Method and system for real time audio in multi-point video conferencing

1. A method of providing real time audio in a multi-point videoconferencing system having at least two videoconferencing endpoints in communication with a multipoint control unit (MCU), the method comprising:

capturing one or more input lossless digital audio streams on each of the at least two videoconferencing endpoints;

transmitting the one or more input lossless digital audio streams from each of the at least two videoconferencing endpoints to the MCU using a frame size in the range 0.25 ms-3 ms;

mixing, using the MCU, the one or more input lossless digital audio streams from each of the at least two videoconferencing endpoints into a plurality of output lossless digital audio streams;

transmitting a different one of the plurality of output lossless digital audio streams to each of the at least two videoconferencing endpoints using a frame size in the range 0.25 ms-3 ms;

playing back the plurality of output lossless digital audio streams on each of the at least two videoconferencing endpoints; and

compressing the one or more input lossless digital audio streams and the plurality of output lossless digital audio streams prior to transmission.

2. The method of claim 1 , wherein the step of transmitting the one or more input lossless digital audio streams and the plurality of output lossless digital audio streams further comprises using a single frame buffer.

3. The method of claim 1 , further comprising using a three-frame buffer for receiving the one or more input lossless digital audio streams from each of the at least two videoconferencing endpoints at the MCU and for receiving one of the plurality of output lossless digital audio streams at each of the at least two videoconferencing endpoints.

4. The method of claim 1 , wherein the step of transmitting the one or more input lossless digital audio streams from each of the at least two videoconferencing endpoints and the plurality of output lossless digital audio streams further comprises transmitting a bit rate of 200 kbps-1 Mbps.

5. The method of claim 1 , further comprising transmitting the one or more input lossless digital audio streams and the plurality of output lossless digital audio streams using Pulse-Code Modulation (PCM).

6. The method of claim 1 , further comprising transmitting the one or more input lossless digital audio streams and the plurality of output lossless digital audio streams using Real-time Transport Protocol (RTP).

7. The method of claim 1 , wherein the step of compressing the one or more input lossless digital audio streams and the plurality of output lossless digital audio streams prior to transmission, comprises applying Huffmann encoding on the plurality of input lossless digital audio streams and the plurality of output lossless digital audio streams.

8. The method of claim 1 , wherein the step of capturing the one or more input lossless digital audio streams on each of the at least two videoconferencing endpoints further comprises bypassing any echo cancellation algorithm of the at least two videoconferencing endpoints.

9. A real time audio multi-point videoconferencing system having at least two videoconferencing endpoints in communication with a multipoint control unit (MCU), wherein

each of the at least two videoconferencing devices comprises:

an audio capturing unit adapted to capture and encode audio into one or more input lossless digital audio streams, and to compress the input lossless digital audio streams;

a transmitting unit adapted to transmit the one or more input lossless digital audio streams using a frame size in the range 0.25 ms-3 ms;

a receive unit adapted to receive an output lossless digital audio stream transmitted with a frame size in the range 0.25 ms-3 ms; and

a playback unit adapted to playback the received output lossless digital audio stream; and the MCU comprises

a receive unit adapted to receive one or more input lossless digital audio streams from each of the at least two video conferencing devices, transmitted with a frame size in the range 0.25 ms-3 ms;

a mixing unit adapted to mix the one or more input lossless digital audio streams from each of the at least two video conferencing devices into a plurality of output lossless digital audio streams; and

a transmitting unit adapted to transmitting a different one of the plurality of output lossless digital audio streams to each of the at least two videoconferencing endpoints using a frame size in the range 0.25 ms-3 ms.

10. The system of claim 9 , wherein the transmitting unit of each of the at least two videoconferencing endpoints and the transmitting unit of the MCU is adapted to transmit the respective input and output lossless digital audio streams using a single frame buffer.

11. The system of claim 9 , wherein the receive unit of each of the at least two videoconferencing endpoints and the transmitting unit of the MCU is adapted to receive the respective output and input lossless digital audio streams using a three-frame buffer.

12. The system of claim 9 , wherein the transmitting unit of each of the at least two videoconferencing endpoints and the transmitting unit of the MCU is adapted to transmit the respective input and output lossless digital audio stream using a bit rate of 200 kbps-1 Mbps.

13. The system of claim 9 , wherein the transmitting unit of each of the at least two videoconferencing endpoints and the transmitting unit of the MCU is adapted to transmit the respective input and output lossless digital audio stream using Pulse-Code Modulation (PCM).

14. The system of claim 9 , wherein the transmitting unit of each of the at least two videoconferencing endpoints and the transmitting unit of the MCU is adapted to transmit the respective input and output lossless digital audio stream using Real-time Transport Protocol (RTP).

15. The system of claim 9 , wherein the audio capturing unit is adapted to compress the input lossless digital audio stream by applying Huffmann encoding.

16. The system of claim 9 , wherein the audio capturing unit is adapted to bypass any echo cancellation algorithm of the videoconferencing endpoint.