Low-Latency WebRTC Streaming via Intermediary Server Transcoding
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Solution Overview
Problem
Existing live media stream content delivery systems suffer from significant latency, which hampers interactive live broadcasting applications where viewer engagement and real-time interaction are crucial, as significant delays between viewer reactions and performer responses become noticeable and disrupt the coordination between participants.
Innovation Solution
A low-latency content delivery network system utilizing intermediary servers that receive WebRTC streams, negotiate quality metrics, and transcode media data to adapt to downstream server capabilities, enabling efficient distribution of live media streams across various devices and network conditions, including the use of WebRTC edge servers for localized transcoding and adaptive bitrates.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of time
If traditional content delivery networks are used for live media streaming, then content can be delivered to various devices and network conditions, but significant latency is introduced that hampers interactive live broadcasting
Solution Approach 1:
The system segments the content delivery network into multiple specialized server types (ingest servers, processing servers, edge servers, origin servers) that handle different aspects of stream processing. This segmentation allows low-latency WebRTC streams to be handled by edge servers closer to viewers while maintaining compatibility with various devices through coordinated processing across the segmented network architecture.
Solution Approach 2:
The patent introduces intermediary servers that act as mediators between the low-latency WebRTC stream source and various downstream devices. These servers negotiate quality metrics and perform adaptive transcoding to bridge the gap between the low-latency stream format and diverse device requirements, reducing overall latency while maintaining device compatibility.
2Adaptability or versatility
If WebRTC streams are transcoded to adapt to downstream server capabilities, then device compatibility is improved, but processing time and complexity increase
Solution Approach 1:
The system applies local quality adaptation by negotiating specific quality metrics with each downstream server based on its capabilities and the viewing conditions. Instead of universal transcoding, each edge server adapts the stream locally to match its specific downstream requirements, reducing overall processing complexity while maintaining device compatibility.
Solution Approach 2:
Quality metric negotiation occurs in advance before the actual streaming begins. The system pre-negotiates compatible quality parameters between upstream and downstream servers, allowing the WebRTC stream to be transmitted in an optimal format without requiring complex real-time transcoding during live broadcasting.
3Manufacturing precision
If quality metrics are negotiated with downstream servers, then stream quality is optimized for specific devices, but communication overhead and setup time increase
Solution Approach 1:
The system implements partial quality optimization by negotiating only the essential quality metrics needed for each downstream server rather than comprehensive optimization. This selective approach achieves sufficient stream quality for each device type while minimizing the time required for quality parameter negotiation and setup.
Data Source
AI summary
Methods, systems and arrangements are described herein for providing low-latency content delivery networks usable for widespread broadcasting of live streaming media, such as live interactive streaming media, over IP networks. Also described are methods, systems and arrangements for enabling low-latency live streaming media to be made available to a wide range of different viewer devices having various capabilities, and under various network conditions. Also provided are methods, systems and arrangements for low-latency creation of the live streaming media broadcasts.