With the 2020 end-of-life date for Flash rapidly approaching, we hear a lot of consternation from developers who are trying to get away from RTMP-based workflows. To them, it seems there are only three ways to create applications with RTMP:
- Use a player that is no longer natively supported in a browser or mobile device (Flash).
- Pay for an expensive custom player.
- Utilize a cache-based methodology (HTTP-based streaming) that increases overall latency.
Moving away from RTMP leaves many people desperately searching for low-latency alternatives that can either handle the origin signals, or deliver in a different protocol. WebRTC has been hailed as an “easy fix,” but it doesn’t scale well and can’t handle broadcast-quality streams. Secure Reliable Transport (SRT) adoption is growing, but workflows require new encoders, decoders and players to support it.
A ‘Flash Alternative’ Doesn’t Mean an RTMP Alternative
The trap that most developers fall prey to is thinking that the same transport protocol must be used throughout their workflow—so, whatever replacement they choose for RTMP in their workflow (for example, WebRTC) must be used from end to end, anyplace RTMP is found.
The fact is that people confuse needing a Flash replacement with needing an RTMP replacement. Simply put: They forget about transmuxing, which provides a simpler solution to keep existing RTMP workflows, while maintaining high quality and low-latency playback options.
What Is Transmuxing?
Transmuxing can also be referred to as repackaging, packetizing or rewrapping. Transmuxing is when you take compressed audio and video and—without changing the actual audio or video content—(re)package it into different delivery formats.
Think of it this way: Your stream is like a fire hose. You want to keep the stream flowing smoothly through the hose with consistent volume and speed. If you select a HTTP-based method, you’re shifting to a bucket brigade, sending your video chunk by chunk. However, if you were to simply connect a different hose of the same diameter, strength and quality, nothing changes except for the type of hose delivering it. That’s your transmuxing.
That’s why we made the Wowza Streaming Cloud service with Ultra Low Latency so flexible. The service will easily accept not only the WOWZ protocol from Wowza technologies, but will also ingest RTMP/RTMPS and RTSP, where it transmuxes it into WOWZ for playback across all devices.
How Can Wowza Help With Transmuxing?
First, when using our Ultra Low Latency Service, you don’t have to replace your existing RTMP infrastructure. This means you don’t have to pay for new encoders, license new SDKs or investigate “emerging” technologies for the sake of going fast. It’s a cheaper way to leverage the strength of RTMP, along with a robust player for web and mobile devices.
Secondly, transmuxing into the Wowza Streaming Cloud service gives you additional overhead to increase your video quality. Unlike WebRTC workflows that limit quality to 720p streaming, Wowza Streaming Cloud can handle 1080p or even 4K broadcast-quality resolutions. Simply send one stream up to Wowza Streaming Cloud and transmux for delivery—or even transcode and transrate for adaptive-bitrate streaming, if latency isn’t an issue.
Wowza provides robust live transmuxing capabilities to power your ultra low latency workflows. By using Wowza Streaming Cloud, you’ll get a powerful, fully managed service to transcode and deliver your content globally to audiences of any size.
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