5.3 Multimedia Communication

< Day Day Up >

Multimedia communication knows two principle ways of delivery. The multimedia files are downloaded and played back, or the multimedia data is usually sent across the network in streams. Streaming breaks multimedia data into packets with sizes suitable for transmission between the servers and clients. The real-time data flows through the transmission, decompressing and playing, just like a water stream. A client can play the first packet and decompress the second while receiving the third, just as in a water pipeline. Thus, the user can start enjoying the multimedia without waiting to the end of transmission. Modern software frameworks, such as the Java Media Framework (JMF), comfortably support both principles (see Section 5.3).

This section discusses first the issues that relate mainly to multimedia communication in its streamed form. We will then discuss networks and network protocol issues. Finally, security aspects are assessed.

For a distributed multimedia system, several features should be of concern:

Remote control: allowing long-distance execution, manipulation, and synchronization of multimedia objects
Unreliable communication handling: taking care of possible data loss or data delays including insertion of frames, frame composition, and so forth
Adaptation of the storage and streaming to different applications, to different network bandwidth and to different presentation devices
Data compression and decoding for multiple platforms, which will concern different data representations
Secure communication, authentication, and copyright protection

Heterogeneous network topologies create delivery paths through different network subsystems from the multimedia servers to the individual clients. As intercontinental network links provide, in time of low congestion, sufficient bandwidths for video streaming (several megabits per second), the network to the home scales for most cases only to the order of 1 mega-bit per second for Asymmetric Digital Subscriber Line (ADSL) and Very High Bit Rate DSL (VDSL) technologies.

However, despite these technological advances, raw bandwidth is not enough for effective delivery of multimedia services, especially when this bandwidth is shared among several systems or applications. It is necessary to provide guaranteed QoS for a flawless end-to-end service; that is, so that the end user receives a continuous flow of audio or video without interruption or noticeable degradation of quality. ^[96]

Communication protocols used for multimedia transmission play an important role in achieving the above goals; that is, to provide high bandwidth to the user with the appropriate QoS.

5.3.1 Reservation-Based versus Reservationless Protocols

Transmission over the Internet with the use of the currently broadly used communication IPv4 protocol family cannot guarantee on-time delivery of time-sensitive information. Hence, two main classes of QoS-aware protocols have been defined.

Reservation-based: In this model, resources are reserved explicitly. The network classifies incoming packets and uses the reservations made to provide a differentiated service. Typically, a dynamic resource reservation protocol is used in conjunction with admission control to make reservations. In the Resource Reservation Protocol, RSVP, ^[97] resources are reserved for unidirectional data going from sender to receiver. The new Internet protocol (IP) generation, IPv6, ^[98] proposes both reservation-based and reservationless QoS. The integrated services model, the int-serv model, in IPv6 reserves resources explicitly using a dynamic signaling protocol. It employs admission control, packet classification, and intelligent scheduling to achieve the desired QoS. Protocols of this type lack the implementation support, as broadly used router technology allows no reservation of a delivery path.

Reservationless: In this model, no resources are explicitly reserved. Instead, traffic is differentiated into a set of classes, and the network provides services to these classes based on their priority. It is necessary to control the amount of traffic in a given class that is allowed into the network so as to preserve the QoS being provided to other packets of the same class. For example, IPv6 ^[99] uses for its diff-serv model the so-called Flow Label and Priority fields in the IP header. A host may use them to identify those packets for which it requests special handling by routers. This presupposes that network nodes have to use intelligent queuing mechanisms to differentiate traffic. ^[100] Several commercial router producers adapted this new protocol family (http://playground.sun.com/pub/ipng/html/ipng-implementations.html). However, old technologies that are not able to differentiate traffic are yet in worldwide use.

5.3.2 Real-Time Delivery Supporting Protocols

In addition to reservation-based protocols, there exits a class of protocols that support the transport of continuous media. ^[101] The most prominent member is the Realtime Transport Protocol (RTP). RTP is an IP-based protocol providing support for the transport of real-time data such as video and audio streams. The services provided by RTP include time reconstruction, loss detection, security, and content identification. RTP is designed to work in conjunction with the auxiliary control protocol RTCP to get feedback on quality of data transmission and information about participants in the ongoing session.

On top of RTP is the Real-Time Streaming Protocol (RTSP).^[102] RTSP is a C-S multimedia presentation protocol to enable controlled delivery of streamed multimedia data over IP networks. It provides VCR-style remote control functionality for audio and video streams, such as pause, fast forward, reverse, and absolute positioning. RTSP is an application-level protocol designed to work with lower-level protocols such as RTP and RSVP to provide a complete streaming service over Internet. It provides a means for choosing delivery channels (such as UDP [User Datagram Protocol], multicast UDP, and TCP [Transmission Control Protocol]) and delivery mechanisms based on RTP.

5.3.3 Multimedia Session Protocols

In addition to real-time supporting and reserving protocols, mechanisms for setting up multimedia sessions are important. In principle, no formal session that sets up mechanisms is needed for multicast communication — senders send to a group address, receivers subscribe to the same address, and communication ensures. However, in practice, we normally need means for users that wish to communicate to discover which multicast address to use, to discover which protocols and codecs to transmit and receive with, and to discover that a session is going to take place at all. Note that session management may be described by MPEG-21 descriptors (see Chapter 3). The view is the following: MPEG-21 provides the open framework, that is, the application interface for the system, and a session protocol provides the implementation of this interface.

The Internet Engineering Task Force (IETF) Section Initiation Protocol (SIP)^[103] protocol is currently the most promising protocol for session initiation and management available.

There are different entities defined in SIP. At first there are the SIP user agents (UAs), which could be separated into UA servers and UA clients. UA clients initiate a call, whereas UA servers only react to calls. Both are statefull devices. "State full" means that in the terminal the state of a call has to be known. The second important entity in SIP is the SIP network server, which could act as proxy or as redirect server. Because of scalability, the network server in the core net should be stateless, whereas the edge servers have to be statefull to provide all required functionality. Location server and register server complete this enumeration.

The session concept of SIP offers a generic way to establish, control, and tear down multimedia sessions between communication end-points. At the time, SIPs mostly concentrate on the implementation of voice over IP (VoIP) services, but there is much ongoing research to show that SIP sessions can be used to establish multimedia streaming sessions as well as interactive multimedia communications. ^[104]

5.3.4 Secure Multimedia Communication

It is also imperative that the network infrastructure support security to appropriate levels to protect customers and ensure privacy. ^[105] The aim of security techniques in this field ranges from authentication and verification to classical encryption and, recently, to copyright protection. The proposal of copyright protection mechanisms is one of the main standardization aims in MPEG-21 ^[106] (Intellectual Property Management; see Chapter 3).

Application-specific data structures are exploited to create more efficient encryption systems. Relevant work for MPEG encryption can be found in Shi and Bhargava ^[107] and Tang. ^[108] Digital watermarking is a recent technology ^[109], ^[110], ^[111] that covers more issues than related approaches; that is, it can be used for copyright protection of digital content, multimedia data authentication, and other purposes. Here, the problem is to embed an imperceptible mark in multimedia data in a secure and robust way so as to establish ownership claims or prove integrity. Watermarking does not depend on any specific data representation, for example, file format, and tolerates distortions resulting from common image processing operations, which is a major robustness requirement.

5.3.5 Reliable Multimedia Communication

Reliable multimedia communication introduces an error-control mechanism. This is mostly used with error-prone wireless links. Two basic approaches are employed, the forward error correction (FEC) and automatic repeat request (ARQ). FEC adds parity bits to the transmitted packets, and this redundancy is used by the receiver to detect and correct errors. FEC maintains constant throughput and has bounded time delay. ARQ only provides error detection capability by requesting retransmission when errors are detected by the receiver. ARQ is simple, but the delay is variable and unbounded. Many alternatives to FEC and ARQ have also been proposed. ^[112]

Supplemental and regularly updated information on multimedia communication techniques, including internet transmission protocols, networks, media packaging, encryption, and watermarking may be found at http://streamingmedialand.com/technology_frameset.html.

^[96]Granville, L.Z., Fleischmann, R.U., Tarouco, L.M.R., and Almeida, M.J.B., Management of networks with end-to-end differentiated service QoS capabilities, in International Workshop on Advances in Information System, Izmir, October 2000, LNCS 1909, Springer-Verlag, New York, pp. 147–158.

^[97]Braden, R., Zhang, L., Berson, S., Herzog, S., and Jamin, S., Resource reservation protocol (RSVP)—version 1 functional specification, Internet RFC 2205, 1997.

^[98]Goncalves, M. and Niles, K., IPv6 Networks, McGraw-Hill, New York, 1998.

^[99]Goncalves, M. and Niles, K., IPv6 Networks, McGraw-Hill, New York, 1998.

^[100]Basturk, E., Birman, A., Delp, G., Guérin, R., Haas, R., Kamat, S., Kandlur, D., Pan, P., Pendarakis, D., Peris, V., Rajan, R., Saha, D., and Williams, D., Design and implementation of a QoS capable switch-router. Comput. Networks, 31, 19–32, 1999.

^[101]Zheng, H. and Boyce, J., An improved UDP protocol for video transmission over internet-to-wireless networks, IEEE Trans. Multimedia, 3, 356–365, 2001.

^[102]http://www.rtsp.org/.

^[103]http://www.ietf.org/ids.by.wg/simple/html.

^[104]Stadler, J., Miladinovic, I., and Pospischil, G., SIP for Unified Data Exchange (SIP4UDE), in Proceedings of the 1st IEEE Workshop for Applications and Services in Wireless Networks 2002 (ASWN2002), Paris, France.

^[105]Guo, G.D., Li, S.Z., and Zhang, H.J., Distance-from-boundary as a metric for texture image retrieval, in Proceedings of the International Conference on Acoustics, Speech, and Signal Processing (ICASSP), Salt Lake City, May 2001, IEEE CS Press.

^[106]Hill, K. and Bormans, J., Overview of the MPEG-21 Standard. ISO/IECJTC1/SC29/WG11 N4041 (Shanghai Meeting), October 2002, http://www.chiariglione.org/mpeg/.

^[107]Shi, C. and Bhargava, B., A fast MPEG video encryption algorithm, in 6th ACM International Multimedia Conference (ACM Multimedia'98), Bristol, September 1998, pp. 81–88.

^[108]Tang, L., Methods for encrypting and decrypting MPEG video data efficiently, in Proceedings of the Fourth ACM Multimedia Conference (MULTIMEDIA'96), New York, November 1996, ACM Press, pp. 219–230.

^[109]Zhao, J. and Koch, E., A digital watermarking system for multimedia copyright protection, in Proceedings of the 4th ACM International Multimedia Conference (MULTIMEDIA'96), Boston, November 18–22, 1996, pp. 443–444.

^[110]Voyatzis, G., Nikolaidis, N., and Pitas, I., Digital watermarking: an overview, in 9th European Signal Processing Conference (EUSIPCO'98), Island of Rhodes, September 8–11, 1998, pp. 9–12.

^[111]Dittmann, J., Stabenau, M., and Steinmetz, R., Robust MPEG video watermarking technologies, in Proceedings of the 6th ACM International Multimedia Conference (ACM Multimedia'98), Bristol, September 1998.

^[112]Liu, H., Ma, H., El Zarki, M., and Gupta, S., Error control schemes for networks: an overview, Mobile Networks Appl., 2, 167–182, 1997.

< Day Day Up >