[sumy23]

1) WebRTC and RTP both have RT in their name. RT stands for real-time. If I recall correctly, the only buffer WebRTC has is the jitter buffer, which is used for packet ordering, not for ensuring that enough has buffered to handle bitrate spikes.

2) Yes, you either need a high keyframe interval or some type of out-of-band signaling framework to generate keyframes. WebRTC uses RTCP. A good question is why does WebRTC feel RTCP is necessary at all? Why not generate a keyframe every N seconds like you do with HLS and remove the complexity of RTCP entirely? The answer is that many clients cannot handle the bitrate at real-time speeds.

[saurik]

1) That is a specific implementation, and has nothing to do with the protocol, which certainly doesn't define a "jitter buffer". People routinely use RTMP--which also has RT in the name--to transfer content to streaming services with massive buffers at every step in the pipeline.