{"id":23510,"date":"2022-12-30T03:32:00","date_gmt":"2022-12-30T08:32:00","guid":{"rendered":"https:\/\/www.contus.com\/blog\/?p=23510"},"modified":"2025-11-03T03:57:21","modified_gmt":"2025-11-03T08:57:21","slug":"webrtc-protocol","status":"publish","type":"post","link":"https:\/\/www.contus.com\/blog\/webrtc-protocol\/","title":{"rendered":"WebRTC Protocol: Why Are They Best For Real-Time Communication?"},"content":{"rendered":"\n

Explore the four simple steps that explain how video and voice calls are built in real-time communication apps using the WebRTC protocol.<\/em><\/p>\n\n\n\n

People do not miss out on important meetings and family gatherings anymore. Video calls and voice calls let us connect to people even from the other side of the world, right from where we are. While real-time communication is what we hail as the reason, we decided to take a deeper look into the details of the technology behind it – WebRTC Protocols.<\/p>\n\n\n\n

When you make a live video call<\/a> now, almost 10+ protocols will work at the backend to make the connection and communication happen. Sounds cool? Let\u2019s discuss the protocols ahead!<\/p>\n\n\n\n

<\/span>What is WebRTC?<\/span><\/h2>\n\n\n\n

Web Real-time Communication<\/a><\/strong>, popularly known as WebRTC is an open-source communication protocol with a set of rules that enable bi-directional and real-time voice, test and video streaming between devices and web browsers.<\/p>\n\n\n\n

These protocols are available as a WebRTC API<\/a>, written majorly in Javascript. This means, your mobile apps and web browsers will be able to support any format of multimedia content when you integrate a WebRTC API for establishing a peer-to-peer conferencing<\/a> (P2P) connection. <\/p>\n\n\n\n

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<\/span>Fundamental Concept Of WebRTC<\/span><\/h3>\n\n\n\n

To illustrate in simple terms, imagine a scenario where you have 2 browsers – Browser 1 and Browser 2 that need to communicate with each other via voice and video calls. <\/p>\n\n\n\n

Without WebRTC<\/p>\n\n\n\n

When Browser 1 sends a message to Browser 2, the message from Browser 1 first reaches a server. <\/p>\n\n\n\n

\"webrtc<\/figure>\n\n\n\n

The server identifies the Browser 2 and delivers the message.<\/p>\n\n\n\n

\"webrtc<\/figure>\n\n\n\n

This transmission using a server might not always be suitable for real-time communication, where the data delivery must happen in less than \u00bd second. <\/p>\n\n\n\n

With WebRTC<\/strong><\/p>\n\n\n\n

Consider the same situation with WebRTC. When this protocol is used for communication between two browsers, it does not require a server. A WebRTC Protocol directly connects peers for communication. <\/strong><\/p>\n\n\n\n

\"webrtc<\/figure>\n\n\n\n

This is the very reason why developers prefer to use WebRTC protocols for their real-time voice and video call applications. And there are many other benefits you\u2019ll get when incorporating WebRTCs with your real-time projects. Here\u2019s the typical advantages of using the protocol.<\/p>\n\n\n\n

<\/span>Why Choose WebRTC Protocols?<\/span><\/h2>\n\n\n\n

WebRTC is a recent trend in the real-time communication<\/a> industry. Developers around the world use this protocol for a variety of use cases. Here, we\u2019ve listed some of the most common reasons as to why WebRTC protocols are important for your real-time apps:<\/p>\n\n\n\n

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  1. Eliminates Integrations For Quality Control:<\/strong> Whenever the network condition changes, IT teams usually require an integration to manually adjust the communication quality and bandwidth. WebRTC eliminates this need, and ensures that the audio and video adjust to the optimal quality according to the varying bandwidths. <\/li>\n<\/ol>\n\n\n\n
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    1. NAT Traversal: <\/strong>It is easy to transfer media between peers with WebRTCs using STUN, TURN and ICE servers that perform NAT traversal by default, when a transfer request is made. <\/li>\n<\/ol>\n\n\n\n
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      1. Packet-loss concealment: <\/strong>Hugely reduces the data packet loss that occurs during audio and video transmission. <\/li>\n<\/ol>\n\n\n\n
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        1. Robust Security:<\/strong> When it comes to security, WebRTCs have built-in encryptions that protect the user data and communication networks from potential vulnerabilities. <\/li>\n<\/ol>\n\n\n\n
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          1. Dynamic jitter buffering: <\/strong>In order to maximize the audio and video experience, the jitter-buffer aligns with the network transmission characteristics to choose the best latency possible.  <\/li>\n<\/ol>\n\n\n\n
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            1. Sub-second Latency:<\/strong> WebRTC helps you maintain a sub-second latency that is less than 3 or 4 seconds, whenever communication happens. <\/li>\n<\/ol>\n\n\n\n
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              1. Browser Compatibility<\/strong>: You can implement WebRTC into any browser – Google Chrome, Mozilla Firefox, Safari that can run on any operating system. <\/li>\n<\/ol>\n\n\n\n
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                1. Eliminates Need For Plugins: <\/strong>WebRTC is an open-source software that does not require plugins or third-party components to establish real-time communication into any app. <\/li>\n<\/ol>\n\n\n\n

                  Also Read: <\/strong>What is the Flutter WebRTC Plugin? A Complete Guide<\/a><\/p>\n\n\n\n

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                  1. Congestion Control:<\/strong> One of the most important focuses for employing WebRTC in real-time communication setup is to identify and control congestion in the network, to avoid collapse. <\/li>\n<\/ol>\n\n\n\n
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                    1. Bandwidth adaptivity: <\/strong>Even during weak network conditions at the user\u2019s end, WebRTC protocols help you align with the bandwidth, and render optimal quality. <\/li>\n<\/ol>\n\n\n\n

                      Now that you know WebRTCs are super-beneficial, let\u2019s look into the protocols involved in the real-time communication process:<\/p>\n\n\n\n

                      <\/span>Protocol Stack<\/span><\/h2>\n\n\n\n

                      WebRTCs involve a variety of protocols. We\u2019ve listed below the ones involved in the workflow explained ahead in this article and also others that are generally used in a WebRTC transmission. <\/p>\n\n\n\n

                      Voice & Video Exchange<\/strong><\/th>Networking<\/strong><\/th>Security<\/strong><\/th><\/tr><\/thead>
                      RTP<\/td>DHCP<\/td>HTTPS<\/td><\/tr>
                      SRTP<\/td>NAT<\/td>SRTP<\/td><\/tr>
                      STUN<\/td>HTTP<\/td>IPsec<\/td><\/tr>
                      SIP<\/td>SNTP<\/td>SSL\/TLS<\/td><\/tr>
                      SIPs<\/td>TCP\/IPv4\/IPv6<\/td>IKE<\/td><\/tr>
                      <\/td>FTP<\/td>SIPS<\/td><\/tr>
                      <\/td>PPP<\/td>SRTP<\/td><\/tr>
                      <\/td>RIP<\/td><\/td><\/tr>
                      <\/td>PPPoE<\/td><\/td><\/tr>
                      <\/td>SNMP<\/td><\/td><\/tr><\/tbody><\/table>
                      Top SIP Trunking Providers<\/a><\/figcaption><\/figure>\n\n\n\n
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                      <\/span>Workflow Of WebRTC Protocols<\/span><\/h2>\n\n\n\n

                      In general, users expect a high-quality rich interaction experience from any voice or video call app they use. This demands several web browser capabilities like feature-rich audio and video processing, support to a variety of network protocols and new application APIs. <\/p>\n\n\n\n

                      WebRTCs wraps these in style with 3 primary APIs:<\/p>\n\n\n\n

                      1. <\/strong>MediaStream <\/strong><\/p>\n\n\n\n

                      This API is used to acquire the audio and video from cameras and video phones, when communication takes place between two or more devices.<\/p>\n\n\n\n

                      2. <\/strong>RTCPeerConnection<\/strong><\/p>\n\n\n\n

                      This API is an interface that takes care of the initiation, maintenance, monitoring and closure of a connection between local computers and remote peers<\/p>\n\n\n\n

                      3.<\/strong> RTCDataChannel<\/strong> <\/strong><\/p>\n\n\n\n

                      Using this API, the arbitrary data between the peers are transferred bidirectionally in a network channel. <\/p>\n\n\n\n

                      The above-mentioned APIs map with the WebRTC protocols to establish a successful real-time connection between browsers. <\/p>\n\n\n\n

                      Clearly, WebRTC is a combination of a variety of configurations and protocols that is used to establish a successful peer-to-peer connection for real-time communication. While the subject of matter is as broad as an ocean, we\u2019ll break this into simple steps and understand the main processes involved in this technology.<\/p>\n\n\n\n

                      <\/span>Step 1: Signaling <\/span><\/h3>\n\n\n\n

                      Let the WebRTC agents know about each other<\/strong><\/p>\n\n\n\n

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                      Protocols Used For Signaling<\/strong><\/h4>\n\n\n\n
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                      1. Session Description Protocol (SDP)<\/li>\n<\/ol>\n<\/div><\/div>\n<\/div><\/div>\n\n\n\n

                        Signaling<\/strong> is the process of exchanging information between the WebRTC agents (peers) to bootstrap the call. <\/p>\n\n\n\n

                        In this process, a plain text protocol named Session Description Protocol (SDP) <\/strong>collects information about each peer, and exchanges it with each other.<\/p>\n\n\n\n

                        The information exchange includes:<\/p>\n\n\n\n