Mobile Blog: Video, Media, Augmented Reality, Rich Media, IVVR

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In desktop video conferencing, the video conferencing application is normally bundled to an Instant Messaging software that includes text chat capabilities. Users can appoint or prearrange a video conference using textual chat. In contrast, the current evolution of video telephony in UMTS (Universal Mobile Telecommunications System) networks based on the circuit-switched 3G-324M service does not seamlessly combine video conferencing with other communication channels. The notion of video telephony in the mobile environment is nearer to standard voice calling than in the stationary world. Therefore, it is more likely that somebody will place a video call without prior announcement. This leads to privacy and inconvenience concerns. The callee might not want to be seen during a conversation for a variety reasons: A video call

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Even with a great deal of marketing, early attempts to convert users to the video telephony technology flopped (Jones and Marsden). In contrast, desktop video conferencing is incredibly popular for private person-to-person conversations and widely used for video conferencing in business environments such as telepresence for computer-supported cooperative work (CSCW).

In desktop video conferencing scenarios, typically a stationary computer is used. Camera and microphone are fixed and usually maintain the same distance from the person participating during the conversation. Moreover, lighting conditions are generally better than

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Media compression, error concealment measures, and the characteristics of wireless networks have side effects on the quality of 3G video telephony and IVVR (Interactive Voice & Video Response) applications.

3G-324M requires only the use of speech codecs. In contrast to audio codecs, speech codecs are designed for speech transmission within a narrow frequency range, making them inappropriate for transmission of music or a range of artificial sounds. This fact needs to be considered when designing IVVR applications

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3G video telephony generally operates over a single 64 kbit/s connection where both parties need to share the available bandwidth. Effectively, the application then is left with 60 kbit/s, or less that are dedicated for both media types, since H.245 call control messages reduce the gross bandwidth. In 3G-324M systems, the bandwidth is allocated dynamically; however, generally said, every party has 50% of the bandwidth available for sending audio and video signals. In a typical unidirectional scenario, 12.2 kbit/s are allocated for the speech codec, and a bitrate of 43-48 kbit/s is allowed for the video data (Sang-Bong, Tae-Jung and Jae-Won).

By employing rate control methods in the media encoders, the network can dynamically change these bitrates depending on network conditions and application demand. When two parties communicate simultaneously, the bitrates for the speech and video codec can be reduced in the encoders of both parties, keeping the overall bitrate below 64 kbit/s. For instance, when just one party shows speech activity, the speech bitrate for the other party can be reduced to a minimum where only comfort noise is generated on the receiver side (Holma and Toskala); AMR (Adaptive Multi-Rate) can perform these bitrate changes every 20ms. For video, the encoder can reduce the average bitrate by either reducing the frame rate or simply dropping frames during transmission. To increase the overall frame rate on the receiver side, the decoder can employ H.263 temporal scalability.

In 3G video telephony, the audio and video signals are bidirectionally streamed over dedicated circuit-switched W-CDMA (Wideband Code Division Multiple Access) paths. Streaming describes media is continuously being received or sent and played back on a terminal. Non-conversational one-way audio or video streaming requires a transport delay variation of below 2s (3GPP (3rd Generation Partnership Project)). In contrast, two-way video telephony introduces even higher real-time requirements with an end-to-end, one-way delay of below 150-400ms

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In the following I describe how the characteristics of wireless networks affect the audiovisual quality of 3G video telephony and especially IVVR (Interactive Voice & Video Response) applications. The impact of these characteristics need to be considered when designing IVVR applications and services:

Wireless networks are inherently error prone. Bitrates in wireless systems tend to fluctuate more as compared with wired networks. In wired networks, phenomena such as fading, shadowing, or reflection are non-existent so that, for the most part, the same bandwidth and much higher bandwidths are present during transmission. Influences on signal propagation cause the constant changing bandwidths in wireless systems. Generally, the receiving power depends on the distance between sender and receiver. The receiving power p decreases proportionally to the square of the distance between sender and receiver:

p=1/d²

where d is the distance between sender and receiver (Schiller).

Receiving power is influenced further by frequency dependent fading, shadowing, reflection at large obstacles, refraction depending on the density of the medium, scattering at small objects, and diffraction at edges.

The effect of multipath propagation can cause jitter when the radio signal reaches the receiver by two or more paths at different times. Moreover, the mobility of the user adds another set of problems that results in fading of received power over time; the channel characteristics change over time and location. This exacerbates the effect of multipath propagation because signal path change will be increased as the user changes his or her location. Changes in the distance between sender and receiver cause different delay variations of different signal parts.

The phenomenon of