# US 12,015,875 B2 · Method and system for real time audio in multi-point video conferencing

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- Number: US 12,015,875 B2
- Document ID: US12015875B2
- Status: Granted
- Kind: B2
- Application number: 17/551,339
- Filed: 2021-12-15
- Published: 2022-06-23
- Granted: 2024-06-18
- Earliest claimed: 2020-12-18
- Assignees: Pexip AS
- Applicants: Pexip AS
- Inventors: Håvard Graff; Tulio Beloqui
- Web page: https://patentgrep.com/patent/US12015875B2

## CPC

- H04N 7/152: {Multipoint control units therefor}
- G10L 19/0017: {Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error (G10L19/24 takes precedence)}
- H04L 12/1827: {Network arrangements for conference optimisation or adaptation}
- H04L 65/65: Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
- H04L 65/70: Media network packetisation
- H04N: PICTORIAL COMMUNICATION, e.g. TELEVISION
- G10L: SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- H04L: TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION (arrangements common to telegraphic and telephonic communication H04M)

## Abstract

A real time audio multi-point videoconferencing system comprising at least two videoconferencing endpoints in communication with a multipoint control unit, and a method of providing the real time audio. The method comprising capturing input lossless digital audio streams on each of the at least two videoconferencing endpoints, transmitting the 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 input lossless digital audio streams 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, and playback the plurality of output lossless digital audio streams on each of the at least two videoconferencing endpoints.

## Claims

### Claim 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.

### Claim 2 (depends on 1)

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.

### Claim 3 (depends on 1)

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.

### Claim 4 (depends on 1)

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.

### Claim 5 (depends on 1)

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).

### Claim 6 (depends on 1)

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).

### Claim 7 (depends on 1)

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.

### Claim 8 (depends on 1)

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.

### Claim 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.

### Claim 10 (depends on 9)

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.

### Claim 11 (depends on 9)

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.

### Claim 12 (depends on 9)

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.

### Claim 13 (depends on 9)

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).

### Claim 14 (depends on 9)

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).

### Claim 15 (depends on 9)

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

### Claim 16 (depends on 9)

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