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This article will help to explain a little about the most common used streaming protocols used today. 

RTMP (Real-Time Messaging Protocol): TCP

Developed by Macromedia in 2002, this protocol is still the default used for live streaming throughout the internet.

This streaming protocol maintains a persistent, stable connection which allows for low-latency communication and is supported by most streaming platforms and software.
It supports the video codec H.264 and the audio codec AAC, but has a relatively high level of latency.

WebRTC (Web Real-Time Communication protocol): UDP and TCP

WebRTC is an open-source standard for real-time communications and supports VP8, VP9, and H.264 video codecs, as well as the Opus audio codec. Because WebRTC is browser based, it allows any modern web browser to become a streaming terminal and has a very low latency and adaptable bitrates to help prevent drops in connections. However the low latency of this protocol can result in unstable connections.

SRT (Secure Reliable Transport): UDP

SRT is an open-source video streaming protocol developed by Haivision and Wowza and is regarded to be the substitute for RTMP at some point in the future as it holds the promise of a stable sub-second latency live stream over variable networks. SRT is codec-agnostic meaning it can support any modern video and audio codec, but its relative youth means it is not yet widely adopted across as many platforms as RTMP.

HTTP LIVE STREAMING (HLS)

HLS streaming protocol is a protocol developed by Apple that has become the mostb widely used streaming protocol on the internet.

HLS is an evolving and versatile protocol that uses adaptive bitrates and can server high quality video from HTTP servers