Low Latency Streaming



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Low-latency streaming has been at the core of Wowza streaming architecture from the beginning. Wowza technology provides fine-grained controls to help you deliver the optimal low-latency experience for your customers.

How Wowza Powers Low-Latency Streaming for Your Use Case


Designing for Low Latency


Delivering low-latency streaming experiences involves balancing three main factors to find the right mix for your purposes.



Encoding protocols, viewing endpoint capabilities, software stack, and other workflow elements all introduce quantifiable latencies. However, an unmanaged network (such as the Internet) will introduce the greatest degree of uncertainty in delivery speed. A common method of reducing latency is to reduce buffer requirements throughout the controllable workflow, but doing so can also make network fluctuations much more visible to the viewer.



As audience size, geographic distribution, and the number of streams increase, network variability also increases, requiring more buffering and/or increased infrastructure complexity.



As the amount of data in your video, determined by resolution, effects, overlays, etc., increases, the corresponding encoding, transcoding, and transmuxing tasks either need more power and bandwidth to process and transmit the extra data in near real time, or will inherently increase latency.

Low-Latency Protocol Considerations

HTTP streaming protocols provide excellent scalability and can be tuned for low-latency delivery, but not ultra-low or real time. This makes them great for broadcasting live events with large viewing audiences using multiple types of endpoints, but not acceptable for video chat.
Stateful protocols such as RTMP and RTSP provide ultra-low-latency audio and video streaming for small to medium audiences, but broader reach requires scaling out linearly by deploying additional media servers in a tiered or edge network configuration. RTMP and RTSP are not natively supported on all endpoints (e.g., iOS devices), and are commonly restricted by firewalls for security reasons.
WebRTC is designed for ultra-low-latency delivery of audio, video, and data. WebRTC chat applications are easily deployable without plug-ins thanks to native stream encoding and playback using compatible HTML5-based browsers. It's perfectly suited to one-to-one or small-group live chat.

Key Features of Low-Latency Streaming with Wowza

Mobile Encoding App & SDK





 Low-latency WOWZ protocol for stream ingest

 Optimized bitrate encoding to suit available network conditions

Cloud Service





 WOWZ, RTMP, and RTSP low-latency protocols for incoming streams




 Cloud platform delivering in-region transcoding resources worldwide

 Workflows available to achieve lower latency (Contact us for more information.)




 Lower-latency delivery options available (Contact us for more information.)

Server Software





 WOWZ, RTMP, and RTSP low-latency protocols for incoming streams

 WebRTC low-latency browser-encoded incoming streams (Read more about Wowza WebRTC software preview)




 Scalable, extensible architecture for cloud deployment and transcoding-workflow optimization

 RTMP, RTSP, and WebRTC delivery for low and ultra-low latency

 Adjustable chunk sizes and key frames for HTTP (Apple HLS) workflows to achieve lower latency at scale




 WebRTC implementation for live stream and bandwidth optimization

 Support for delivery via stateful protocols and chunk-size-reduced HTTP-based protocols 

Need Help?

Contact a Wowza technical specialist to find the right workflow for you.