Voice Conferencing over IP Networks

Traditional telephone conferencing has been accomplished by way of a centralized conference bridge. An Internet Protocol (IP)-based conference bridge is subject to speech distortions and substantial computational demands due to the tandem arrangement of high compression speech codecs. Decentralized architectures avoid the speech distortions and delay, but lack strong control and have a key dependence on silence suppression for endpoint scalability. One solution is to use centralized speaker selection and forwarding, and decentralized decoding and mixing. This approach eliminates the problem of tandem encodings and maintains tight control, thereby improving the speech quality and scalability of the conference. This thesis considers design options and solutions for this model, and evaluates performance through live conferences with real conferees. Conferees found the speaker selection of the new conference model to be transparent, and strongly preferred the resulting speech quality to that of a centralized IP-based conference bridge.