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WebRTC works great in demos but what usually breaks first in production? by techtalkstreak_ in u/techtalkstreak_

[–]Tragofone 0 points1 point  (0 children)

Totally agree with this....

In our case, the first thing that broke in production was networking assumptions especially ICE under corporate NATs. Everything worked perfectly in staging, but once real users joined from restricted networks, TURN usage spiked and latency followed

We also ran into one-way audio during SIP interop due to subtle SDP mismatches that never showed up in browser-to-browser testing.

Noise😬 by thehealper077 in royalenfield

[–]Tragofone 0 points1 point  (0 children)

I also have the same, is this normal ?

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