By choosing the right protocol to broadcast your live stream you will be able to:

- Optimize latency according to your event type and needs
- Match with the broadcast standards of each destination points.

To find out everything you need to know about broadcasting your live stream with the protocol that suits you best, scroll down the page!

MOST-USED STREAMING PROTOCOLS

LRT

LiveU Reliable Transport, or LRT™, is the video and audio delivery protocol developed over the last decade by LiveU, the inventor and patent holder of cellular bonding. The LRT™ protocol supports low latency, high resiliency video and audio transmission and was built from the ground up to accommodate the special properties of 4G/5G cellular as well as more traditional wireless and wired networks. In particular, LRT™ is optimized to support the bonding of multiple IP connections.

RTMP (Real-Time Messaging Protocol): TCP

Developed by Macromedia and acquired by Adobe in 2005, RTMP remains the most-used protocol. It maintains a persistent, stable connection and allows for low-latency communication. RTMP transfers stream data that is split into small packets between Flash Player and a server to ensure minimal interruptions and artifacts. Also, because of the legacy factor, RTMP is supported by most streaming platforms and software.

• Video codecs: H.264
• Audio codecs: AAC
• Latency: 3 - 30 seconds

➕ Pros: Multicast support, low buffering, wide platform support

➖ Cons: Old codecs, somewhat low security, relatively high latency

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

WebRTC is an open-source standard for real-time communications. WebRTC supports high-quality VP8 and VP9 (besides the old H.264), as well as the Opus audio codec. 

One of the biggest advantages of WebRTC is that it transforms millions of browsers into streaming terminals without any additional plugins needing to be installed. Moreover, WebRTC supports sub-second latency, which means no more delays! Lastly, the protocol uses an adaptable bit rate technology, which allows it to automatically adjust video quality and prevent any drops and interruptions.

• Video codecs: VP8, VP9, H.264 (H.265+ AV1 in progress)
• Audio codecs: Opus
• Latency: Less than one second

➕ Pros: No plugins needed, sub-second latency, supported codecs

➖ Cons: Instability due to sub-second latency

SRT (Secure Reliable Transport): UDP

SRT is an open-source video streaming protocol developed by Haivision and Wowza. It is widely considered to be a substitute for RTMP soon. Sharing the same advantages, SRT is taking the next step and making the dream of stable live streams with a sub-second latency a reality. It allows you to live-stream your content over suboptimal networks.

The developers state that SRT protects your live videos from jitters, bandwidth fluctuation, and packet loss. Moreover, SRT is similar to FTL and WebRTC in terms of sub-second latency, which allows for nearly real-time communication. Additionally, it is also stated that the protocol is codec-agnostic, meaning it supports any modern video and audio codec. Unfortunately, considering it is still an emerging technology, SRT is not widely supported.

• Latency: Less than one second

➕ Pros: High quality, stability, sub-second latency, strong codec support

➖ Cons: Weak platform support

APPLE HTTP LIVE STREAMING (HLS)

HLS streaming protocol is an alternative protocol developed by Apple. Today, HLS is the most widely used streaming protocol on the internet.

HLS is an adaptive bitrate protocol and also uses HTTP servers. This protocol is an evolving specification, as Apple adds features regularly.

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