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WebRTC Wonders: Transforming Call Services

Posted by Janvi Anand
Call Handling Services

WebRTC, or Web Real-Time Communication, is a rapidly developing force in the rapidly evolving field of communication technology. It is especially influential in the specialist calling services industry. This extensive lesson tries to give a complete grasp of WebRTC by exploring its fundamental ideas, complex features, and significant influence on contact centres and other call handling services.

Originally designed by Google as an open-source project, WebRTC is a collection of communication protocols and application programming interfaces (APIs) that have been painstakingly engineered to enable real-time data, video, and audio communication within web browsers. To provide smooth peer-to-peer communication without the need for extra plugins or outside apps, WebRTC's main objective.

With skill, the basic components of WebRTC are covered in this lesson, including important details like RTCDataChannel, RTCPeerConnection, and MediaStream. It offers insights into the intricate architecture of WebRTC, emphasizing its significance in reshaping communication experiences on the web. As WebRTC continues to redefine the landscape of real-time communication, this guide serves as a valuable resource for comprehending the intricate details and overarching impact of this groundbreaking technology. Additionally, it delves into the pivotal role of WebRTC signaling in call setup and offers insights into its application in call Centres, heralding a transformative shift in user experiences and technical capabilities. As WebRTC continues to mold the landscape of call handling services, this guide aspires to be an enlightening resource, fostering a profound appreciation for the groundbreaking potential embedded in this transformative technology.

Did You Know? 

WebRTC is an open-source project supported by major browsers like Chrome, Firefox, Safari, and Edge. This has contributed to its widespread adoption and compatibility.

Unpacking the Basics of WebRTC

What is WebRTC?

Google invented WebRTC, which is revolutionizing online real-time communication. With its painstakingly created application programming interfaces (APIs) and communication protocols, this open-source project presents a novel way to enable real-time voice, video, and data transmission directly within web browsers. The core of WebRTC is its dedication to enabling smooth peer-to-peer communication without the need for any plugins or third-party apps. WebRTC reimagines the web-based communication environment by providing an intuitive and seamless experience without the customary need for add-on applications.

Core Components of WebRTC

1. MediaStream (getUserMedia): This API allows web applications to access a user's camera and microphone, facilitating the capture of audio and video streams.

2. RTCPeerConnection: At the core of WebRTC, this API manages the connection between peers, overseeing tasks such as encryption, bandwidth management, and media transmission.

3. RTCDataChannel: Essential for bidirectional communication of arbitrary data between peers, this API is vital for applications requiring data exchange beyond audio and video.

WebRTC Signaling

While WebRTC enables direct communication between browsers, a separate process known as signaling is required to set up the call. Signaling involves the exchange of session control messages between peers to coordinate communication. While WebRTC doesn't define a signaling protocol, commonly used ones include SIP (Session Initiation Protocol) or WebSocket.

WebRTC in Call Centre: A Shift in Call Handling Services

Elevating User Experience

Seamless Browser Integration:

1. WebRTC's standout feature lies in its native integration with web browsers. Call Centre can now offer customers the convenience of making calls directly from a website without external applications, significantly enhancing the overall user experience.

No Downloads or Plugins:

2. Unlike traditional communication solutions requiring downloads, WebRTC operates natively within browsers. This eliminates friction associated with installations, ensuring users can engage in real-time communication effortlessly.

Advancing Call Handling Capabilities

Voice and Video Calls:

1. WebRTC enables call centres to incorporate high-quality voice and video calls directly into their web applications, particularly beneficial for scenarios requiring visual assistance or detailed explanations.

Screen Sharing:

2. For technical support or troubleshooting, WebRTC's screen-sharing capabilities prove invaluable. Agents can view a user's screen in real time, facilitating quicker issue resolution.

Collaborative Tools:

3. WebRTC supports the integration of collaborative tools within call handling services, including document sharing, collaborative editing, and interactive whiteboards.

Technical Advantages for Call Centre

Reduced Latency:

1. WebRTC's peer-to-peer architecture minimizes latency, crucial for real-time communication in call handling services.

Adaptive Bitrate Control:

2. The adaptive bitrate control mechanism ensures optimal video quality based on available bandwidth, crucial for maintaining call quality in varying network conditions.

Secure Communication:

3. WebRTC incorporates encryption protocols, ensuring secure communication channels and maintaining the confidentiality of sensitive information exchanged during calls.

Implementation Challenges and Solutions

Firewall and NAT Traversal:

1. WebRTC may face challenges in traversing firewalls and NAT setups. Implementing ICE (Interactive Connectivity Establishment) and STUN (Session Traversal Utilities for NAT) protocols can address these issues, enabling successful peer-to-peer connections.


2. Efficient scalability of WebRTC-based call handling services requires strategic architecture design. Implementing serverless or edge computing solutions can enhance scalability.

Browser Compatibility:

3. While major browsers support WebRTC, ensuring consistent performance across different browsers requires regular testing and updates to mitigate compatibility issues.

Future Trends and Innovations

WebRTC in Mobile Applications:

1. The proliferation of smartphones makes mobile integration a logical next step. Call handling services are likely to extend their capabilities to mobile applications, leveraging WebRTC for seamless communication on diverse devices.

Augmented Reality (AR) and Virtual Reality (VR):

2. The integration of AR and VR technologies with WebRTC holds immense potential for immersive call handling experiences, especially in sectors like customer support.

AI-Powered Call Handling:

3. Infusing WebRTC with artificial intelligence can lead to intelligent call handling services, where AI algorithms analyze voice interactions, automate responses, and provide valuable insights for service quality improvement.

The Future Landscape of Call Handling Services

As technology evolves, WebRTC stands as a key enabler for transforming call handling services. Its impact extends beyond conventional voice calls, ushering in an era of multimedia-rich communication directly within web browsers.

Embracing WebRTC empowers call Centre to redefine user experiences, enhance technical capabilities, and adapt to emerging trends. While challenges exist, strategic implementation and ongoing innovation are key to unlocking WebRTC's full potential in call handling services.

In conclusion, the journey of WebRTC in call handling services marks an exploration of the possibilities that real-time communication brings to the digital forefront. As we navigate this dynamic landscape, the fusion of WebRTC with call handling services signals a paradigm shift towards more accessible, efficient, and user-centric communication experiences.

Real-World Case Studies: WebRTC Revolutionizing Call Handling Services

1. Zendesk's WebRTC Integration for Customer Support


Zendesk, a leading customer service software company, strategically incorporated WebRTC into its platform to elevate customer support interactions. The objective was to offer users a seamless means of connecting with support agents directly within the Zendesk interface.


Traditional customer support methods often involved lengthy phone calls or the use of external communication tools. Zendesk aimed to streamline this process, reducing resolution times by integrating real-time communication capabilities.


By leveraging WebRTC, Zendesk empowered users to initiate voice and video calls directly within the Zendesk environment. This innovation facilitated more personalized interactions between support agents and customers, resulting in quicker issue resolution.


1. Reduced Resolution Times: The integration of WebRTC significantly decreased the time required to resolve customer queries as agents addressed issues more effectively through direct communication.

2. Enhanced User Experience: Customers appreciated the ability to connect with support agents instantly, eliminating the need for external communication channels.

2. Twilio's WebRTC-Powered Call Center Solutions


Twilio, a cloud communications platform, harnessed the capabilities of WebRTC to enhance its call center solutions. The primary focus was on providing businesses with a reliable, scalable, and browser-based approach to constructing contact center applications.


Traditional call center solutions often necessitated complex infrastructure and lacked scalability. Twilio aimed to simplify the process of building and scaling call center applications while maintaining high-quality communication.


Twilio integrated WebRTC to offer browser-based voice and video calling capabilities. This enabled businesses to create customized call center applications without the need for extensive hardware or software installations.


1. Scalability: WebRTC made Twilio's call center solutions highly scalable, accommodating the evolving needs of businesses without compromising call quality.

2. Cost-Efficiency: The browser-based approach reduced the demand for extensive infrastructure, leading to cost savings for businesses deploying Twilio's call center solutions.

3. Google Meet's WebRTC-Powered Video Conferencing


Google Meet, Google's video conferencing platform, exemplifies the impact of WebRTC on large-scale communication services. WebRTC was instrumental in providing users with a seamless video conferencing experience directly through web browsers.


Enabling users to join video meetings without the need for additional plugins or applications posed a significant challenge. Google aimed to create a universally accessible video conferencing solution.


WebRTC became the linchpin of Google Meet's architecture, enabling users to join video conferences directly through supported web browsers. This eliminated the need for downloads or installations, simplifying the user experience.


1. Widespread Adoption: Google Meet's browser-based approach contributed to its widespread adoption, allowing users to initiate or join meetings with minimal friction.

2. Device Agnosticism: WebRTC ensured Google Meet's accessibility across various devices and platforms, providing users with flexibility and convenience.

4. LiveChat's Real-Time Communication Enhancement


LiveChat, a customer service platform, embraced WebRTC to enhance its real-time communication capabilities. The focus was on enabling businesses to engage with customers through voice and video directly within the LiveChat interface.


LiveChat recognized the increasing need for more personalized and immediate customer interactions. The challenge was to seamlessly incorporate real-time communication features into the existing platform.


WebRTC integration allowed LiveChat users to initiate voice and video calls directly from chat sessions. This feature empowered businesses to provide instant support, fostering a more engaging and efficient customer service experience.


1. Increased Customer Satisfaction: The ability to escalate text-based chats to voice or video calls resulted in higher customer satisfaction, especially in situations where complex issues required a more personalized touch.

2. Efficient Issue Resolution: Agents could better understand and resolve customer issues through direct communication, leading to quicker resolution times.


The real-world case studies presented to showcase the profound impact of WebRTC as a revolutionary force in the realm of call handling services. In the sphere of customer support, exemplified by Zendesk's strategic integration, WebRTC has transcended traditional communication barriers. By facilitating seamless voice and video calls within the Zendesk platform, users experience heightened engagement and faster issue resolution, underscoring how WebRTC enhances user experiences and expedites query resolution.

Twilio's adoption of WebRTC in call center solutions demonstrates the technology's versatility, offering businesses a scalable and browser-based approach. This not only ensures reliable communication but also contributes to cost-efficiency by eliminating the need for extensive infrastructure. Google Meet's video conferencing platform attests to the universal accessibility achieved through WebRTC, making large-scale communication effortlessly available through supported web browsers. Finally, LiveChat's integration of WebRTC for real-time communication underscores the technology's role in boosting customer satisfaction and efficient issue resolution, highlighting its transformative potential in diverse industries.

As businesses increasingly recognize the value of real-time communication, WebRTC emerges as a foundational technology reshaping the future landscape of call handling services, promising continued advancements in user-centric experiences, scalability, and cost-effective solutions.


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