Comparing WebRTC and SIP is comparing apples to oranges. WebRTC is a media engine with an application program interface...
(API) on top of it that happens to be built into a Web browser, while session initiation protocol (SIP) is a signaling protocol that happens to need the use of a media engine.
In some ways, WebRTC and SIP complement each other -- like when you are trying to reach an SIP-based service from a Web browser. But in other ways, WebRTC can end up competing indirectly with SIP. How can that be?
As a free technology embedded in browsers, WebRTC technology reduces the barrier of entry to those needing to develop VoIP-related services, while at the same time opening up this rather "arcane" knowledge of VoIP to Web developers.
Many Web developers will end up using other signaling protocols such as XMPP, MQTT or a proprietary one instead of SIP. The reasons for that vary:
- They have no experience with SIP.
- They don't need SIP for their specific use case.
- They already have other means of signaling for their use case, such as messaging inside a dating service.
In these cases, WebRTC ends up replacing SIP indirectly.
But in other cases, vendors who use WebRTC technology will adopt SIP either because they want to connect it to legacy telephony services -- and as of now SIP is the best way to do that -- or because they like SIP's architecture and capabilities.
To sum things up, WebRTC and SIP aren't in direct competition. WebRTC won't replace SIP or make it irrelevant. WebRTC will just reduce SIP's usefulness in many of the use cases where real-time communication is required.
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