Tandem-free operation for VoIP conference bridges

Traditional telephone conferencing has been accomplished by way of a centralized conference bridge. The tandem arrangement of high compression speech codecs in conventional VoIP conference bridges lead to speech distortions and require a substantial number of computations. Decentralized architectures avoid the speech degradations and delay, but lack strong control and depend on silence suppression to make the endpoint bandwidth and processing requirements scalable. One solution is to use centralized speaker selection and forwarding, and decentralized decoding and mixing. This approach eliminates the problem of tandem encodings but maintains centralized control, thereby improving the speech quality and scalability of the conference. This paper considers design options and solutions for this model in the context of modern IP telephony networks. Performance was evaluated with real conferees over live conferences using a PC-based conferencing test bed, built using a custom softwarebased bridge and a third-party endpoint. Conferees strongly preferred the speech quality of the new arrangement to that of a conventional VoIP conference bridge.