WebRTC, not being an application, can be thought of as the framework for
great new solutions that can be developed without needing or having to
download plugins or other special apps. With WebRTC any device with a
browser can be enabled for real-time peer-to-peer communications where one
device can establish communication with another device just using the
browser thereby deceasing latency. As more communication businesses embrace the benefits of real-time
communication we will see WebRTC
companies start to take integral roles within the VoIP industry which will
incorporate these advances into their products and solutions. Some hosted
VoIP providers are already integrating WebRTC, starting with voice after which they
plan in the future to move to file transfers and video.
WebRTC mandates encryption which secures communication while still supporting the most popular high quality audio and video codecs. Instant communication through websites and video conferencing is proving to become a major piece of communication development, adding enormous capabilities to companies that currently only offer voice products. This convenient technology makes meeting and then connecting easier than ever. Adding the ability to manage calls directly through the web UI and integrate a call link into a website to enable easy access to free calls for customers offers an outstanding advantage.
Grandstream already uses WebRTC with IPVideoTalk their cloud-based video, audio, and web conferencing service that allows users to engage in meetings from anywhere at anytime from most browsers and communication devices. Well designed, easy to use this product received a 2017 WebRTC Product of the Year Award. Hosted VoIP companies are embracing WebRTC, with both OnSip and RingCentral already winning accolades for outstanding achievements. Google and FaceBook already implement WebRTC in their chats.
Here's the steps for a session with WebRTC:
WebRTC uses UDP, just like VoIP. UDP, unlike TCP, does not require an acknowledgement making it faster. It sends a continuous stream where some packets may be lost. With voice or video these packets are discarded, the end result being negligible. Conversations and video pictures are still usable if packet loss is low. However when sending data files or documents, any packets that are lost can corrupt the entire document, which becomes a major challenge to overcome.
Video conferencing is taking a big step forward with WebRTC. However, bandwidth capacity will remain to be of particular importance, especially for video communication due to its greater requirements.