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Nowadays telephony is still used under a specific network integration that is not all IP powered like PSTN numbering and specific rules to manage its traffic with operators.
Until this day, Voice over IP has disrupted the telephony but still providing the same phone user experience; at the end, users are still dialing to a 9-10 digits number and all the call process is mostly the same. Have you ever thought of extending VoIP with different kinds of endpoints to improve its coverage and accessibility? WebRTC is a standard for the web real-time communications or RTC that can be connected to your SIP backbone allowing a web call interface for voice, video, chat messaging, screen sharing or many other smart online interactions.
Build a stronger long-term relationship with your clients.
Not only RTC is used for video conferencing, it can significantly extend your VoIP services. You can provide a convergent access from any modern web browsers or mobile apps, too. Think about the opportunity for your customers to make Voice over IP calls without installing anything on their side - all they need is a browser and and some APIs to make it part of their business processes. This is a key element to make your VoIP service much more attractive and build a stronger relationship with your clients. Provide a good integration and convergence which allows to connect more endpoints and help your customers to improve their communication.
Always stay innovative with all your VoIP services.
RTC is currently becoming more accessible thanks to the WebRTC standard and the convergence of the main web browser in the market. Of course, it’s not easy to integrate with SIP, that’s why you have to provide these features and capacity to your customers to make it as easy as possible for them. As VoIP is quickly becoming a commodity, not providing open interfaces to add value is a risk for any provider that doesn’t pay attention to digital transformation.
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