Audio/Video Conferencing

The A/V Conferencing service provides multiplexing of audio and video media. In the case of audio, the A/V Conferencing Server mixes the audio feeds from every participant before returning the mixed audio to each participant.

The A/V Conferencing Server uses the real-time audio (RTAudio) codecs for audio and real-time video (RTVideo) codecs for video. Both protocols are designed to optimize performance in high-latency, low-bandwidth networks such as the Internet. Two-way communications are peer to peer. Therefore, for voice calls (which make up the large majority of audio communications), the A/V Conferencing Server is not involved.

The protocol used by the Audio/Video Conferencing service is secure real-time transport protocol (SRTP) over User Datagram Protocol (UDP) (SRTP/UDP). SRTP/UDP uses the port range 49152-65535.