Bridged Audio Conferencing With Selective SIP-TDM Stream Control
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Solution Overview
Problem
Current systems face challenges in managing mixed audio streams and maintaining reliable connections across SIP and TDM communications in conference environments, particularly in scenarios involving failed or dropped calls, and lack effective mechanisms for seamless interoperability and service continuity.
Innovation Solution
The system provides dynamic audio stream management, continuous connection attempts, and automatic reconnection mechanisms to enhance call reliability, while supporting high-availability operations with seamless integration of mixed-protocol communications and service continuity through failover to secondary nodes.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If mixed audio streams from SIP and TDM communications are managed simultaneously, then interoperability between different communication protocols is improved, but system complexity increases
Solution Approach 1:
The patent employs an audio bridge system as an intermediary component that sits between SIP and TDM communication protocols. This mediator translates and manages audio streams between the two different protocols, enabling interoperability without requiring direct integration between SIP and TDM systems. The audio bridge handles protocol conversion, codec matching, and stream management centrally, reducing overall system complexity while maintaining adaptability across multiple communication standards.
Solution Approach 2:
The audio bridge system is designed with multi-functional capabilities to handle both SIP and TDM communications simultaneously. It can manage multiple audio streams, support various codecs (G.711, G.722, OPUS), and provide conference bridging services across different protocol types. This universal design allows a single system to perform multiple functions rather than requiring separate dedicated systems for each protocol, thereby improving interoperability without proportionally increasing complexity.
2Reliability
If continuous connection attempts and automatic reconnection mechanisms are implemented, then call reliability is improved, but use of energy increases
Solution Approach 1:
The system implements periodic connection attempts with configurable intervals rather than continuous retry loops. The auto-connect mechanism schedules reconnection attempts at defined time intervals (e.g., every 30 seconds, 1 minute, or 5 minutes), allowing the system to balance reliability with resource consumption. This periodic approach ensures calls are retried after failures while avoiding excessive energy usage from constant reconnection attempts.
Solution Approach 2:
The system performs preliminary actions by pre-configuring reconnection parameters and establishing connection pools before failures occur. It maintains pre-established SIP and TDM connection resources that can be quickly reactivated rather than creating connections from scratch during reconnection attempts. This preliminary preparation reduces the energy and computational resources needed during actual reconnection events while maintaining high call reliability.
3Reliability
If failover mechanisms are implemented for high-availability environments, then service continuity is improved, but device complexity increases
Solution Approach 1:
The system implements failover mechanisms with pre-configured backup nodes and redundant communication paths. Before primary nodes fail, secondary nodes are already prepared and registered in the system, ready to immediately assume service responsibilities. This beforehand cushioning ensures service continuity without requiring complex real-time decision-making or dynamic reconfiguration during failures, thereby maintaining reliability while controlling complexity through advance preparation.
Solution Approach 2:
The failover system utilizes parameter changes in node status and connection routing to manage high availability. When a primary node fails, the system changes operational parameters by switching active connections to secondary nodes and updating routing tables. This parameter-based approach to failover is more efficient than structural reconfiguration, as it maintains the same physical infrastructure while dynamically adjusting operational parameters to ensure service continuity with minimal complexity increase.
Data Source
AI summary
Systems and methods are disclosed for bridging SIP calls in conference environments that include both SIP and TDM legs. A request to bridge a SIP call from a specific conference number is received, and the system configures caller and called IDs for the bridged call. Audio is selectively sourced from all primary SIP legs excluding dynamic legs, all sequence SIP legs excluding dynamic legs, or a mixed audio stream including dynamic legs. The bridged call is established using either a default SIP trunk configured in a node or a unique SIP trunk if no default is set. The system supports mixed-protocol integration, continuous connection attempts, automatic reconnection, RTP/RTCP timeout protection, and high availability with node failover. Additional features include codec configuration via SDP, support for multiple simultaneous bridged calls, on-demand call drop, and a user portal for managing bridging services. Bridging does not disrupt ongoing conferences or original configurations.


