WebRTC server: What is it, and do you need one?

If you’ve ever been immersed in a live online event, you know it’s the biggest buzzkill to be jolted out of it by a 10-second delay. For businesses providing real-time communications, such delays can frustrate users and potentially lead to a loss of trust and revenue, emphasizing the need for seamless, instantaneous interactions.

Designed to deliver ultra-low latency streaming, WebRTC (Web Real-Time Communications) caters to the soaring demand for real-time interactive video. But how does it achieve this, and how can you leverage its power for seamless communication?

Read our guide to learn more about how WebRTC can elevate your real-time communications, and find out if WebRTC is truly the best option for your business.

What is WebRTC (Web Real-Time Communications)?

WebRTC is an open-source project that enables real-time communication capabilities for web and mobile applications. With WebRTC, developers can create applications that support video, audio, and data communication through a set of APIs.

One of the standout features of WebRTC is its peer-to-peer (P2P) nature. Unlike traditional communication methods that rely on intermediary servers to relay information, WebRTC allows two users to communicate directly with each other. This direct communication pathway enhances efficiency and ensures scalability by eliminating the need to manage and maintain third-party servers.

WebRTC also boasts broad support across major browsers and operating systems, making it a versatile choice for developers aiming to reach a wide audience without compatibility concerns.

The three pillars of WebRTC

WebRTC stands on the shoulders of three foundational JavaScript APIs:

1. GetUserMedia API: The starting point for any WebRTC interaction. This API taps into a user's camera and microphone, capturing raw video and audio data. From video conferencing to live streaming, it all begins here.
2. RTCPeerConnection API: The backbone of WebRTC. Beyond encoding raw data for efficient transmission, it establishes and maintains the peer-to-peer connection and ensures a smooth flow of media data between peers.
3. RTCDataChannel API: WebRTC isn't just about video and audio. This API enables the transmission of other data types, like text messages or files, paving the way for a richer communication experience.

With these foundational elements in place, let's explore the intricacies of WebRTC's server requirements and the various server types integral to its functioning.

What is a WebRTC server?

A WebRTC server is a computer that handles some aspects of establishing peer-to-peer connections, transmitting data, and maintaining connection stability for real-time communication. There are four key WebRTC servers:

1. Media servers
2. Signaling servers
3. TURN servers
4. NAT traversal servers

Generally speaking, WebRTC servers handle the process of negotiating a connection between two devices in a real-time conversation and managing the connection once it’s established. Most of the time, the communicating devices do most of the data sending and receiving. But if you're building your own WebRTC services, you need WebRTC servers to deliver reliable, high-quality real-time communication using WebRTC.

Does WebRTC need a server?

While WebRTC can function in certain scenarios without a server, it does require them for most real-world applications, especially those that operate over the internet. These servers help in tasks like navigating through firewalls, routing data beyond local networks, and ensuring stable and reliable communication.

There are several core types of WebRTC servers:

NAT traversal servers

These servers are essential for establishing connections across different network types and navigating through NAT (Network Address Translation) barriers. There are two main kinds of NAT traversal servers:

1. STUN (Session Traversal Utilities for NAT) servers help discover the public IP address of a device behind a NAT, facilitating peer-to-peer connections.
2. TURN (Traversal Using Relays around NAT) servers act as relay servers by passing data between WebRTC peers when a direct connection fails due to NAT or firewall restrictions.

Signaling servers

These servers coordinate the initial handshake between WebRTC peers, setting up the session's parameters. Signaling servers don't handle media data but are crucial for initiating and maintaining a connection.

Media servers

These servers are especially important for applications that go beyond simple one-to-one communication. Media servers handle tasks like:

• Mixing audio and video streams for group calls or conferences.
• Recording calls for playback or archiving.
• Transcoding media to ensure compatibility between different devices or codecs.

Each server type plays a unique role, ensuring the seamless operation of WebRTC, especially in diverse and complex network

environments.

Common challenges with WebRTC

While WebRTC offers a powerful platform for real-time communication, developers and users alike often encounter a handful of challenges:

NAT traversal issues

Even with STUN and TURN servers, establishing a peer-to-peer connection can sometimes be tricky due to certain restrictive NATs or firewalls.

Echo issues

Users sometimes experience echo during calls, which can be due to various reasons, including hardware issues or lack of proper echo cancellation.

Synchronization

Keeping audio and video streams synchronized can be challenging, especially with fluctuating network conditions.

Variable network conditions

WebRTC needs to adapt to different network conditions, which can lead to varying video quality or occasional drops in connection.

Browser inconsistencies

While WebRTC aims to be universally supported, there can be slight differences in how browsers implement it, leading to potential compatibility issues.

Security concerns

Ensuring secure connections and addressing potential vulnerabilities is a constant challenge, especially with evolving cyber threats.

WebRTC's challenges highlight the inherent difficulties in achieving seamless real-time communication. As with many technologies, its promise is tempered by practical obstacles.

Voice SDK vs. WebRTC

While WebRTC has revolutionized real-time communication, there's another contender in the arena: the Voice SDK. This SDK (software development kit) equips developers with tools, libraries, and protocols to seamlessly weave voice communication into applications.

Voice SDK is a commercial product that provides a more complete set of features for creating real-time communication applications. It typically includes features such as call control, audio and video quality, and security. Unlike WebRTC, which often requires setting up your own server, Voice SDK offers a serverless alternative, simplifying the integration process. Businesses and organizations that need a high-quality, reliable, real-time communication solution often use Voice SDKs.

What makes Voice SDK superior to WebRTC?

While WebRTC has its merits, Voice SDK stands out for businesses seeking advanced features, customization, and reliability. For those prioritizing quality across platforms, Voice SDK is often the better choice for several reasons:

• Features: Voice SDKs often surpass WebRTC in features, boasting advanced noise cancellation, voice analytics, and more.
• Customization: Voice SDKs offer developers the liberty to tailor voice functionalities, ensuring a distinct user experience.
• Reliability: SDKs are performance-optimized, promising top-notch voice communication even in less-than-ideal network conditions.
• Integration: SDKs can be integrated with analytics, CRM systems, and more, offering a comprehensive solution.
• Support: Many Voice SDKs are versatile, functioning seamlessly across platforms, ensuring users have a consistent experience.

Unlock the power of real-time communication with Telnyx Voice SDK

WebRTC has reshaped the landscape of real-time communication, offering a robust foundation for developers. However, to achieve seamless, high-quality communication, integrating the right tools is essential. This is where the Voice SDKs come into play.

Telnyx's Voice SDK allows developers to integrate WebRTC into their apps effortlessly, bridging the gap between browsers and ensuring real-time communication. With the Telnyx SDK, you can also communicate with other devices with your SIP infrastructure through our JavaScript SDK, connecting WebRTC and traditional communication protocols. This integration enhances the customer experience and streamlines your communications setup.

But it doesn't stop there. When you build WebRTC communications powered by Telnyx's global communications network, it ensures reliability, scalability, and top-tier performance.

How to set up Voice SDK with Telnyx

If you’re integrating Voice SDK with Telnyx, here's a basic roadmap:

1. Register with Telnyx: Sign up for a Telnyx account to access our Mission Control Portal.
2. Establish a SIP connection: Create a SIP connection using credentials in the portal.
3. Enable SIP URI calls: Adjust settings in the portal to allow your SIP connection to receive calls from a SIP URI.
4. Configure your profile: Set up an outbound voice profile in the portal, which dictates how your traffic is routed from Telnyx.
5. Access your API key: Generate an API key from the portal, allowing you to interact with the Telnyx API.
6. Create a token: For secure authentication, create an on-demand credential and generate a token.
7. Test your setup: Use the Voice SDK demo app to make a test call, ensuring everything functions correctly.

Talk to a Telnyx expert to enhance your voice communications and elevate your user experience with Telnyx’s Voice SDK. Or use our Quick Start Guide to get started today.