We are building a voice and video solution using WebRTC - mostly to work on Chromium-based browsers such as Google Chrome and Microsoft Edge.
On the Chrome developer console - network tab, there is a feature to see how your app performs under various simulated network conditions such as fast-3G, slow-3G, or altogether going offline.
Unfortunately, the network simulations do not work on WebRTC apps. I am assuming this is due to WebRTC using UDP protocol, and the network throttle is built for TCP.
Can you suggest ways to simulate varying network conditions? We want to test how the video app performs on varying bandwidth and latency conditions. Are there any plugins that would simulate this? It would be good to have the Google Chrome team extend their network throttling feature to include voice, video, and data streams that primarily run on UDP.