Piggybacking Voice Quality Data on SIP Signaling
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Solution Overview
Problem
Current IP multimedia subsystem (IMS) technologies lack efficient methods to measure and report bi-directional voice quality during call termination or mid-call modifications without generating additional SIP signaling traffic, which is essential for providing quality assurance in IP-based telephony and multimedia services.
Innovation Solution
The proposed solution involves piggybacking voice quality data onto existing SIP signaling messages during call termination or mid-call modifications, allowing the network to process and generate quality reports without creating additional SIP traffic, by utilizing User Equipment (UE) to measure and accumulate voice quality metrics and insert them into SIP messages sent to the Call Session Control Function (CSCF), which then generates accounting records and bi-directional voice quality reports.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If voice quality data is measured and reported during call termination or mid-call modifications, then quality assurance capability is improved, but additional SIP signaling traffic is generated
Solution Approach 1:
The patent combines voice quality data reporting with existing SIP signaling messages (BYE, RE-INVITE, UPDATE) that are already transmitted during call termination or mid-call modifications. The UE inserts voice quality metrics into the body or headers of these existing messages, merging the quality reporting function with the call control signaling function, thereby avoiding additional traffic generation
Solution Approach 2:
The SIP signaling messages serve dual purposes: their primary function for call control (termination or modification) and an additional function for voice quality reporting. This multi-functionality allows the system to extract quality assurance capability from existing signaling infrastructure without creating separate dedicated reporting channels
2Measurement precision
If separate voice quality measurement mechanisms are implemented, then measurement precision is improved, but device complexity increases
Solution Approach 1:
The User Equipment (UE) performs self-measurement of voice quality metrics using its own resources and capabilities. The UE autonomously measures metrics such as MOS, packet loss, jitter, and delay, and self-generates the quality reports without requiring external measurement infrastructure, thereby maintaining measurement precision while minimizing added complexity
Data Source
AI summary
A call session control function (CSCF) receives, from a first user device engaged in a call with a second user device, a first Session Initiation Protocol (SIP) signaling message piggybacked with first data related to a voice quality of an inbound direction of the call at the first user device. The CSCF receives, from a second user device engaged in the call with the first user device, a second SIP signaling message piggybacked with second data related to a voice quality of an inbound direction of the call at the second user device. The CSCF extracts the first data related to the voice quality from the first SIP signaling message, and extracts the second data related to the voice quality from the second SIP signaling message. A charging collection function (CCF) determines a bi-directional voice quality of the call based on the first data and the second data.