WEBRTC: No paced sending causes packets drop.

[jianjun0606]

When playing WebRTC, the server-side sending is not smoothed, which may cause packet loss.

I am analyzing the UDP/RTP packets sent by the server to the client when playing WebRTC, and I found that the I-frame is sent within almost 1ms, which has a high probability of causing UDP packet loss. It is recommended to do a smoothing process when sending.

The attachment is my packet capture screenshot. You can see that the I-frame is sent within about 0.3ms, without smooth sending.

[winlinvip]

Why would it cause packet loss? If the UDP buffer is large enough, it shouldn't be a problem.