WebRTC vs VoIP

Theres is a bit of confusion in the telecommunication industry as to whether or not WebRTC is compatible with or runs against VoIP. WebRTC is a viable Internet Protocol (IP) communications system that parallels and runs alongside the internet-based phone system VoIP. VoIP services and WebRTC solutions are both heavily promoted in the business and residential fields. So the confusion lays here: are VoIP and WebRTC providers friends or foes? Can the two systems coexist, do they overlap, and how does the client benefit from these?

The similarities

WebRTC and V.VoIP are similar in that both aim to enhance the user experience and enable any consumer device (whether it be mobile phone, fax, internet etc.) to effortlessly connect from anywhere and on any network internationally.

The differences

The primary difference between the two services is that VoIP uses a multitude of variants such as VoIP over DSL/cable modem, voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS), while WebRTC is solely focused on browser-based communications.


VoIP is an online telecommunications system which offers simpler and more efficient technology than traditional phone service. VoIP uses advance phone technology in order to make phone calls from the office or home more cost effective and with more features. Standard telephone systems uses telephone lines to transmit phone calls, using physical circuits for connection. Since VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices. Millions of people and businesses have switched to VoIP in order to save money as well as to be able to access the same lines from any place and any device.

Benefits that most VoIP providers include are: around the clock customer service; reduced costs compared to traditional phones; no installation or service fees; free ad-ons including unlimited calling to the US and Canada, unlimited extensions, 1,000 free toll-free minutes, high-definition video-conferencing, desktop integration with popular CRMs, online PBX controls, virtual extensions, remote access, auto-attendants, and unlimited extensions for multiple office locations.


WebRTC (Web Real-Time Communication) is an API being drafted by the World Wide Web Consortium (W3C). Put simply, its a software intermediary that makes it possible for application programs to interact with each other and share data. WebRTC is used to enable browser-to-browser applications for voice calling, P2P file sharing, and video chat without plugins. WebRTC is an emerging technology that are accessed with JavaScript APIs and currently in development are an audio and video data stream as well as API which allow for two or more users to communicate browser-to-browser, real-time gaming, text chat, file transfer and other online based sharing.

Connecting vs. clashing

WebRTC makes it feasible for web developers to enable VoIP into their Web-based applications. Since WebRTC is in its early stages of development, it does not include any signaling protocol which leaves this choice and development and integration to the developer. By integrating a signaling protocol into WebRTC, a developer can create a full VoIP soft client on a browser.
One nice example of such VoIP soft client is CryptoVoIP SIP WebRTCDialer which uses SIP Protocol for signalling. The good part of CryptoVoIP web dialer is that it does not require web-sockets or webrtc support in SIP Servers or softswitch. It works with any existing SIP infrastructure seamlessly. CryptoVoIP WebRTC Proxy converts all WebRTC web-sockets communication to legacy SIP and RTP before coming to your SIP Network.

WebRTC based Products

  • Video RTC Gateway Interactive Powers provides WebRTC and RTMP gateway platforms ready to connect your SIP network and able to implement advanced audio/video calls services from web browers or mobile applications.
  • REVE WebRTC – SIP Gateway REVE WebRTC-SIP gateway is a solution created by REVE Systems which uses WebRTC technology to upgrade your SIP network and enable it to make and receive audio/video calls from any web or mobile browser.
  • CryptoVoIP WebRTC Based WebDialer CryptoVoIP WebRTC based web dialer can place VoIP calls using the SIP protocol directly from the browser without installing any plugin or executable.For demo for testing this web dialer with your SIP Server or switch please follow the steps: 1) Open the link: CryptoVoIP WebRTC Web Dialer 2) Enter your sip account details. 3) Enter your SIP Server IP:port 4) Once the dialer shows registered, you can call to a PSTN no. For more information visit http://www.cryptovoip.in/webrtc.html or contact sales@cryptovoip.in for any enquiry. www.cryptovoip.in
  • CryptoVoIP Click2Call Solution Just a click on a Call us button on a webpage from any visitor of your website from any device is enough to place a voice or video call to your configured number or ring group. A very nice solution to reduce the overheads of toll free numbers. Visit www.cryptovoip.in or send enquiry to sales@cryptovoip.in
  • CryptoVoIP Conferencing solution:- CryptoVoIP provides video conferencing solution. Visit www.cryptovoip.in or send enquiry to sales@cryptovoip.in

See also

Created by: admin, Last modification: Mon 07 of Aug, 2017 (22:27 UTC) by ivrpowers
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