WWW: Beyond the Basics

11. Real-time Audio and Video

11.4 Network for Real-time Applications

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.

11.4.1 Real-time Network Requirement

Table 2 gives required transmission rates for audio and video.


Table 2. Typical Audio and Video Transmission Rates

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.


11.4.2 Real-time Networks

Networks are divided into three categories according to the distance between end-systems (computers). They are Local Area Networks (LANs), Metropolitan Area Networks (MANs), and Wide Area Networks (WANs). Before discussion LANs and WANs as real-time audio and video carriers, we first introduce some basic concepts to understand network.

11.4.2.1 Some Basic Concepts


11.4.2.2 LANs

LANs are generally used on campuses and in companies which connect local computers together. The three conventional LAN technologies are the Ethernet , Token Passing Ring (TR), and FDDI. All are connectionless networks. Their access speed are shown in the following.

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.

  1. 100 Mbps Ethernet. Two modes exist. The first is called 100 Base-T fast Ethernet and uses the regular Ethernet access method operating at 100 Mbps. The other 100 Mbps Ethernet mode is called 100 VG-Any LAN.
  2. FDDI-II, which is an additional enhancement of FDDI to fully support real-time multimedia services.

11.4.2.3 IP WANs

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.

11.4.2.4 ATM

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:

  • No error protection or flow control on a link-to-link basis.
  • ATM operates in a connection-oriented mode.
  • The header functionality is reduced.
  • The information field length is relatively small.

    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.

    [PREV] [NEXT] [UP] [HOME] [VT CS]

    <shaohong@csgrad.cs.vt.edu>
    Last modified: Sun Dec 8 1996