WebRTC vs SIP: Is There a Differences?
The most commonly used real-time communication protocols for IP-based video and audio communications these days are the session initiation protocol (SIP) and web real-time communications (WebRTC). These protocols have been extensively used in softphone and video conferencing applications.
SIP, is an application-layer control protocol developed by the Internet Engineering Task Force (IETF). SIP is the implemented standard for initiating, altering, and suspending an engaging user session which includes multimedia elements for instance voice, video, and instant messaging. SIP’s prime goal is communication among multimedia devices.
As we become familiar with a world of new acronyms, it is interesting how the term WebRTC and SIP (Session Initiation Protocol) are often used to denote the same core technologies. Many companies use both in tandem to devise the best communication solutions. Are both these technologies really different or they do have some commonality? Let’s find out.
SIP and WebRTC share a unique relationship with each other. Both are infrastructures developed to support real-time communication and collaboration over the Internet. The difference however lies in the way both operate and in their capability. Although both seem to be similar in appearance, it does not make them competitors but rather siblings.
WebRTC is not to be mistaken as a newer form of SIP. Instead, both WebRTC and SIP, are VoIP technologies. WebRTC expands on and integrates SIP functionality. That being said, the two infrastructures embrace a union where both complement each other.
In order to understand the difference and similarities between WebRTC and SIP, we first need to understand what these two mean individually and how they function.
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What is WebRTC?
In 2011, Google built an open-source solution known as Web Real-Time Communications (WebRTC). This solution enabled peer-to-peer communication in web browsers and mobile applications through the use of application programming interfaces (APIs) which comprise audio, video, and data transfers.
WebRTC simplifies communication by reducing barriers. This disruptive technology leverages plugin-free APIs and can be used in both desktop and mobile browsers. Today, WebRTC is supported by almost all major browser vendors. Before the birth of WebRTC, external plugins were needed to achieve the same functions.
“WebRTC empowers traditional web browsers to perform browser-to-browser communication such as voice calling, video chat, and data sharing without the need of any plugin.”
In the last decade, WebRTC has seen steady growth in popularity and adoption. By 2016 there were an estimated 2 billion browsers enabled to work with WebRTC. It has also logged over a billion minutes and 500 terabytes of data transmission per week from browser communications.
WebRTC’s main function is to allow access to devices. Users can access the microphone or camera on their phone or laptop, or even the entire screen. They can capture displays and even have that screen shared or recorded remotely.
WebRTC allows sending and receiving any type of data besides videos and calls. No wonder that WebRTC is becoming a popular choice for real-time communications. It is an open-source project, is free for commercial or private use, is available in all modern browsers, and since the source code is portable it can be used in mobile apps as well. WebRTC has created a vibrant ecosystem of different vendors and companies, which, in turn, has enabled the creation of new use cases and business models.