Views: 0 Author: Site Editor Publish Time: 2026-11-02 Origin: Site
A picture that arrives two seconds late is a picture you cannot have a natural conversation around. In interactive scenarios — from remote education and collaboration to medical training — latency can be the difference between a smooth exchange and a frustrating experience.
Many conventional streaming workflows use segmented delivery protocols. Video is divided into chunks, packaged, buffered, and delivered to the playback device. This approach is reliable and highly scalable, making it well suited to broadcast distribution and video-on-demand applications.
However, segmentation and buffering introduce additional delay. The receiving device may need to wait for video segments to be generated and buffered before playback begins. Depending on the protocol, configuration, and network environment, this can result in several seconds of latency — acceptable for many broadcast applications, but less suitable for real-time interaction.
For years, professional AV deployments often had to balance low latency against compatibility and ease of access. Achieving a responsive viewing experience could require dedicated software, specialized players, or point-to-point hardware.
Consider a medical teleteaching session. An experienced surgeon is demonstrating a procedure to a group of remote medical students. A live video feed from the operating room is delivered to their screens.
The instructor points to a structure and explains what the students should observe. If the video feed is several seconds behind, the visual information no longer matches the instructor's explanation in real time. Questions and responses become harder to coordinate, and the teaching experience loses its natural flow.
A similar problem occurs in online classrooms. A teacher asks a question and waits for a student to respond. When the video and audio are delayed, students may speak over one another, reactions arrive late, and the interaction feels less natural.
WebRTC (Web Real-Time Communication) is an open technology standardized through the W3C and IETF. It is designed to enable real-time audio, video, and data communication between compatible endpoints, including mainstream web browsers.
Under suitable network and configuration conditions, WebRTC can support end-to-end latency of approximately 300 milliseconds. Actual performance depends on the encoder, network conditions, browser, and playback device.
The practical result is a more accessible way to deliver interactive video. A compatible browser can become a viewing endpoint for a camera feed, a video switcher output, or another professional video source — without requiring a dedicated playback application.
When network conditions allow, WebRTC can establish a direct peer-to-peer media connection between endpoints without relying on an external cloud media relay. In other environments, STUN or TURN services may be required to establish connectivity. This distinction is particularly relevant for organizations with network security, data-residency, or infrastructure requirements.
Importantly, WebRTC does not have to replace established streaming protocols. Different protocols serve different purposes within a professional AV workflow:
RTMP / RTMPS: Commonly used for live streaming to online platforms and streaming servers.
SRT: Designed for reliable video transport over networks where packet loss and fluctuating conditions are concerns.
RTSP / RTP: Frequently used in IP camera systems and professional LAN-based video workflows.
UDP Unicast / Multicast: Useful for distributing video streams across IP networks, depending on network design.
HLS: Suitable for scalable delivery to browsers and mobile devices, although it typically introduces more latency than real-time communication workflows.
WebRTC: Particularly useful when viewers need to watch and respond to live video with minimal delay.
To support applications that require more responsive video delivery, ORIVISION is bringing WebRTC support to two professional video encoders: the EH901 HDMI Video Encoder and the ES901 SDI Video Encoder.
The two models are designed for different video input environments. The EH901 accepts HDMI input, while the ES901 is designed for SDI-based workflows. Both are intended to integrate professional video sources into IP-based distribution systems.
With WebRTC support, these encoders can help users access live video through compatible web browsers, making them suitable for applications such as remote teaching, live demonstrations, monitoring, and interactive video communication.
Depending on the model and configuration, key capabilities include H.265, H.264, and MJPEG encoding; multi-protocol streaming; flexible bitrate and output settings; audio processing; on-screen information; web-based management; and integration features. Please refer to each model's product page for confirmed specifications and available options.
Resolution helps define image quality, but latency determines how effectively people can interact with live video.
As professional AV continues to expand into interactive applications — including remote education, live instruction, remote collaboration, and medical training — low-latency delivery is becoming an increasingly important part of system design.
WebRTC provides a browser-friendly approach to real-time video communication. Combined with professional hardware encoding and support for established IP streaming protocols, it gives system integrators more flexibility when designing video distribution workflows.
Tell us about your video source, network environment, and latency requirements. The ORIVISION team can help you evaluate an encoding solution for your application.
