Pure IP has launched a WebRTC service offering to expand the communication technology to our global customers. SIP trunking and WebRTC are complementary technologies, and when they are combined, they can offer new opportunities in managing company communication.
WebRTC enables real-time communication over peer-to-peer connections. It essentially turns any browser into a phone by utilising the inbuilt real-time protocols without the complexity of installing add-ons or having to use a 3rd party voice application. Pure IP believes it will offer our customers the ability to provide another secure and robust method for communication.
Pure IP is offering the WebRTC solution as a complementary product to our global SIP trunking service so it can provide our users with more communication options. The key differentiator of the Pure IP WebRTC service is that we will manage the WebRTC session and deliver the call to the customer's environment as a standard SIP call. This innovative approach requires the minimum amount of changes on the customer’s side and with little effort, provides a new secure communication channel.
So How Does It Work?
WebRTC is a browser-based technology and is supported on most browsers. Pure IP’s customers are provided with code samples which allow for simple and rapid integration into existing web technologies. The customer’s website effectively becomes an intermediary, allowing customers to control which end users can dial which numbers should they wish. The customer’s website serves some code to the end user, which enables them to place a call via Pure IP’s global infrastructure. This global infrastructure converts these calls for interoperability with SIP and other telephony technologies.
Notably, the audio path is connected between the end user and Pure IP’s regional WebRTC service. This makes call quality independent of any factors that are related to the customer’s environment. Pure IP delivers the call to the customer’s network via SIP without requiring any changes to the customer’s phone system.
We see two immediate cases for the adoption of WebRTC with SIP trunking:
Example Scenario A: Adding WebRTC support for conferencing services.
By adding WebRTC as a mechanism for conferencing, customers can eliminate the need for users to dial a traditional PSTN local or toll-free number to join a conference call. For our customers on Skype for Business, it also eliminates the requirement to connect to a Skype conference via the Skype client. Users can simply connect via a PC, mobile, or tablet to a Pure IP WebRTC session, and Pure IP will deliver that call to the customer's conference environment as a SIP call. The diagram below shows the call flow and protocols. The immediate benefit is to enable a whole new channel for users to connect to a conference service.
Example Scenario B: Adding WebRTC support for sales, customer services, and support.
We all want to offer our customers a secure, reliable, and easy to use method to contact us. By adding WebRTC, it turns any Internet-enabled device into a voice communication device. With Pure IP managing the WebRTC solution, the Pure IP customer will only need to establish a SIP trunk to Pure IP, or they can provide a PSTN number, and we will manage the end-to-end WebRTC call. The contact centre does not need to be replaced, upgraded, or add any functionality to enable a new communication channel.
The Pure IP WebRTC service is designed to accommodate easy adoption in multiple use cases and will support any web-based click-to-call services.
Gary Forrest, CEO of Pure IP, commented on the launch. “At Pure IP, we continue to expand our portfolio of effortless voice solutions to meet our enterprise customers telephony needs while bringing them efficiency and savings through their adoption.”
Pure IP is a market leader and global provider of custom-built, secure voice networks.
