When companies look for a Computer Telephony Integration with Salesforce, one of the first questions is that it must be possible to handle calls via the computer. From a technical point-of-view, this is possible already for a long time. However, different technologies are used to make this happen. Where classic Unified Communication Platforms communicate via Session Initiation Protocol (SIP), Amazon Connect – and so also Salesforce Service Cloud Voice – uses WebRTC.
So, what is WebRTC and how does it compare with SIP? And is it important to have a basic understanding of WebRTC?
We believe it is important! Sure it is a technical topic, but it is important to understand what happens ‘under-the-hood’ so that you may correctly evaluate your solution architecture.
WebRTC stands for Web Real-Time Communication. It allows for audio and video communication to work in web pages.
As described here above, audio communication happens via your browser. When implementing Salesforce Service Cloud Voice, which uses Amazon Connect in the background, it is important to understand that only Chrome and Mozilla Firefox are supported.
The diagram shows the flow of actions when handling voice calls via Amazon Connect. When using Service Cloud Voice, this translates as follows:
- A user logs into the Omni-Channel widget in Salesforce and makes him/herself available for accepting phone calls
- A voice connection is established between the browser session that is used for Salesforce, and the Amazon Connect server using WebRTC
- Voice connectivity to traditional phones (PSTN) is established between Amazon Connect and AWS’s telco carrier partners
- When using call recording, the recordings get stored in the configured S3 bucket
- Amazon S3 server-side encryption is used to encrypt call recordings
We describe the use case of WebRTC handling voice calls. But as it also supports video communication, a possible (future?) use case could be to implement a ‘Call me’ button on your website that allows a customer to connect instantly and directly with a contact center agent, right from their browser.
How does WebRTC compare to SIP?
Given the usage of a (local) Unified Communication Platform like Avaya or Cisco, you may already be used to calling via a SIP. You may use a desk phone, or have a softphone client application installed on your workstation. And so, when you make calls via the UC platform, SIP is used for the technical communication between your PABX and a client application.
In general, SIP allows for making real-time sessions over your network for voice, video and messaging applications. So…how is WebRTC different from SIP?
Actually, we may think of WebRTC as an evolution of SIP. The same technologies come into play, but WebRTC allows the integration in web browsers, eliminating the intermediate step of soft phones. Given the integration in your web browser, WebRTC also allows easy integration in mobile or tablet implementations.
Although the underlying technology is similar, it is important to understand the difference that WebRTC brings. Where SIP communication ‘ends’ in a dedicated softphone application, the connectivity and quality of the call may clearly get differentiated from other communications. With WebRTC, all – Salesforce, ongoing voice call, other applications – is handled via the browser.
Want to know more?
Are you planning or evaluating an implementation of Salesforce Service Cloud Voice? But you don’t exactly know how to start? Get in touch with us for more information.