SIP Phone Silence Detection for VoIP Call Termination
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Solution Overview
Problem
In VoIP systems, the inability of PSTN gateways to generate a BYE message can prevent proper termination of VoIP calls, leading to wasted network resources and memory usage when attempting to leave voice messages.
Innovation Solution
A method and apparatus for a SIP phone to receive and analyze codec frames from a PSTN gateway, incrementing a counter when a specified portion of the frames indicates silence, and terminating the call when the counter reaches a threshold, thereby facilitating call termination even in the absence of a BYE message.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If a PSTN gateway is used to connect PSTN phones with SIP phones, then interoperability between PSTN and VoIP networks is achieved, but the gateway may fail to generate a BYE message to terminate the VoIP call properly
Solution Approach 1:
The SIP phone performs self-service by monitoring its own received audio frames and autonomously determining when to terminate the call based on silence detection, without requiring the PSTN gateway to send termination signals.
Solution Approach 2:
The system uses feedback from the audio stream itself (monitoring for silence periods) to trigger call termination, creating a closed-loop control mechanism that adapts to actual call conditions rather than relying on external termination signals.
2Ease of operation
If the SIP phone waits for a BYE message from the PSTN gateway to terminate the call, then proper protocol termination is maintained, but network resources are wasted and memory is consumed during prolonged waits
Solution Approach 1:
The SIP phone performs preliminary action by proactively monitoring audio frames for silence periods and initiating termination based on detected silence, rather than passively waiting for the gateway's BYE message to arrive.
Solution Approach 2:
The system dynamically adjusts the termination behavior based on real-time audio conditions, transitioning from static protocol waiting to dynamic silence-based termination decisions that adapt to actual call state.
3Measurement precision
If the SIP phone continuously monitors audio frames for silence, then accurate call termination timing is achieved, but processing overhead and computational resources are increased
Solution Approach 1:
The system applies partial action by monitoring only specific portions of the audio stream (looking for silence periods rather than analyzing every audio parameter), achieving sufficient precision for termination without excessive computational complexity.
Data Source
AI summary
The invention relates to the field of voice over Internet protocol (VoIP) and more specifically to a system and method of terminating a VoIP call. In certain exemplary embodiments, the present solution provides in a voice over Internet protocol (VoIP) network, a method of terminating a call received from a public switched telephone network (PSTN) phone by a session initiation protocol (SIP) phone via a PSTN gateway, in the absence of a terminating signal from the PSTN gateway, the method comprising: (a) receiving at said SIP phone a sequence of codec frames from said PSTN gateway; (b) determining if a specified portion of each codec frame in said received sequence of codec frames is a specified value; (c) if said specified portion is said specified value, incrementing a counter by an incrementing amount; and (d) when said counter reaches a specified threshold, terminating said call In the absence of a BYE message, session termination of a VoIP call can still be facilitated using the method and apparatus of the present invention.