How AES-67, the New Audio-over-IP Standard, Will Bring the Convergence of Telecommunications, Intercom, Radio and Television Broadcast Studio Audio

Traditionally, due to previous practical technical limitations, the audio quality of telecommunications and intercom systems was not as high as studio audio. Applications requiring long distance high quality audio required the use of specialized provisioning and equipment in parallel to the existing telecommunications systems. — IP computer networks have long since erased the distinction between the local LAN and global networks. As high speed wide area networks (WANs) with better performance and reliability have come online, the reality of erasing the difference of local vs remote audio presents itself. — AES67 is the protocol designed to take advantage of this capability for audio. Using AES67, the audio quality of telecommunications and intercommunications can be the same as in-studio audio, and furthermore, the systems directly interconnected by using the single interoperability protocol. — This shift is more significant than eliminating the economic redundancy of parallel systems. It fundamentally enables new workflows, coordinating and combining the efforts of production staff and talent, in geographically combined or diverse locations. — Audio traffic is no longer just communication; it can be contribution as well. — Combining the ease of making a connection like a phone call, the practically unlimited flexibility of routing of the network, with the pristine high fidelity of digital studio audio, AES67 brings the convergence of telecom, radio and television studio audio and intercom.

Published

2015-07

Content type

Original Research

Keywords

AES67, convergence, least common denominator audio, SIP, WAN, telecom, intercom, radio, TV audio, broadcast production

DOI

10.5594/M001620

ISBN

978-1-61482-955-3