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Compressed Video Communications - Sadka A.

Sadka A. Compressed Video Communications - John Wiley & Sons, 2002. - 283 p.
ISBN: 0-470-84312-8
Download (direct link): compressedvideo2002.pdf
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by means of an IP-based transport network. The IP packets and all relevant overlying transport protocol headers are forwarded to the Subnetwork Dependent Convergence (SNDC) protocol layer which formats the network packets for transmission over the GPRS network. The SNDC protocol carries out header compression and the multiplexing of data from different sources. The Logical Link Control (LLC) layer operates above the Radio Link Control (RLC) layer to provide highly reliable logical links between the mobile station and the Serving GPRS Support Node (SGSN). Its main functions are specifically designed to maintain a reliable link. If the network packet size does not exceed the maximum LLC frame size (1520 octets), each IP packet is mapped onto a single LLC frame. The LLC frames are then passed onto the RLC/MAC (Medium Access Control) layer where they are segmented into fixed-length RLC/MAC blocks. At the MAC layer, multiple mobile stations are allowed to share a common transmission medium. GPRS allows each time slot to be multiplexed between up to eight users, and allows each user to use up to eight timeslots, thereby achieving great flexibility in the resource allocation mechanism. The RLC blocks are arranged into GSM bursts for transmission across the radio interface where the physical link layer is responsible for forward error protection, as described in Section 5.5.2. In the physical link layer, interleaving of radio blocks is performed and methods to detect link congestion are also employed. Figure 5.3 depicts the logical architecture of a GPRS network connection involving a Mobile Station (MS) and a Base Station Subsystem (BSS).
The GPRS service introduced in the GSM system is an intermediate step towards the third-generation UMTS network. EGPRS (Enhanced GPRS) is an enhanced version of GPRS that allows for a considerable increase in throughput availability to a single user given enough traffic availability from active sources and benign interference conditions. This implies that EGPRS can provide video services with higher data rates than is possible with GPRS. EGPRS uses the same
Application IP/ X23
Figure 5.3 GPRS logical protocol architecture
protocol architecture of GPRS described above, with improvements of the modulation scheme employed in the EDGE (Enhanced Data rate GSM Evolution) radio interface that lead to the increase in throughput availability. Similarly, UMTS uses an innovative radio access approach to increase the available capacity of the radio interface. The UMTS infrastructure is integrated with GSM so that the UMTS core network can perform both the circuit- and packet-switching functions. However, the major technological innovations of UMTS are incorporated in the packet-switched IP nodes. The structure of the packet switched part of the UMTS core network is similar to that of the GPRS described above, where the BSS access segment is replaced by the UTRAN (Universal Terrestrial Radio Access Network) access network that is based on W-CDMA (Wideband Code Division Multiple Access) technologies. The connection between the UMTS core network and UTRAN access network is guaranteed by a new interface called Iu, which specialises in managing both the packet-switched and the circuit-switched components. The main improvements achieved by UMTS compared to GPRS are in the IP mobility management and the quality of service control. UMTS offers a range of QoS levels that are suitable for real-time video communications, namely those specified in the conversational and streaming classes. The main feature that defines the capability of a QoS class to accommodate a real-time video service is its sensitivity to delay. The conversational class allows videoconferencing sessions in which the delay factor must be minimised and the temporal relationship between various streams (voice and video for instance) must be maintained stationary. In the streaming class that allows for real-time streaming of multimedia data, the requirement for low transfer delay is not stringent but the various stream components must be kept temporally aligned. In addition to the conversational and streaming classes, UMTS offers the interactive QoS class which enables the mobile user to interact with a remote device on the network such as a video database or a website. The main requirements of this class are a limited round-trip delay and data integrity represented by low bit error rates.
5.5 QoS Issues for Packet Video over Mobile Networks
In real life, transmitted video packets are subject to loss and the contained information is susceptible to bit errors. When packets are corrupted, any one of three possible kinds of error might result. If the sequence number of the packet is affected, the decoder becomes unable to figure out the correct order of packet transmission. As a result, the depacketiser fails to merge the information of consecutive video packets in order to properly reconstruct the video sequence. This has a damaging effect on the video quality regardless of whether or not the data bits of affected packets have arrived intact. The second kind of error arises when some of the payload of a certain video packet is hit by errors in such a way
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