SDWAN Audio Aggregation for VoIP Quality
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Solution Overview
Problem
Conventional VoIP systems over LTE connections face challenges in maintaining high-quality voice calls due to packet delays, jitter, and congestion, especially when a large number of calls are active, leading to degraded voice quality and instability, even with prioritization techniques like MPLS.
Innovation Solution
A system that digitizes incoming PSTN-compatible or High Definition Voice audio signals, segments them, compresses according to preprogrammed rules, aggregates samples, and transmits them over a SDWAN using optimized data link protocols, with decompression and jitter buffering at the receiving end to regenerate high-quality audio signals.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If conventional VoIP systems transmit a large number of voice packets over LTE/SDWAN networks, then voice call capacity increases, but packet delays, jitter, and congestion occur leading to degraded voice quality
Solution Approach 1:
The patent combines multiple voice packets into single aggregated packets for transmission over the network. Instead of sending individual voice packets separately, the system bundles them together, reducing the total number of packets transmitted while maintaining all voice communication channels open. This aggregation reduces network congestion and improves voice quality stability while preserving high call capacity.
2Reliability
If voice packets are prioritized using MPLS, then voice quality is improved, but network complexity and device requirements increase
Solution Approach 1:
The patent introduces an intermediary aggregation function that sits between the voice packet sources and the network transport layer. This intermediary component bundles voice packets before transmission, acting as a mediator that simplifies network requirements. Instead of requiring complex MPLS prioritization infrastructure, the system uses this intermediary aggregation layer to ensure reliable voice quality transmission over standard networks.
3Speed
If individual voice packets are transmitted separately, then real-time voice communication is maintained, but network congestion and packet loss increase under heavy load
Solution Approach 1:
The patent segments the voice communication stream into aggregated packet groups rather than individual packets. Each aggregation contains multiple voice packets that are transmitted together as a unit. This segmentation approach maintains real-time communication by keeping aggregation intervals short while improving reliability by reducing the total number of packet transmission events, thereby minimizing network congestion and packet loss.
Data Source
AI summary
An automated telecommunications system includes a first system that receives audio frequency signals such as PSTN-compatible, voice over IP, or high definition voice, decodes and interprets said incoming signals according to the type and format, and transmits digital messages to a second system over a software defined wide area network (SDWAN). Said second system receives and interprets digital messages incoming from the first system, encodes and regenerates outgoing audio frequency signals. The system may be bi-directional and operate over a software defined wide area network, such as an IP based wireless or wired data network. The functionality of said first and second systems may be combined at a single location and interface with a conventional PSTN compatible VoIP system or High Definition Voice system at either or both ends and allow PSTN, conventional VoIP and High Definition Voice communications to share the same packet data stream over a SDWAN connection.