Substream’s technology is designed to deliver high-quality, real-time audio and video communication across a broad spectrum of applications. From video calls and large-scale webinars to interactive broadcasts and virtual environments, our infrastructure supports seamless and scalable media experiences.
The concept of real-time communication over the internet has evolved from early experiments in network resilience to today’s sophisticated global networks. Initially, the internet's architecture, which emerged in the 1960s, was designed to be fault-tolerant and flexible, contrasting sharply with the rigid, circuit-switched networks of the time. This packet-switched model has laid the foundation for all subsequent internet communications, including real-time audio and video.
WebRTC: Web Real-Time Communication (WebRTC) is a pivotal standard that enables browser-based real-time audio and video communications without the need for additional plugins or software. Substream leverages WebRTC due to its wide adoption in modern web browsers and its support for real-time media capabilities.
WebRTC facilitates direct peer-to-peer connections, optimizing for low latency and enabling potentially end-to-end encrypted communication. However, for scenarios involving more than two participants or requiring enhanced quality control, media traffic is typically routed through media servers.
Media Servers: Substream’s infrastructure uses media servers to manage and route the audio and video traffic efficiently. These servers are not only crucial for handling large-scale communications but also enhance the quality and reliability of connections, especially across long distances.
Our media servers operate using both TCP and UDP protocols. While TCP ensures reliable packet delivery, UDP is preferred in real-time communications for its lower latency and overhead, despite its lack of built-in delivery guarantees.
Delivering real-time audio and video over the internet presents unique challenges such as latency, packet loss, and jitter. Substream’s network architecture is optimized to minimize latency through strategic server placement and advanced routing protocols.
Packet loss can degrade audio and video quality, and Substream uses techniques like packet retransmission, error correction, and adaptive bitrate adjustments to mitigate this. Jitter, or the variation in packet arrival times, is managed with jitter buffers and other strategies to ensure a consistent quality of experience.
The landscape of real-time communication is continually advancing, with ongoing research into more efficient codecs, better network management techniques, and more robust security measures. Substream is committed to staying at the forefront of these developments, continuously enhancing our infrastructure to support the next generation of real-time communication applications.
By leveraging a deep understanding of network and media technology as well as the practical challenges of delivering high-quality real-time audio and video, Substream aims to provide a platform that not only meets the current demands but also anticipates future needs. This commitment ensures that our customers can build and scale their applications confidently, knowing they are supported by a state-of-the-art real-time communication service.