System and method for testing real-time communications between browsers and contact centers

Patent metadata

Field	Value
Publication number	US-12375606-B2
Application number	US-202418754746-A
Country	US
Kind code	B2
Filing date	Jun 26, 2024
Priority date	Aug 20, 2020
Publication date	Jul 29, 2025
Grant date	Jul 29, 2025

Abstract

A system and method for bi-directional quality testing of web real-time communications (WebRTC) sessions. In an embodiment, the system and method comprise an operator network, cloud contact center, cloud contact center agent application, and a synthetic software agent comprised of agent automation software, injected API shim code, virtual audio devices, audio processing applications, and media servers, capable of performing automated and to end communication testing. In order to provide end to end testing, especially with respect to voice quality, the synthetic agent software may control and monitor the audio channels (both send and receive) of the browser communication session.

First claim

What is claimed is:

1. A system for bi-directional quality testing of web real-time communications (WebRTC) sessions, comprising:
a WebRTC application programming interface (API) shim operating in a web browser of a computing device and configured to:
intercept a WebRTC API call from a client application operating a WebRTC API in a web browser of the computing device;
inject redirection code into the WebRTC API, the redirection code comprising instructions to:
connect with an external media server, and establish a real-time transport protocol (RTP) media connection from the client application to an external communication device through the external media server; and

pass the WebRTC API call to the WebRTC API for handling according to the redirection code; and

the external media server operating outside of the web browser and configured to:
accept the connection from the WebRTC API of the web browser;
establish the real-time transport protocol (RTP) media connection from the client application to the external communication device through the external media server;
capture either incoming media in the form of audio data or video data from the external communication device, or outgoing media in the form of audio data or video data from the client application, or both; and
send the captured media to an audio/video quality analyzer; and

the audio/video quality analyzer comprising separate media server software operated in tandem with WebRTC mechanisms employed by a contact center application running inside of the browser and configured to:
receive the captured media; and
determine an audio quality or a video quality of the captured media; and

send the determined audio quality or video quality of the captured media to the external media server.

2. The system of claim 1 , wherein the external communication device is located at a cloud contact center, and a traversal using relays around network address translation server is used to relay requests from the cloud contact center to a computer operating a WebRTC engine.

3. The system of claim 1 , wherein:
the system further comprises a software virtualized speaker operating on a computing device and configured to operate in accordance from instructions contained in the redirection code; and
the redirection code further comprises instructions for the software virtualized speaker to:
take priority of operation of audio output over other connected audio devices, if any, attached either to the computing device or to the external communication device;
generate synthetic audio; and

inject the synthetic audio into the real-time transport protocol (RTP) media connection from the client application to the external communication device through the external media server, either as the incoming media or the outgoing media.

4. The system of claim 3 , wherein:
the system further comprises a software virtualized microphone operating on a computing device and configured to operate in accordance from instructions contained in the redirection code; and
the redirection code further comprises instructions for the software virtualized microphone to:
take priority of operation of audio input over other connected audio devices, if any, attached either to the computing device or to the external communication device; and
capture the synthetic audio generated by the software virtualized speaker.

5. The system of claim 1 , wherein the external media server handles both media processing and agent automation for the purposes of testing automated agent or client responses in the real-time transport protocol (RTP) media connection from the client application to the external communication device through the external media server.

6. The system of claim 1 , wherein the redirection code is a polyfill of API code that bypasses the web browser's internal media server for purposes of calculating voice quality outside of the limitation of the browser-native WebRTC implementation.

7. A method for bi-directional quality testing of web real-time communications (WebRTC) sessions, comprising the steps of:
operating a WebRTC application programming interface (API) shim in a web browser of a computing device to perform the steps of:
intercepting a WebRTC API call from a client application operating a WebRTC API in a web browser of the computing device;
injecting redirection code into the WebRTC API, the redirection code comprising instructions to:
connect with an external media server, and
establish a real-time transport protocol (RTP) media connection from the client application to an external communication device through the external media server; and

passing the WebRTC API call to the WebRTC API for handling according to the redirection code; and

operating the external media server outside of the web browser to perform the steps of:
accepting the connection from the WebRTC API of the web browser;
establishing the real-time transport protocol (RTP) media connection from the client application to the external communication device through the external media server;
capturing either incoming media in the form of audio data or video data from the external communication device, or outgoing media in the form of audio data or video data from the client application, or both; and
sending the captured media to an audio/video quality analyzer; and

operating the audio/video quality analyzer comprising separate media server software operated in tandem with WebRTC mechanisms employed by a contact center application running inside of the browser to perform the steps of:
receiving the captured media; and
determine an audio quality or a video quality of the captured media; and
sending the determined audio quality or video quality of the captured media to the external media server.

8. The method of claim 7 , wherein the external communication device is located at a cloud contact center, and a traversal using relays around network address translation server is used to relay requests from the cloud contact center to a computer operating a WebRTC engine.

9. The method of claim 7 , further comprising the steps of:
operating a software virtualized speaker on a computing device which is configured to operate in accordance from instructions contained in the redirection code;
including further instructions in the redirection code for the software virtualized speaker to perform the steps of:
taking priority of operation of audio output over other connected audio devices, if any, attached either to the computing device or to the external communication device; and
injecting the synthetic audio into the real-time transport protocol (RTP) media connection from the client application to the external communication device through the external media server, either as the incoming media or the outgoing media.

10. The method of claim 9 , further comprising the steps of:
operating a software virtualized microphone on a computing device which is configured to operate in accordance from instructions contained in the redirection code;
including further instructions in the redirection code for the software virtualized microphone to perform the steps of:
taking priority of operation of audio input over other connected audio devices, if any, attached either to the computing device or to the external communication device; and
capturing the synthetic audio generated by the software virtualized speaker.

11. The method of claim 7 , wherein the external media server handles both media processing and agent automation for the purposes of testing automated agent or client responses in the real-time transport protocol (RTP) media connection from the client application to the ex

Assignees

• Cyara Solutions Pty Ltd

Inventors

• Willshire Geoff

Classifications

• H04L67/141

  Setup of application sessions (admission control or resource allocation in data switching networks H04L47/70) · CPC title

• H04L43/50

  Testing arrangements · CPC title

• G06F9/547

  Remote procedure calls [RPC]; Web services · CPC title

• H04L65/65

  Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP] · CPC title

• H04L67/133

  Protocols for remote procedure calls [RPC] · CPC title