Dynamic PBX Identifier Routing for WebRTC-SIP Call Integration
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Solution Overview
Problem
Current systems that employ the WebRTC protocol do not integrate with existing telephone networks such as private branch exchange (PBX) systems that utilize session initiation protocol (SIP) trunks and/or the public switched telephone network (PSTN), and assigning unique telephone network identifiers to each user can be costly and complicated to manage.
Innovation Solution
A communication server system that dynamically links active client accounts to reservable PBX system identifiers based on call event details, routing call data between client devices and user devices using WebRTC and SIP protocols, and managing SIP trunks to facilitate integration with PBX systems.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If unique telephone network identifiers are assigned to each user, then integration with PBX systems is achieved, but cost and infrastructure complexity increase
Solution Approach 1:
The patent implements dynamic identifier assignment where PBX system identifiers are allocated on-demand based on call events rather than being statically assigned to each user. The communication server dynamically links client accounts to available PBX identifiers during call events, allowing the system to adapt to varying staffing levels and turnover without managing permanent unique identifiers for each user.
Solution Approach 2:
The communication server acts as an intermediary between WebRTC client devices and PBX systems. It manages the mapping between client accounts and PBX identifiers, handling the complexity of identifier allocation, assignment, and routing. This mediator approach allows integration with PBX systems without requiring each user to have a dedicated permanent identifier.
2Device complexity
If dynamic linking of client accounts to PBX identifiers is implemented, then cost and complexity are reduced, but protocol conversion requirements increase
Solution Approach 1:
The communication server serves as a protocol conversion intermediary, translating between WebRTC protocols used by client devices and SIP protocols required by PBX systems. This mediator handles the protocol translation automatically during call events, enabling integration without requiring manual configuration or complex custom integration code at each endpoint.
Solution Approach 2:
The communication server provides universal compatibility by supporting both WebRTC and SIP protocols simultaneously. It can handle various call event types (incoming calls, outgoing calls, conferencing) and dynamically adapt to different scenarios, making the system universally applicable across different device types and network configurations without requiring separate specialized solutions.
Data Source
AI summary
A communication server system may one or more processors and one or more non-transitory, computer-readable media storing instructions that, when executed by the one or more processors, cause the communication server system to detect a call event relating to a telephone network identifier. The system may dynamically link one or more active client accounts to a reservable private branch exchange (PBX) system identifier based on first details of the call event and second details of the one or more active client accounts. The PBX system may employ a session initiation protocol (SIP). The system may route call data between one or more devices associated with the active client accounts dynamically linked to the reservable PBX system identifier and a user device associated with the telephone network identifier.