HLS or MPEGTS

MPEGTS (=ts):
This one has two parts which are MPEG and TS. MPEG stands for Moving Picture Experts Group which was a company founded in 1988 specialized in video streaming stuff and later joined with Adobe. TS, on the other hand, stands for Transport Stream.
This format can provide audio, video, and metadata like subtitle, epg or a different form of data to lock the stream. Mpeg has an error correction feature to keep the integrity of the video whenever the signal is low.

MPEGTS (=ts) is a standard digital container format for transmission and storage of audio, video, and Program and System Information Protocol (PSIP) data. It is used in broadcast systems such as DVB, ATSC and IPTV.

MPEGTS (=ts) is the older, widely supported format which is not aware of the network conditions.
That means it sends video chunks at a standard rate no matter the speed/condition of the internet.

HLS (=m3u8):
It stands for HTTP Live Streaming. In this structure, the broadcasting server divides the stream into separate 10 seconds sections of mp4. This is great when the broadcaster requires to stream multiple streaming qualities so that the spectator could use the fitting bitrate of the video considering the network bandwidth.
HLS was formerly invented and used by Apple on their devices. The only downside is that HLS has 10 to 30 seconds of lag so if live streaming does really matter, it is not suggested.

HLS (=m3u8) is an HTTP-based adaptive bitrate streaming communications protocol implemented by Apple Inc. as part of its QuickTime, Safari, OS X, and iOS software. Client implementations are also available in Microsoft Edge, Firefox and some versions of Google Chrome. Support is widespread in streaming media servers.

HLS (=m3u8) on the other hand is the newer standard, developed by Apple. Not yet 100% supported by all programs/devices. This standard is aware of the network speed/congestion. That means it can sense and switch the resolution of the video “on the fly“, depending on the network conditions.

RTMP (=rtmp):
Real-Time Messaging Protocol specification. The Real-Time Messaging Protocol (RTMP) was designed for high-performance transmission of audio, video, and data between Adobe Flash Platform technologies, including Adobe Flash Player and Adobe AIR.

RTMP (=rtmp) is used to stream audio, video or data and is originally a proprietary protocol introduced by Macromedia (owned by Adobe). The protocol is TCP-based and offers therefore persistent connections. In short, RTMP encapsulates MP3/AAC audio and MP4/FLV video multimedia streams.

RTMP (=rtmp) If you do live streaming with KeyCDN, RTMP will not be used for video delivery to end users. If the streaming software is based on RTMP, you can use this protocol to initially stream your content to your server or a third party service that transforms the RTMP stream to HLS. A wide range of RTMP encoders can be used to create the live stream (e.g. FMLE, OBS, or WireCast).

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First option should be MPEGTS (=ts) ;)

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