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1. If latency is your main concern, it could be optimized by the following. Note that each of these are a tradeoff between latency and quality. a. lowering the maximum latency of the jitter buffer (currently 2s) b. tuning the x264enc parameters c. Using a lower resolution, framerate, and bitrate 2. A/V data is transmitted using WebRTC with our go sdk

Thank you for your help. We need to connect livekits from two domains for real-time video conferencing. These two domains are isolated, with another streaming system in between that can only rely on RTMP for transmission. Our current idea is for both livekits to use egress for streaming and ingress for attracting traffic. We lack relevant experience and do not know if this is correct. We also want to know the minimum latency that can be achieved in such a scenario and how to achieve the optimal configuration for real-time transmission.

I can’t tell you offhand what the minimum latency will be but I would expect it to be a few seconds. If it’s possible to skip or replace the rtmp system, it may be worth exploring the option to build an adapter between the two domains using webrtc