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WebRTC
WebRTC is designed to enable high-quality peer-to-peer connections on native mobile and browsers with simple APIs.
WebRTC is renowned for its NAT Traversal functionality. WebRTC is designed for peer-to-peer communication. It aims to find the most suitable direct route avoiding NAT gateways and firewalls for peer communication via a process named ICE. As part of this process, the WebRTC APIs find the public IP Address using STUN servers and fallback to a relay server (TURN) when direct communication cannot be established.
By setting the host server public network without firewall, the NAT Traversal will always be successful.
Video compression is an indispensable part of the pipeline that greatly contributes to a smooth streaming experience. Even though It is not compulsory to fully know all of VP8/H264’s video coding details, understanding its concepts helps to demystify streaming speed parameters, debug unexpected behavior, and tune the latency.
Video Compression for a streaming service is challenging because the algorithm needs to ensure the total encoding time + network transmission + decoding time is as small as possible. In addition, the encoding process needs to be in serial order and continuous. Some traditional encoding trade-offs are not applicable – like trading long encoding time for smaller file size and decoding time or compressing without order.
The idea of video compression is to omit non-essential bits of information while keeping an understandable level of fidelity for users. In addition to encoding individual static image frames, the algorithm made an inference for the current frame from previous and future frames, so only the difference is sent. As you see in the the Pacman example below, only the differential dots are transferred.
Similarly, the audio compression algorithm omits data that cannot be perceived by humans. The audio codec with the best performance currently is Opus. Opus is designed to transmit audio wave over an ordered datagram protocol such as RTP (Real-time Transport Protocol). It produces lower latency than (mp3, aac) with higher quality. The delay is usually around 5~66.5 ms
Pion is an open-source project that brings WebRTC to Golang. Rather than simply wrapping the native C++ WebRTC libraries, Pion is a native Golang implementation for better performance, better Golang integration, and version control on constitutive WebRTC protocols.
The library also provides sub-second latency streaming with many great built-ins. It has its own implementation of STUN, DTLS, SCTP, etc… and some experiments of QUIC and WebAssembly. This open-source library itself is really a good source of learning with a great document, network protocol implementations and cool examples.
The Pion community, led by a very passionate creator, is lively and has many quality discussions about WebRTC. If you are interested in this technology, please join http://pion.ly/slack – you will learn many new things.