RTN is a stable, high quality early transmission network for ultra-low latency multi-person audio and video communication scenarios. Even with the decentralized distributed architecture design, RTN allows end users to access from the edge nodes in close proximity, using intelligent routing algorithms to calculate the optimal path transmission in real time, achieving global end-to-end millisecond low latency, effectively solving the problem of routing links and bandwidth costs.
It can realize end-to-end millisecond, cross-regional ultra-low latency interaction to meet the demand of real-time audio and video interaction scenarios.
With the packet loss recovery mechanism and jitter countermeasure mechanism in weak network environment can be self-adaptive code flow, to ensure the stability and smoothness of terminal use.
The global multi-room server clustering and distributed architecture enables smooth scaling at the second level, with theoretically no concurrency cap, and can withstand the ultra-high concurrency pressure of user influx.
Based on intelligent algorithms to automatically plan the optimal line and can monitor the state of the line in real time dynamically, switch in time when abnormal, dynamically adjust the buffer time at the receiving end to ensure high quality audio and video transmission.
Provide multi-system and WeChat applet live streaming SDK, support open broadcast, watch, continuous mike, interaction, window style, etc.
For playing and downloading on-demand, playback, replay, recording, playback of chat history, Q&A history, documents (chapters), variable speed, etc.
Provide multi-system and WeChat applet live streaming SDK, support open broadcast, watch, continuous mike, interaction, window style, etc.
