Given the increasing mobility of the global work force, enabling users to interact with your VoIP endpoints via a web or mobile app interface will arguably be one of the most significant integrations that will be leveraged for business telephony systems in 2021. In this article, we examine how and why to integrate your VoIP system with WebRTC.
WebRTC is an open-source project that provides browsers and other mobile applications with the capability of real-time communication via the use of application programming interfaces (APIs). This allows users viewing a web page to have a rich communications experience with your business that can include text, voice and video—all without downloading any plug-ins or installation files. Instead, the technology is embedded directly into the browser or mobile app.
SIP, on the other hand, is a protocol that enables VoIP endpoints to connect and communicate. SIP is also the protocol that connects VoIP implementations to the public switched telephone network (PSTN).
By integrating WebRTC and SIP-based VoIP systems, you can seamlessly combine a web or mobile app user’s experience with voice services. Web and app users can have direct access to services available via SIP, VoIP, and the PSTN, all from within their browser.
What’s more, web-based or app-based services can be enriched with a voice interface to interact with a company’s call center or help desk, providing users direct access via voice or video to a live or AI agent that can serve them directly.
Depending on the specific service being offered, this can be further integrated with desktop sharing and other collaboration tools, delivering an almost all-encompassing set of features necessary for a one-stop-shop user experience.
Vendors are increasingly creating IP PBXs that integrate with other services. Both on-premise systems and cloud-based SIP service providers enable their products to interact with a variety of other platforms. The primary method for achieving this, especially for WebRTC implementation, is through APIs. These are a standardized set of rules allowing disparate systems to cooperate, coordinate, and integrate their services.
If you’re ready to integrate your voice network with your web or mobile app infrastructure, you’ll have to ensure that the telephony system you are using supports API integration. If the system was purchased recently, it is very likely that this capability is already built in.
For cloud-based systems, almost all providers support APIs. On most of these services, you'll see a section on the web interface that says API or 3rd party apps integration. Sometimes there are preconfigured APIs for things like Salesforce, Zendesk, Teams, or other well-known products.
For on-premise IP PBX appliances, how the integration is implemented depends on the vendor. For example, Yeastar's S-series IP PBXs allow you to integrate APIs by enabling them on one of the settings pages, then inputting the appropriate parameters to integrate them. Click here for specific instructions.
Grandstream's UCM IP PBXs have WebRTC already integrated and enabled. For instructions on how to easily set up a WebRTC connection through the Grandstream UCM6xxx web UI user portal, click here. For integrating Grandstream IP PBXs with other applications using APIs, refer to this document.
For other vendors, consult their API guides accessible from the manufacturers’ websites. Not all vendors provide turnkey API integration with WebRTC, so you may be required to hire a programmer to do the integration. While this may seem more costly up front, it may also for a more flexible, customizable, and deeper integration.
For 2021, consider integrating your VoIP telephone system with your web or mobile applications to improve the user experience for both employees and customers, allowing them to interact with your business communications systems seamlessly, from the convenience of their browser.
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