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Video telephone systems are convenient media for people to communicate remotely in life. If the video can be analyzed in the process of video chat to extract the relevant physiological indicators of the communicators such as emotions or stress, it can help the communicators to make proper conversations. In this paper, we propose (1) Similar Brightness Area Selection Method (2) and Difference-driven Weight-Setting Method to measure the heart rate in the non-contact image mode that can be adapted to the video signal in the videophone. After the image is compressed, it is transmitted through the network. After being transmitted to the video receiver for image decompression, we also propose that the luminance (Y) instead of Green channel (G) of the image should be utilized based on Photoplethysmography (PPG) to infer the heart rate. In our experiment, we have adjusted the weight range of our Difference-driven Weight-Setting Method for four commonly used video compression codecs including H.264, H.265, VP8, and VP9 together with five different compression bit rates and the results have shown to be effective in improving the accuracy of the heart rate measurement. The root mean square error (RMSE) of the heart rate measurement has also been calculated as evaluation metrics in the experiment. The mean and standard deviation of the overall RMSE is 2.14±0.52 bpm (beats per minute) that have shown the feasibility of our proposed methods to effectively and accurately measure the heart rate in the video telephony system.

In conventional telesonography, the ultrasound video is encoded at a constant bit rate and transmitted to the expert side for diagnosis, this is highly bandwidth demanding and less adaptable to the varying network conditions. In contrary, Internet based videophone applications adapt to the network conditions with variable bit rate encoding and error concealment algorithms. Since portable ultrasound scanners are coming with Internet connectivity, this motivated us to evaluate the viability of using existing Internet based videophone services for real-time telesonography. Popular videophone services like Skype, Facebook and WebRTC is chosen for this purpose. The performance of these services is analyzed by varying network parameters like bandwidth, packet loss, delay and packet error in a controlled set up. Liver ultrasound video is considered for the study. A total of 273 different network settings is analyzed for evaluating the performance of each videophone service, constituting a total of 819 instances for evaluating three services. Three medical experts were participated in the subjective study. From an extensive analysis, an inference is made that the Internet based videophone services are reliable enough for telesonography provided a network with minimum bandwidth of 1600 kHz, delay up to 200 ms, packet loss up to 3% and packet error up to 3%.

The purpose of this study is to construct a framework for enhancing telecom operators' network services. To achieve this, we propose “SightFinder,” an enhanced videophone service in which the media is processed on the network by multiple pattern recognition engines working in parallel. This service alerts users when they cannot see information and explains the information when they cannot understand it.

The goal of this project is to design a user interface for the videophone system to enable real-time video chat. The system includes voice and 3D talking heads and able to communicate using very low bandwidth because no video data are transmitted through network. Instead of video, facial expressions of speakers are transmitted along with voice in the real time. At the other end, talking head will show those facial expressions and will render the voice. The facial structure of the talking head will resemble the speaker's face.

A system architecture that enhances existing videophone service with Augmented Reality (AR) technology is designed. The architecture is used to prototype an “AiR Stamp” system that allows users to place virtual objects on video images as if they were affixing stamps to actual scenes. System feasibility is evaluated by applying it to a remote support service scenario where the stamps are used for telepointing. The video latency and accuracy of stamp positions are confirmed to be at a usable level.

This paper presents a prototype of full-duplex hands-free videophone add-on device for digital television sets. It consists of a digital camera and microphone array which are connected to the TV set through the High-Definition Multimedia Interface. The system is extendable with a 3G modem to become capable of working in a stand-alone mode as a video phone for 3G network. The add-on provides: hands-free video communication, excellent sound quality via TV loudspeakers and simultaneous watch of the TV broadcast. While the communication with remote user is active, the existing sound from the TV broadcast is efficiently suppressed with the embedded audio processing engine. This is possible with microphone array usage and sophisticated voice processing algorithms in the background. This add-on aims to extend functionality of TV sets and improve user experience.

This paper presents an integrated hands-free audio and video communication system for digital television sets. The system is an embedded real-time add-on device of the standard digital TV set. It consists of a digital camera and microphone array which are connected to the TV set through the High-Definition Multimedia Interface. The system is extendable with a 3G modem to become capable to work in a stand-alone mode as a video phone for 3G network. The hardware architecture of the system consists of: (1) the central processing module and (2) remote controller with voice call interface and support for typing text messages. The main software components are: (1) video decoding, (2) engine, (3) On-Screen Display support for videophone, (4) remote control handling, (5) display control and (6) sound control. The system provides hands-free video communication, excellent sound quality via TV loudspeakers and simultaneous watch of the TV broadcast. While the communication with remote user is active, the existing sound from the TV broadcast program is efficiently suppressed with the embedded audio processing engine. This is possible with microphone array usage and sophisticated voice processing algorithms in the background. The system aims to extend functionality of TV sets and improve user experience.

This paper describes the design and realization of a kind of WLAN video telephone based on WAPI. The hardware system consisted of two parts: application processing subsystem and base band processing subsystem. Take SIP as the call control protocol, H.264 as the video codec and G711 as the audio codec. Using WAPI as the security mechanism of wireless data transmission, and solved the security problem of password and guarantee the quality of the video and audio service.

The paper introduces a solution of IP videophone based on the platform of TMS320DM642 of TI Co. The solution is that coding/decoding algorithm of H.264 video and G.711 audio and their parallel real-time processing are realized in the same chip in order to achieve real-time communication for videophone.

Within industrialized countries healthcare systems currently change to cope with the upcoming consequences of the demographic change. One of the most serious challenges is the maintenance of the area-wide supply chain of medical care despite the threatening shortage of physicians. In this context, telemedical services for communication between doctors and patients gain in importance. Crucial for the success of such electronic services is the choice of the medium, which must be appropriate for this special purpose of use and, finally, accepted by its users. In this paper, an exploratory survey was conducted to detect acceptance motives of five different media (face-to-face, telephone, videophone, video conference, interactive wall) in two different usage situations. 103 respondents participated (17–83 years). Findings show that for the standard case, a face-to-face consultation is still highly preferred compared to any telemedical applications. For emergency situations, participants' attitudes change: A telephone consultation was similarly well accepted than face-to-face communication. As the most comfortable service a face-to-face-consultation was corroborated, followed by the videophone consultation.

Recently, the quantity of data has known a big evolution especially with the emergence of many video applications over networks such as the videophone and the videoconferencing, and multimedia devices such as the high-definition TV and the personal digital assistants. So, it was crucial to reduce the quantity of data stored or transmitted by compressing it spatially and temporally. Hence, motion estimation and compensation are employed in video coding systems to remove temporal redundancy while keeping a high visual quality. They are the most important parts of the video coding process since they require the most computational power and the biggest consumption in resources and bandwidth. Therefore, many techniques have been developed to estimate motion between successive frames. In this paper, we will present our motion estimation and compensation method applied on the discrete wavelet transform coefficients and based on the block matching algorithm which is the simplest, the most efficient and the most popular technique. Additional techniques are introduced to accelerate the estimation process and improve the prediction quality.

Advanced computer technology, internet communication technology and synthesis wiring technique were used and personality requirement and living subsystem such as security defense, light control, curtain control, gas valve control, information household electric appliance and scene linkage were combined in intelligence home system. New living experience was acquired by internet intelligent control and management. System control can be come true by local control and telecontrol. Local control is to control lamp and household electric application by internet switching. Telecontrol is to control them by telecontroller, timing controller, integrated controller or telephone and computer. Intelligent home control system can realize some functions: telecontrol function, sound telephone telecontrol function, timing control function, central control function, sense function, internet telecontrol function, global internet video monitoring function and safety defense warning function. Now the research focuses of intelligence home control system are videophone, entrance guard system and safety defense warning subsystem. These subsystems are mutual independence and can't realize information share, which made setting and use inconvenience. In the paper, the new system is be studied and these control subsystem and electric apparatus will be connected by uniform central control unit to realize information share and home intelligent telecontrol.

With the development of the multimedia and network technology, video applications such as VOD, video-phone etc. are used more and more widely in our life. Consider by the high performance and easy implementation, we select TMS320 c6416 DSP platform and implement the MPEG-4 decoder. Based on the architecture and instruction characteristic of DSP, structure-level, program-level, code-level optimization was conducted respectively. Experimental data show that the decoder implemented can decode video in real time efficiently.

The architecture design of VOIP videophone system composed of TMS320DM642 and ARM9 is described in this paper firstly. Then the scheduling design of VOIP videophone under DSP / BIOS real-time operating system is introduced. The system control and data operation management achieved by HPI Interface design between DSP and ARM9 is elaborated further in detail. Practical results show that it has solved the complex scheduling and control of the system, improving the efficiency of the system implementation.

This paper presents a videophone service system based on the robot for young children. The proposed system for videophone service consists of a server, which mediates and manages client devices, and a robot, which acts as a client device. Robot in this paper is implemented to the embedded system, which has ARM11 hardware and embedded Linux for the OS. To make an easy phone call for young children, the robot video phone (RVP) module composed of the robot videophone main module and two kinds of libraries is proposed. To avoid the complicated usage of robot based videophone service for young children, the RFID technology is adopted. As young children touch a RFID card to the robot, the robot can make a call to the specific number, which is contained in the RFID card.

Combining video encoder and content analyzer to improve the encoding efficiency by content-aware algorithms is very challenging now. For the aiming application of low cost hardware real-time encoder with face detection for videophone, this paper proposes a face detection algorithm to detect each macroblock (MB) as one part of a face or not. This face detection algorithm has a unique estimation-and-verification process and can be combined with a H.264 encoder by MB level pipeline architecture. 97.91% MBs in faces can be detected. VLSI architecture of proposed face detection algorithm is designed and an area of 4.3k gates is achieved. Power consumption is only 1.45mW at 100MHz. The detection speed achieves 1315fps in CIF sequences.1

Web-IMS integration offers an opportunity to provide novel services that combine rich web contents and IMS real-time communication functions. A rich Internet application (RIA) is a promising way to provide Web-IMS cooperative services for web terminals without modification to the terminals. We propose an architecture that separates the IMS client function (SIP/RTP stack) from RIA as a shared SIP/RTP component. Web-IMS cooperative RIAs call the service-level APIs of the SIP/RTP component to use IMS communication functions so that web developers can implement Web-IMS cooperative RIAs without having expert IMS knowledge. We describe two system architectures; one dynamically deploys the component as an RIA and the other deploys the component at an external gateway. Prototype evaluation shows the feasibility of the architectures; the overhead of dynamic deployment is about 0.48 seconds, and the videophone latency caused by media translation at the external gateway is less than 132.8 milliseconds.

Region-of-interest (ROI) can be applied in H.264 video encoder to enhance subjective quality and reduce computation complexity. For the aiming application of low cost hardware real-time encoder in videophone with faces as ROI, this paper proposes a face detection algorithm to detect each macroblock (MB) as one part of a face or not. This face detection algorithm has a unique estimation-and-verification process and can be combined with a H.264 encoder by MB level pipeline architecture. 97.91% MBs in faces can be detected. VLSI architecture of proposed face detection algorithm is designed and an area of 4.3k gates is achieved. Power consumption is only 1.45mW at 100MHz. A ROI based H.264 encoder with dynamic parameters is proposed to enhance subjective quality and reduce the rate-distortion-optimization (RDO) complexity. The PSNR in ROI increases for 4.8dB under similar bit rate. Encoding time is reduced to 54.4% in videophone-like sequences.

This chapter contains sections titled: Video Codec Outline The Principle of Motion Compensation Transform Coding The Codec Outline Initial Intra-Frame Coding Gain-Controlled Motion Compensation The MCER Active/Passive Concept Partial Forced Update of the Reconstructed Frame Buffers The Gain/Cost-Controlled Inter-Frame Codec The Bit-Allocation Strategy Results DCT Codec Performance under Erroneous Conditions DCT-Based Low-Rate Video Transceivers System Performance Summary and Conclusions

This chapter contains sections titled: Turbo-Equalized H.263-Based Videophony for GSM/GPRS Wideband Burst-by-Burst Adaptive QAM-Based Wireless Videophony A UMTS-Like Videophone System: Turbo-Coded Burst-by-Burst Adaptive Joint Detection CDMA and H.263-Based Videophony H.263/OFDM-Based Video Systems for Frequency-Selective Wireless Networks Subband-Adaptive Turbo-Coded OFDM-Based Interactive Videotelephony Digital Terrestrial Video Broadcasting for Mobile Receivers Satellite-Based Video Broadcasting Summary and Conclusions Wireless Video System Design Principles

In this paper, we propose an adaptive system for improving videophone transmission over EDCA. We consider that a videophone contains a constant bit rate (CBR) voice source and a rate-adaptive video source. Two issues are addressed in this research. Firstly, how to solve the AP bottleneck problem, and secondly, how to adjust video source rate to improve the network performance. For the first issue, we propose the adjustment of the transmission opportunity (TXOP) to give AP a higher priority in voice transmission in order to eliminate the AC3 transmission bottleneck at the AP. For the second issue, our principle is to guarantee the throughput of voice traffic while transmitting as much video traffic as possible. Moreover, we consider more realistic multi-rate WLANs, where multiple transmission rates are used in the PHY layer depending on the underlaying channel conditions.

In this paper, we propose an improved coeff_token variable-length decoding (VLD) method in the CAVLC decoder. The number of memory accesses is one of the significant issues in the applications such as DMB and videophone services, where the low-power consumption and high-speed operation are required. In order to cut down the heavy memory accesses, a new fast VLD method is developed through the careful examination of the codewords in variable-length code tables (VLCTs). The simulation results show that the proposed method achieves an approximately 60% memory access saving without video-quality degradation, compared to the existing fast coeff-token VLD method.

We present a multimedia quality integration function that can estimate the overall quality of a videophone service considering multimodality and interactivity. That is, it takes into account individual audiovisual qualities and their delays. Videophone services are rich multimedia services that are expected to become popular on the next-generation network (NGN). To provide ones with sufficient quality, we must properly evaluate, design, and manage users' perceptual quality of service, i.e., quality of experience (QoE). Subjective quality assessment tests were conducted using a PC-based point-to-point interactive videophone application to clarify how overall multimedia quality characteristics depend on the individual audiovisual qualities, absolute audiovisual delay, and media synchronization. From the test results, we formulated a multimedia quality integration function that can accurately evaluate overall multimedia quality. In addition, verification test results showed that this function achieved good performance in terms of correlation coefficient and evaluation error between subjective and estimated qualities. Our model will be a useful QoE planning tool capable of providing feedback on individual qualities as well as the overall multimedia quality of videophone services.

This paper presents a comparative analysis of alternative approaches that have been developed by IEEE, 3GPP and others for handover of PSTN and Internet sessions among multiple wireless technologies including UMTS, WiFi, WiMAX, Mobile-Fi and WPANs. It evaluates these approaches from the viewpoint of provision of QoS, Quality of Service, for seamless handover of voice, videophone, gaming, audio/video streaming and file download, and shows how QoS at the IP layer can be interworked with QoS at the wireless link layers to provide the QoS required by the user application. The method is illustrated by examples of video streaming and voice over IP.

In this paper, a content-based error resilient RDO mode decision scheme is proposed for Internet videophone applications. The experimental results based on typical head-and-shoulder video sequences under common Internet channel condition show that, comparing with conventional error resilient RDO mode decision schemes, the proposed mode decision scheme improves the average luminance PSNR of visually important foreground area consistently regardless of the packet loss rate.

In this paper, we provide an evaluation for our proposed real-time virtual videophone using talking heads over low-bandwidth wireless networks, in which a person's face is substituted with a talking head to save bandwidth. Results show large enough gains in communication bandwidth to make the architecture feasible.

In this paper we present a novel approach to video telephony where we are able to exploit the benefits of the Universal Plug and Play (UPnP) protocol: self-configuration, discovery and control of available devices connected to the network. At the time being, UPnP does not deal with Video Voice over IP (VVoIP) applications. Integrating a signaling protocol like the Session Initiation Protocol (SIP) with UPnP allows interaction between the videophone and other UPnP compliant devices, like a TV set or a media server. This interaction, for example, makes it feasible to display the caller's image on the television screen, thus allowing the user to relax on the sofa and naturally converse. Another scenario is the one where the user sends to the conversation partner a set of still pictures or short video clips acquired from a UPnP media server already available in the home premises. An advantage of this distributed approach is its cost effectiveness, since it allows to reuse existing UPnP devices also in the context of video telephony. The integration between SIP and UPnP has been achieved introducing a new UPnP device, the VVoIP device, which is a SIP User Agent able to interoperate with standard UPnP MediaServers and MediaRenderers.

We propose a computational opinion model for estimating video quality of videophone services. Opinion models for speech such as the E-model have been established and widely used; however, little attention has been given to opinion models for video quality estimation. Our proposed opinion model is useful as a network-planning tool for assessing several video parameters that affect the quality of videophone services. First, we established a function for estimating video quality affected by coding degradation, which expresses the quality of a video affected by coding bit rate and frame rate. Second, we established a packet loss degradation index that estimates the degree of video quality degradation due to packet loss. Finally, we integrated these two functions into the opinion model for estimating video quality. We applied this model for video quality estimation of a videophone service with various video formats and displayed video sizes. The results indicated that the estimation errors of our model were less than the mean of the 99% confidence intervals for the subjective scores. Therefore, our model could be applied to effective design, implementation, and management of videophone applications and communication networks.

In this age of multimedia convergence, video based applications for conferencing and streaming are seeing big market traction. The main challenge of implementing these systems is a tradeoff between several factors like bandwidth, image quality, implementation cost and speed (in terms of mega cycles per second). Moreover to implement this encoder-decoder system (CODECs) in a real time system based on any digital signal processor (DSP) is more difficult because of their restricted resources like memory and CPU speed. In this paper, we present some novel approach, which reduces the computational complexity and also meets the memory constraint of the target processor for a standard H.264 baseline encoder without sacrificing the rate-distortion performance. The proposed algorithms are applied on standard test sequence of different resolutions. The results obtained from a considerably large number of test sequences show the strength of our proposed algorithm. We claim the betterment in performance measured in mega cycles per second (MCPS) at the cost of negligible loss in image quality. Moreover we gain the betterment in image quality using algorithm modification in rate controlling. Implementation details are also presented for a QCIF H.264 baseline encoder in a TMS320C55x DSP, which has only 256 KB RAM and 150 MHz clock speed. The enhanced performance is achieved thorough formulation of novel adaptive algorithms for rate control and motion estimation. We achieve almost 40% MCPS reduction at the cost of less than 1% reduction in image quality. We also get betterment in image quality by our proposed rate-controlling algorithm

A wavelet-based videophone scheme proposed, where the video bits are unequal error protection (UEP) using irregular convolutional codes (IRCCs). The proposed system uses adaptive arithmetic coding (AAC) for encoding the motion vectors and individual wavelet subband coefficients. The turbo-equalized IRCC-aided videophone scheme is capable of attaining a near unimpaired video quality for channel signal-to-noise ratios (SNRs) in excess of about 4.5 dB over a five-path dispersive AWGN channel

The videophones based on Internet protocol (IP) are gaining increasing acceptance because of the enhanced compression capabilities using H.264/MPEG-4 AVC. H.264 offers high compression, but it also demands high degree of complexity. Hence, optimization to the maximum extent is required for designing the H.264 video encoder with real time performance on DSPs. In this paper a highly optimized H.264 baseline profile video encoder on a programmable DSP processor is presented.

In this paper, we present an efficient face transmission system for MPEG-4 mobile videophone via the wireless Internet. The system consists of two stages: region segmentation and facial region detection. In the region segmentation, the input image is segmented into an appropriate set of arbitrary regions using a watershed algorithm. In facial region detection, the facial regions are identified from the segmented results using a skin-color model. The results of region segmentation and facial region detection are integrated to provide facial regions with accurate and closed boundaries. This system can be used to transmit a face to a corresponding person via the wireless Internet.

In The case of a system for delivering “live streaming” video and audio from the internet under a videophone mechanism in real-time, the major issue is to assure video quality and usability at the same time. To solve this issue, NTT DoCoMo has developed technology that utilizes the “video-streaming gateway” provided by “M-stage V Live” service.

We propose a Vector-Quantized (VQ) video codec that induces deliberately-introduced code constraints. The complete set of these code constraints are unambiguously described by a novel block-based modification of a symbol-based variable length coding trellis structure. This trellis is employed during VQ encoding to represent video information with a Minimum Mean Squared Error (MMSE) reconstruction. Additionally, the proposed trellis structure is employed in the video decoder to obtain a posteriori probability based MMSE estimates of the transmitted VQ tiles. Since the trellis structure unambiguously describes the code constraints induced by the video codec, its employment guarantees the automatic recovery of valid sets of VQ tiles. The proposed video codec is serially concatenated with a trellis-coded modulation scheme for transmission over a fast-fading Rayleigh fading channel. In the receiver, iterative-decoding convergence that eliminates all perceivable channel-induced video distortion is shown to be supported within 1.29 dB of the system's 2 bit/s/Hz channel-capacity limit of 3.96 dB.

This paper presents an architecture for a multimedia application platform able to support real-time group communication activities (such as: Videophone, Teletraining, Computer Supported Collaborative Work) between remotely located members of groups by delivering multimedia communication capabilities directly at their personal computers ("desktops"). The paper focuses on a component-based design approach that gives the potential to integrate into the platform various Application Components (Audio, Video, Text chat, File transfer, Slide show, Shared Whiteboard or text editor, etc.) depending on the activity supported and thus permitting to customise the platform for the specilic activity. Apart from the basic architecture, alternative connection schemes depending on the capabilities of the network are surveyed. A discussion of the advantages and disadvantages of the described architecture is also given andfinally, an-implernentation aiming to support Teletraining activities is presented.

An economical multipoint audiovisual communicating system technology for the ISDN basic rate interface is introduced. The system offers multipoint audiovisual communication using the ring-type connection via the two B channels in the ISDN basic rate interface for each terminal. The system's connection control offers a terminal skipping procedure. Multipoint image processing produces a divided display by digital data manipulation. Finally, an ISDN basic rate interface videophone using this system technology is described.

Mobile phones have become essential part of modern life and continue to change the way people communicate with each other. The camera phones have become ubiquitous in the recent past and video communication services such as videophone are getting rolled out by cellular providers. This would require video compression algorithms that are resilient to transmission errors and support resource-adaptive playback. We propose a video codec that addresses the two key issues in video delivery to mobile devices: (1) error resilience; and (2) resource adaptive playback. We show that the proposed codec is simple to implement on low-resource terminals such as mobile phones and outperforms MPEG-4 coding in terms of loss tolerance.

A perceptual weighting model is proposed for effective rate control so as to enhance the perceptual coding quality of videophone. We exploit two categories of factors affecting the perception of the human visual system, stimulus-driven factors and cognition-driven factors. In order to achieve a simple, but effective, perceptual weighting model, we use luminance adaptation and texture masking as the stimulus-driven factors, while skin color serves as the cognition-driven factor in the videophone application. Both objective and subjective quality evaluations of videophone-like sequences in the H.263 platform validate the effectiveness of our perceptual weighting model.

We present a method for extracting local visual perceptual clues and its application for rate control of videophone, in order to ensure the scarce bits to be assigned for maximum perceptual coding quality. The optimum quantization step is determined with the rate-distortion model considering the local perceptual clues in the visual signal. For extraction of the perceptual clues, luminance adaptation and texture masking are used as the stimulus-driven factors, while skin color serves as the cognition-driven factor in current implementation. Both objective and subjective quality evaluations are given by evaluating the proposed perceptual rate control (PRC) scheme in the H.263 platform, and the evaluations show that the proposed PRC scheme achieves significant quality improvement in the block-based coding for bandwidth-hungry applications.

This paper focuses on the interactions between uplink admission and congestion control strategies in UTRA-W-CDMA. The proposed strategies are analysed in the framework of a representative multiservice scenario where conversational and interactive traffic are present and the presented solutions are compliant with 3GPP UTRA FDD specifications. Furthermore, the results consider a complete approach where all the radio resource management strategies are taken into account, including admission, congestion, short term RRM, handover and power control feasible solutions. Therefore, obtained results allow to devise a range of guidelines for the joint design of RRM algorithms.

In this paper, the authors reported the use of an e-learning system based on a mobile videophone in an education program at home for a student with severe physical impairments. System use for two months showed that the subject was able to attend school curricula and engage in discussion with teachers and classmates through the system.