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Audio

In principle, the distribution of audio to conference participants is somewhat simpler than the distribution of video, since audio compression and decompression is well within the capabilities of even low power workstations, and even (uncompressed) PCM audio at 64kb/s is within the capability of many networks. However, this situation is not quite as simple as may be assumed at first sight. In almost all conferences, audio is the single most important communication medium. It is more sensitive to delay than video (assuming that lip synchronisation is not an absolute priority) if two way communication is required, and is much more sensitive to jitter and packet loss than properly packetised video. This may be surprising at first, but in the face of high packet loss, video can compromise quality and resort to pure intra compression of pictures which (from a user's point of view) is relatively unaffected by packet loss. Audio has no such option. Recent tests of the Ebone UK-Paris and UK-Stockholm links show around 20-30% packet loss at 64kb/s, which results in largely incomprehensible audio.

Generally speaking, audio should be multicast directly to all receiving sites from the speaker's machine. Assuming the Mbone (Multicast Tunnel Backbone) has been configured to utilise the best quality links, regardless of their geographical routing, this will subject audio to the minimum delays and packet loss. There is less of a format conversion problem than with video, as most workstations can perform the decoding of whichever format the originator chooses in software.

However, the CMMC will not be completely unused for audio distribution for several reasons:

1. Remote sites may be connected by ISDN using hardware codecs or video phones which combine the audio and video data into a single H.221 serial stream. The CMMC will separate this audio from the video stream, and then multicast to directly to the other receiving sites.

2. A remote site may participate using only a normal telephone to connect to an ISDN line at the CMMC, and then receive just the audio from the conference. This may be used to supplement the shared workspace where the loss or jitter on the packet network is too great for comprehensible audio transmission.

3. Some sites may require very low bandwidth feeds in order to overcome local bandwidth limitations. Although such feeds would ideally be provided from the nearest remote site with adequate bandwidth, it is much easier for such a service to be provided automatically from the CMMC. As with other CMMC functionality, this will be distributed or replicated in the long run.

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