Bamba--Audio and video streaming over the Internet

IBM Journal of Research and Development, Mar 1998 by Willebeek-LeMair, M H, Kumar, K G, Snible, E C

The World Wide Web has become a primary means of disseminating information, which is being presented increasingly through multiple media. The ability to broadcast audio and video information is becoming a reality with the advent of new media-streaming technologies. Most of the emerging streaming systems require high-bandwidth connections in order to deliver audio and video of suitable quality. In this paper we present a mediastreaming system, called Bamba, that delivers audio and video over low-bandwidth modem connections with the use of standard compression technologies. Bamba offers highquality audio and video over low-bit-rate connections and can operate using a standard HTTP server. The Bamba video is enhanced with special provisions for reducing the effect of errors in a lossy-network environment. Bamba adheres to existing standards wherever possible. Finally, Bamba has been fully implemented and deployed both internally at IBM and externally.

1. Introduction The World Wide Web (WWW) has become a primary means of disseminating information. Initially, the type of information distributed was primarily in the form of text and graphics. Later, images and stored audio and video files emerged. These audio and video files are downloaded from a server and stored at the client before they are played. Most recently, streamed audio and video have become available from both stored and live sources on the Web. Audio and video streaming enables clients to select and receive audio and video content from servers across the network and to begin hearing and seeing the content as soon as the first few bytes of the stream arrive at the client. Streaming technology involves audio and video compression, schemes for stream formatting and transmission packetization, networking protocols and routing, client designs for displaying and synchronizing different media streams, and server designs for content storage and delivery. In this paper we present a system for audio and video streaming (with code name Bamba) developed at the IBM Thomas J. Watson Research Center. Bamba has been deployed within IBM and was demonstrated externally on the official Web site of the 1996 Olympics. It has since been made available for free download from the IBM AlphaWorks* Web site.1

Today's computer-network infrastructures, including the Internet, were not designed with streaming in mind. Streaming media requires that data be transmitted from a server to a client at a sustained bit rate that is high enough to maintain continuous and smooth playback at the receiving client station. A primary objective in developing Bamba is to stream audio and video across the Web through very-low-bit-rate connections. Audio is sufficiently compressed to stream over modem connections at 14.4 Kb/s, and video at 28.8 Kb/s. The system that has been developed not only achieves the low-bit-rate goal, but can also be extended to support higher-bit-rate streams to provide higher-quality streaming over intranets or higher-bandwidth Internet connections. Furthermore, when streaming is not possible because of congestion or insufficient bandwidth availability, the Bamba player (client software) at the receiving client automatically calculates how much data to preload in order to maintain continuous playback. This allows clients connected via low-bit-rate connections to fall back to a download-andplay mode and still receive the higher-bit-rate content.

Existing audio and video streaming technologies In recent years, there has been much research and development in the areas of audio and video streaming as well as videoconferencing. Videoconferencing differs from audio and video streaming in that the communication is bidirectional, and end-to-end delays must be very low (

Audio and video streaming differs technically from its videoconferencing counterpart in that it can afford greater flexibility in end-to-end delays when the data is transmitted across a network and in the fact that stored content may be manipulated off-line with additional processing. These begin to merge when one considers live audio and video streaming applications (e.g., Internet, radio, and TV). The most relevant of the ITU standards is H.323, which defines audio/visual services over LANs for which quality of service cannot be guaranteed [5]. This standard specifies a variety of audio and video coders and decoders (CODECs) as well as signaling protocols to negotiate capabilities and set up and manage connections [6]. The underlying transport specified is the Real-time Transport Protocol (RTP) [7]. This protocol, defined by the IETF, is intended to provide a means of transporting real-time streams over Internet Protocol (IP) networks. A new protocol, the Real Time Streaming Protocol (RTSP), just proposed to the IETF, more directly addresses the issues of delivering and managing multimedia streams [8]. Clearly, this area is still evolving as new protocols are being defined and refined to satisfy a wide range of emerging networked multimedia applications. There are a large number of audio and video streaming systems available in the market today [9]. These include VDOLive**,2 StreamWorks**,3 Vosaic**,4 VivoActive**,5 InterVU**,fi, and RealAudio**.7 VDOLive, Streamworks, Vosaic, and RealAudio are based on proprietary client-server systems that transport their audio and video streams by means of User Datagram Protocol (UDP/IP) connections. This unreliable transport does not retransmit lost packets and is blocked by most firewalls unless they are specially reconfigured. The others use HTTP (based on TCP/IP) [10]. VDOLive employs a proprietary hierarchical compression technique that allows the server to adapt the video-stream bandwidth to the available network connection bandwidth. StreamWorks, Vosaic, and InterVu are based on MPEG** [11], while Vivo uses H.263 [12]. In general, these systems are designed to work over higher-bandwidth LAN connections and not at modem speeds. At modem speeds, the MPEG-based systems revert to slide-show-type video.