the interruption of the prediction at the source coder at the packet boundary. Though the header sizes could be reduced by the introduction of parameter sets [11] and RoHC, they are still not negligible. Hence, a careful selection of the packet length adapted to channel and video conditions is vital.

'&

# " $       !    4 ()75 6( 29 C 59 % 5 D 0 A 6 3 D 1 @

E FE Q I RPH G S

B 4 8

 

ing many coding options [12]. In JVT Coding, ratedistortion optimized coding is used for selecting MB modes, reference frames and motion vectors. In addition, if we have knowledge of certain channel characteristics, e.g., a packet loss or bit error rate model, this can also be used in the encoder as will be discussed in more details in the subsection B and C. Although these methods in general require some knowledge on the expected channel conditions, they are robust in a way that only a rough estimate on the channel characteristics is sufficient. Moreover, the decoder includes in addition to a simple previous frame error concealment (EC) and an advanced EC scheme which uses correctly received information to estimate the lost information in a recursive way [13]. For I-pictures a weighted pixel value averaging is applied whereas for P-pictures boundary-matching-based motion vector recovery is utilized. B. Optimized MB Mode and Reference Frame Selection The number of intra coded MBs per picture can be determined at the encoder by various means. For transmission in error-free environments, typically the algorithm for rate-distortion optimization might sometimes select an intra MB due to compression efficiency. In error-prone environments, it may be desirable to increase the number of intra MBs to limit the temporal propagation of any error occurred. The current encoder test model only allows updating an entire row of MBs periodically. However, intra coded MBs in general show bad coding efficiency and their use should therefore be limited. Error-prone transmission shows random losses making the decoding result a random variable as well. Therefore, the encoding decisions are optimized with regards to the expected values of the decoding random variable. Hence, a rate-constrained MB mode and reference frame selection based on the expected mean square error distortion and the rate is utilized. This scheme selects intra MB updates very carefully by trading of distortion versus rate given a probability of channel errors. The expected decoding distortion is estimated by computing the sample average of the decoding random variable via running N channel-decoder pairs in the encoder in parallel to simulate the statistics of the channel and its impact on the decoded video. This provides an estimate of the expected decoder distortion in the encoder. In our implementation, the statistical process of loosing a packet is modeled independently for each of the N decoders. The packet loss process for each decoder is also assumed to be i.i.d., and the slice loss probability p is assumed to be known at the encoder. To be more precise, let us define the set of possible combinations of MB modes and reference frames as SMB including the option to code a MB in INTRA, i.e. without temporal error propagation, or in INTER, i.e., with tempo-

ral error propagation but higher coding efficiency. Then, for each MB the MB mode m is selected according to

The JVT Coding test model encoder allows choosing the slice size in different modes. The number of MBs in each slice can be specified. Therefore, the packets usually differ in length. Especially packets containing intra information or high motion areas result in bigger size, and, therefore, are more susceptible to bit errors. Therefore, in a different mode the maximum number of bytes in one packet can be specified. This assures that packets have similar length and, therefore, are almost equally susceptible to bit errors and resulting losses. The packet length can be adapted so that the loss probability is below a certain threshold if the channel conditions are known at the encoder. Obviously, an individual packet length selection adapted to video content and error characteristics similar to the previously presented adaptive intra update can help to trade off overhead versus compression efficiency in an optimized way. This topic is part of ongoing research. V. SELECTED SIMULATION RESULTS A. Simulation Parameters We will briefly present simulations for selected parameters. An IP based conversational service at a mobile speed of 3 km/h at 64 kbit/s is simulated. We assume interactive services with a small end-to-end delay and therefore, no radio retransmissions are applied. The bearer bit error rate is at about 510-4. The QCIF video test sequences Foreman (10s, 300 frames) and Hall Monitor (10s, 300 frames) are transmitted at a frame rate of 7.5 fps and 15 fps, respectively [7]. Since no rate control is currently present in the JVT video encoder, a fixed quantization parameter is selected so that the total video bit-rate including the packetization overhead does not exceed 64 kbit/s. For each sequence, 50 decoding runs are performed where each run starts at a different predefined starting position in the bit error file. The packet loss probability obviously depends on the packet length. An evaluation of the bit error pattern file shows for example that the loss probability of a packet of length 200 bytes is about 2%, whereas for a packet of length 500 bytes it increases to about 5%. B. Simulation Results Different experiments have been carried out. We only report on a selected subset and show the benefits of different JVT Coding error resilient modes and test model extensions. The entropy coding method applied in all cases is the simpler UVLC. Note that applying CABAC would result in even better results for all experiments. Extensive simulation results as well as selected decoded video sequences are available1. For both video sequences, we report for each investigated case the luminance PSNR averaged over all frames and all runs (av) for advanced
1

EC (AEC). Additionally for Foreman, the average over all frames for the worst-case (wc) run are reported. Furthermore, results for previous frame EC (PFEC) are given as well, since for this sequence, the two concealment methods provide quite different results, while the concealment itself depends on the number of MBs within a slice. The results for different encoder and decoder settings are shown in Table 1. Table 1 Results in PSNR (in dB) for Foreman (FM) and Hall Monitor (HM) for different encoder/decoder settings PFEC FM AEC FM HM av wc av wc av PSNR PSNR PSNR PSNR PSNR 17 13 26.441 15.607 26.441 15.607 32.683 1 19 19 29.380 24.781 29.380 24.781 34.110 2 23 16 30.094 28.637 30.094 28.637 35.398 3 20 18 29.641 22.436 30.519 24.025 33.158 4 22 25 30.436 28.911 30.701 29.463 30.036 5 23 19 30.131 26.168 30.377 28.677 33.887 6 21 16 30.974 29.215 31.169 30.046 35.628 7 22 16 30.719 29.152 30.762 29.890 35.625 8 In Experiments 1, 2 and 3 an entire frame is transmitted in one packet. Therefore, both EC schemes perform identically as the AEC only exploits spatial correlations within one frame. In experiment 1 no error resilience tools have been applied. The results for the average PSNR are acceptable as for the Foreman sequence the R-D optimized mode selection selects the intra MB mode quite frequently. However, the worst-case performance is very poor indicating that without error resilience encoding methods, a very bad decoding quality might occur occasionally. In experiment 2, the introduction of regular intra updates (periodically 1 row of MBs is updated every frame) provides a significant improvement. Even better results can be obtained by adaptive intra updates as applied in experiment 3. Especially the worst-case PSNR is increased significantly and, therefore, the variance of the receiver quality is reduced. A frame loss rate of 10% is assumed causing a large number intra MB updates. Please note that also the quantization parameter is increased due to the lower coding efficiency of INTRA coding. It can be seen that the adaptive coding scheme also adapts the QP appropriately. For the Foreman sequence many more intra MBs are used for the adaptive intra update compared to the regular intra update. The QP matching the required bit rate is much higher for experiment 3 compared to experiment 2. For the Hall Monitor sequence this is different though the channel statistics are equivalent. Therefore, it is obvious that the redundancy necessary to cope with packet loss is adapted not only to the channel statistics but also to the video content. This shows the validity and importance of the rate-distortion approach. Exp QP FM HM

Available at http://www.ei.tum.de/~stockhammer

In experiments 4-8 slices have been introduced to obtain shorter packets. In these experiments, typically the AEC provides gains as spatial correlations within one frame can be exploited at the decoder. Experiment 4 is equivalent to experiment 1 except that each row of MBs is transmitted in a separate slice. This allows frequent resynchronization, but also reduces coding efficiency. The selected QP in this case is 20 for the Foreman sequence, 18 for the Hall Monitor sequence. The performance compared to experiment 1 is slightly increased, especially for the AEC. Introducing regular intra updates as done in experiment 5 does improve the quality significantly for, but not for the Hall Monitor sequence. For the Foreman sequence the amount of intra updates for experiment 4 fits quite well the video content and channel statistics. In experiments 6, 7 and 8 the combination of adaptive intra updates and the slice feature is investigated. In all cases it can be seen that the AEC can improve the decoded quality, especially in the worst scenarios. In experiment 6 the slice loss probability at the encoder is assumed to be 3% and the slice length was the same as in experiment 4 and 5, i.e. one row of MBs in one slice. The gains in this case are significant compared to all other non-adaptive cases, for PFEC as well as for AEC. Again it can be observed that for the Foreman sequence the setting in experiment 5 is well matched and the adaptive intra update is slightly worse for experiment 6. This is, as the channel statistics cannot be modeled accurately in the encoder with an independent packet loss model. However, for the Hall Monitor sequence this experiment shows significant gains compared to experiment 5. Therefore, the robustness of adaptive intra updates is obvious. In experiment 7, the slice length is set to 33 MBs so we have 3 packets per video frame. A loss probability of 5% is assumed at the encoder. The results of this experiment outperform all other experiments as the tradeoff of packet overhead, intra updates and loss rate is well matched. The AEC improves the results slightly as spatial correlations can be exploited if not all packets of one frame are lost. Finally, in experiment 8, the maximum packet length is limited to 256 bytes and the loss probability assumed at the encoder is 5%. The performance is very similar to the results in experiment 7. However, in general the approach limiting the packet length rather fixing the number of MB can provide better result. VI. CONCLUSIONS AND OUTLOOK The JVT Coding project promises some significant advances in the state-of-the-art of standardized video coding, including key aspects designed with mobile applications in mind. In addition to excellent coding efficiency with halving the bit-rate compared to all existing standards, the JVT Coding project also takes into account network adaptation in the inherent design. This includes

the definition of appropriate mobile test conditions, the integration of network related and error-resilient features in the standard and, finally, the extension of the test model software to fully exploit the integrated features. This allows to provide appropriate judgment of adopted and proposed features and to finally come to a standard, which will help to improve the quality of low bit-rate video in 3G mobile environments significantly. Further work will be conducted within the JVT to improve coding efficiency as well as network and application friendliness and to provide a standard suitable for different applications and networks with special focus on mobile video applications. REFERENCES
[1] T. Wiegand, Working Draft Number 1, Revision 0 (WD-1), Joint Video Team (JVT) of ISO/IEC MPEG and ITU-T VCEG, JVTA003, January 2002. [2] ITU-T, Video Coding for Low Bit-Rate Communication, ITU-T Recommendation H.263, Version 1: November 1995, Version 2: January 1998, Version 3: Nov. 2000. [3] G. Sullivan, T. Wiegand, and T. Stockhammer, Using the Draft H.26L Video Coding Standard for Mobile Applications, in Proc. ICIP 2001, Thessaloniki, Greece, October 2001. [4] ISO/IEC JTC1, Generic Coding of Audiovisual Objects Part 2: Visual (MPEG-4 Visual), ISO/IEC 14496-2, Version 1: January 1999, Version 2: January 2000; Version 3: January 2001. [5] S. Wenger, M. Hannuksela, and T. Stockhammer, Identified H.26L Applications, ITU-T SG 16, Doc. VCEG-L34, Eibsee, Germany, Jan. 2001. [6] ITU-T Q15-I-60, Common Conditions for Video Performance Evaluation in H.324/M error-prone systems, VCEG (SG16/Q15), Ninth Meeting, Redbank, NJ, October 1999. [7] G. Roth, R. Sjberg, G. Liebl, T. Stockhammer, V. Varsa, and M. Karczewicz, Common Test Conditions for RTP/IP over 3GPP/3GPP2, ITU-T SG16 Doc. VCEG-M77, Austin, TX, USA, Apr. 2001. [8] G. Roth, R. Sjberg, G. Liebl, T. Stockhammer, V. Varsa, and M. Karczewicz, Common Test Conditions for RTP/IP over 3GPP/3GPP2 Amendments and Software, ITU-T SG16 Doc. VCEG-M37, Santa Barbara, CA, USA, Sept.. 2001. [9] D. Marpe, G. Blttermann, G. Heising and T. Wiegand, Video Compression using Context-Based Arithmetic Coding, in Proc. ICIP 2001, Thessaloniki, Greece, October 2001. [10]R. Kurceren and M. Karczewicz, A Proposal for SP-frames, ITUT SG 16 Doc. VCEG-L27, Eibsee, Germany, Jan. 2001. [11]S. Wenger and T.Stockhammer, H.26L over IP and H.324 Framework, ITU-T VCEG-N52, VCEG (SG16/Q6), Fourteenth Meeting, Santa Barbara, CA, September 2001. [12]G.J. Sullivan and T. Wiegand, Rate-Distortion Optimization for Video Compression, IEEE Signal Processing Magazine, vol. 15, no. 6, pp. 74-90, Nov. 1998. [13]V. Varsa, M. Hannuksela, and Y. Wang, Non-normative error concealment algorithms, ITU-T VCEG-N62, VCEG (SG16/Q6), Fourteenth Meeting, Santa Barbara, CA, September 2001. [14]T. Wiegand, and B. Girod, Lagrangian Multiplier Selection in Hybrid Video Coder Control, in Proc. ICIP 2001, Thessaloniki, Greece, October 2001. [15]R. Zhang, S. L. Regunathan, and K. Rose, Video Coding with Optimal Inter/Intra-Mode Switching for Packet Loss Resilience, in IEEE JSAC, vol. 18, no. 6, pp. 966-976.