This section presents issues involved in the computer networks that support real-time audio and video applications. The requirements of real-time applications are large network bandwidth and low latency bulk data transfer. If the application involves more than two parties, then multipoint communications need be considered.
Table 2 gives required transmission rates for audio and video.
INFORMATION TYPE |
BIT RATE | QUALITY AND REMARKS |
---|---|---|
VIDEO | 64-128Kbps 384 Kbps - 2 Mbps 1.5 Mbps 5-10 Mbps 34/45 Mbps 50 Mbps or less 100 Mbps or more |
Video telephony (H.261) Videoconferencing (H.261) MPEG-1 TV quality (MPEG-2) TV distribution HDTV quality Studio-to-studio HDTV video downloading |
AUDIO | p * 64 Kbps | 3.1 KHz, or 7.5 KHz, or hi-fi baseband signals |
Audio and video data require very high transfer rates or bandwidth even when the data is compressed. For example, an MPEG-1 session requires a bandwidth of about 1.5 Mbps. Not only does the transfer rate have to be high, it must also be predictable. The traffic pattern of multimedia data transfer is stream-oriented, and the network load is long and continuous.
Because audio and video data must be synchronized when it arrives at the destination site, networks should provide synchronized transmission, thus, audio and video networks must provide the low latency transmission.
Traditional networks are used to provide error-free transmission. However, most multimedia applications can tolerate errors in transmission due to corruption or packet loss without retransmission or correction. In some cases, to meet real-time delivery requirements or to achieve synchronization, some packets are even discarded. As a result, we can apply light weight transmission protocols to audio and video networks. These protocols cannot accept retransmission, since that might introduce unacceptable delays.
In audio and video networks, some communications are multipoint, which
means that communications involve more than two participants. For
example, videoconferencing often holds for several groups of participants.
Multipoint communications can be implemented by using multicasting
technology, which replicates a single input signal and delivers it to
multiple destinations.
The bit rate (bits per second) given above assumes that only two connected stations communicate. If many stations use the shared medium, the bit rate is much different. For example, for a hundred stations transmitting frames of 1000 bits on average over a 10 Mbps Ethernet, the maximum aggregate bit rate is 3.6 Mbps. The conclusion is: Individual Ethernet or Token Rings connecting a hundred stations have insufficient bandwidth for real-time transmission of digital audio and video, except for compressed telephone-quality sound.
Several options are available to enhance the performance of the current LAN technologies in order to better support real-time applications. The following gives two examples.
WANs typically covers entire countries. Two main WANs, which are used for real-time audio and video transmissions, are Internet and ATM B-ISDN.
The Internet Protocol (IP) is a packet-switched technology based on the connectionless mode. IP can operate over virtually any underlying transport mechanism, at bit rates ranging from 300 bps to 100 Mbps. Because IP based on connectionless mode, it can not reserve any resources. This may result in unpredictable bit rates when supporting real-time applications. However, for most real-time applications, the achievable bit rate and transit delays are sufficient.
Two transport layer protocols, transmission control protocol (TCP) and user datagram protocol (UDP), were developed with IP. TCP provides a reliable end-to-end service by using error recovery. TCP is not practical because error recovery mechanism will increase network latency.
UDP is an unreliable service which does not prevent transfer errors. So UDP use less data transfer time. This property satisfies real-time audio and video communication. Many applications that operate over the Internet use UDP.
The advantage of the IP technology lies in the support of multicasting. Multicast Backbone (MBone) is a physical world-wide implementation of the IP multicasting technology over the Internet.
The other transport protocol is Resource Reservation Protocol (RSVP).
It was designed in 1994 to improve the support of real-time applications by
IP networks. RSVP allows resources to be reserved without a call
set-up mechanism.
Broadband Integrated Services Digital Network (B-ISDN) is a network concept which represents the extension of the narrowband ISDN . The goal of B-ISDN is to define an application interface and a corresponding WAN with conversational, distributed, messaging and query services of different bandwidth requirements. Further, the goal of B-ISDN is to provide connectionless and connection-oriented services for transmission of different media.
In 1988, asynchronous transfer mode (ATM) was selected as
the technology to support B-ISDN. ATM is
a packet-switching technology based on connection-oriented mode,
though a connectionless service can be emulated. It allows the
systems to operate at a much higher rate than the usual packet switching
system. The higher rate is
achieved because of the following characteristics:
Before data are transferred from a multimedia terminal to the network, a logical (or virtual) connection setup phase must allow the network to perform a reservation of the necessary resources. If no sufficient resources are available, the connection is refused and the requesting terminal is notified. The ATM networks are suitable, with their speed and bandwidth characteristics, to offer a transmission service for the real-time audio and video applications.
For more information, see Fluckiger.
This section described the network requirement of real-time audio and video and currently used network technology. In the next section we introduce two real-time audio and video systems: telephony and videoconferencing.
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Last modified: Sun Dec 8 1996