Packetization in WebRTC is the process used to take audio and video frames and prepare them for sending over the network.
A media frame can be considerably smaller or larger than the MTU size which means we will either be underutilizing the network or fragmenting these frames over multiple network packets.
To properly receive such frames, construct them back and play them, RTP packetizes the media frames, splitting them into multiple packets in a way that makes dealing with packet loss, reordering and other network artifacts manageable.
Packetization is especially important in video codecs, where the frames are usually larger than a single network packet size. Different video codecs have different payload headers to them which indicate how to packetize and depacketize the codec in a specific manner, based on the capabilities and characteristics of the codec.
