WebRTC Signaling Servers: Everything You Need to Know

17 May 2023

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WebRTC signaling is an unwinding technology that has forever changed Internet communications. The statistics speak for themselves - Custom Market Insights (CMI) predicts that the global WebRTC signaling servers market will expand at a CAGR of about 35% between 2022 and 2030, from an estimated 4 billion USD in 2021 to 87 billion USD by 2030. This indicates the demand for WebRTC application development is increasing.

But are we absolutely clear about what exactly WebRTC signaling is and why it’s so important? Let’s shed some light on it.

Introduction to WebRTC Technology

Web Real-Time Communication technology is designed to facilitate communication between users in web browsers and mobile applications. It was developed by Google in 2011 and is now standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The technology utilizes a blend of JavaScript APIs, browser-specific protocols, and standardized communication protocols to deliver a stable, low-latency communication channel between users.

WebRTC introduces a peer-to-peer communication model across browsers, extending the client-server semantics. The so-called SIP (Session Initiation Protocol) Trapezoid (RFC3261) serves as the inspiration for the most generic WebRTC architectural paradigm.

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General WebRTC Architecture. Image Credit: O’Reilly

One of the critical advantages of WebRTC is its peer-to-peer approach, which eliminates the need for a centralized server to enable communication. This decentralized approach makes communication fast and safe since the intermediary server is absent and can’t be hacked or disrupted.

WebRTC technology also comes with advanced security features such as encryption, which ensures that all communication is secure and private. As a result, users can communicate freely and confidently, knowing that their data is protected from malicious actors.

Another noteworthy feature of WebRTC is adaptability. It can operate across multiple browsers and devices, including desktop computers, smartphones, and tablets. This multi-platform support allows for greater accessibility and flexibility.

WebRTC Signaling Servers

The exchange of information guarantees that parties are connected, can share data, and communicate. WebRTC signaling servers play a crucial part in establishing peer-to-peer communication between browsers. In simple terms, the servers foster the exchange of data between two or more parties communicating over the Internet.

Image credit: TechTarget

WebRTC signaling servers use a Session Description Protocol (SDP) to send information about the user's network IP address, port number, and media type. The SDP messages are exchanged between the two parties, and the signaling server acts as a middleman, ensuring the information is sent and received correctly. The signaling server also handles negotiating the type of media to be utilized, such as audio and video codecs.

Different types of WebRTC signaling servers, such as TURN (Traversal Using Relay NAT) and STUN (Session Traversal Utilities for NAT), can be used to establish connections. For instance, TURN servers come to play when communication is impossible due to restrictive firewalls or NATs (Network Address Translators). Take a look at how it works: Image for post

Image credit: DZone

STUN servers, on the other hand, are utilized to overcome NAT traversal issues.

To get a WebRTC call up and running, the clients must exchange various vital data. This includes

Session-control messages are essential for opening and closing the lines of communication. These messages act as the backbone of the WebRTC infrastructure – without them, establishing a call would be impossible.
Error messages are necessary to ensure that any issues during the call can be quickly identified and resolved. Sharing this info helps prevent any disruptions to the call and ensures that it runs smoothly from start to finish.
Media metadata encompasses a range of details related to the media being used in the call – codecs, codec settings, bandwidth, and media types. By exchanging this data, the users allow the technology to configure and optimize media for the best possible call quality.
Critical data used to establish secure connections is also required in order to set up a WebRTC call. This data helps to ensure that the call is entirely safe and that the privacy of both parties is protected throughout the conversation.
Network data is exchanged between clients to help facilitate the connection itself. This includes the host's IP address and port from outside the network.

Signaling Servers and their Role in WebRTC

WebRTC is a brilliantly designed open-source technology that facilitates instant communication between two web browsers without needing third-party plugins or software. However, to make WebRTCreal-time communication possible, we must include signaling servers in the mix.

So what is the signaling server for WebRTC, and why is it crucial?

Put simply, signaling servers are the intermediaries that facilitate the exchange of data between two devices over the Internet. Without these servers, WebRTC would not be able to establish a connection between two browsers, and the communication process would fail.

But that is just the tip of the iceberg. Let's dive deeper into why signaling servers are indispensable for WebRTC.

First, signaling servers are responsible for exchanging important data: network addresses, connection settings, media capabilities, etc. Without this information, the two browsers would simply fail to find each other.
Secondly, signaling servers help in negotiating the protocols required for the exchange of media data between the two browsers. WebRTC supports several protocols for data transmission, including ICE, STUN, and TURN, that allow for media data exchange smoothly and securely. The signaling servers help to negotiate and establish the proper protocol for data transmission, depending on the network conditions and other factors.
Thirdly, signaling servers facilitate establishing connections between the two browsers. WebRTC is designed to work on a peer-to-peer basis, where the data is exchanged between the two browsers directly without needing a central server. The signaling server for WebRTC helps negotiate the initial connection between the two browsers. In the event of a network failure or firewall issue, it reroutes the connection through a TURN server.

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Types of WebRTC Servers

We've already covered WebRTC signaling servers. But three more types of WebRTC servers are frequently required for WebRTC to function.

WebRTC application servers: They act as a middleman between the front-end interface of an application and its backend technology. These servers handle the signaling process and assist in establishing a connection among different devices. Additionally, they help manage session information, handle group calling, and offer enterprise-level features such as reporting and authentication.
NAT traversal servers for WebRTC: Network Address Translation (NAT) is a technology that enables multiple devices to share the same IP address. However, this technology can cause issues when establishing a direct connection between devices, especially for peer-to-peer WebRTC communication. NAT traversal servers help overcome these issues by enabling devices behind a NAT to exchange real-time data. These servers act as intermediaries between devices, allowing them to communicate directly without issues.
WebRTC media servers: These servers play a crucial role in WebRTC by managing and transmitting media streams such as audio, video, and data. They are responsible for managing media traffic, converting media files into a suitable format for transmission, and providing services such as recording, mixing, and broadcasting. Media servers can operate in two modes: Selective Forwarding Unit (SFU) and Multipoint Control Unit (MCU). SFU manages media streams separately, whereas MCU mixes and transmits media streams.

How WebRTC Signaling Works

WebRTC signaling starts with exchanging Session Description Protocol (SDP) messages between two endpoints. During the WebRTC signaling process, SDP messages help endpoints determine the media capabilities of each device and negotiate a suitable codec for transmitting media. The process involves two crucial steps: offer and answer.

First, the initiating endpoint sends an SDP offer message with a set of preferred media streams. The offer message contains information about the sender's media capabilities, codec preferences, and network configuration. The receiving endpoint responds to the offer with an SDP answer message, providing a list of compatible media streams and codecs.

The exchanging of SDP messages is performed using a signaling protocol like WebSocket, HTTP, or the Session Initiation Protocol (SIP). These protocols signify the transfer of signaling information between endpoints until the connection is established.

Common Signaling Mechanisms

There are several commonly used signaling mechanisms in WebRTC.

Long-polling is a traditional technique that involves repeated requests from the client to the server, where the server holds the request open until new data is available or a timeout occurs. This mechanism can be a helpful way to keep a persistent connection between the client and the server and can work well in many cases. However, long-polling can be inefficient in certain situations due to the significant overhead of repeated requests, especially if the server must maintain many open connections simultaneously.
HTTP involves sending information from the server to the client over a single HTTP connection held open for an extended period. This can be an efficient way to transfer a large amount of data between the client and the server. However, HTTP streaming can also have performance issues if there is a significant delay between the server's response and the client's receipt of the data.
Finally, SIP over the WebSocket mechanism is used to initiate and control media sessions between two or more parties. It enables communication by sending messages between the clients and the server over a single TCP connection. SIP over WebSocket provides an efficient and secure way to establish and control WebRTC sessions without the performance challenges of long-polling or HTTP streaming.

Dealing with Internet Issues

WebRTC Signaling mechanism is powerful, but the Internet is not always perfect. It often presents several challenges that can interrupt the connection between two peers. WebRTC Signaling Mechanism comes equipped with ICE, which is a protocol that is designed to handle Internet issues efficiently. ICE calculates the fastest and easiest NAT traversal route for a packet to reach its destination peer using various techniques.

One of the common Internet issues that the WebRTC ICE server deals with is network address translation (NAT). NAT is a common network design that enables multiple devices to share a public IP address. However, it can sometimes cause problems establishing a connection between two peers. When this happens, ICE springs into action and collects a list of possible ICE candidates, which are essentially potential addresses that can get the peers to connect with each other.

ICE then uses STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relay NAT) servers to transmit packets through the network. STUN is used to detect a device's public IP address and port number, while TURN is used when a direct connection fails due to NAT or firewall restrictions. With the help of these servers, ICE can calculate the fastest and easiest NAT traversal route for a packet to reach its destination peer.

Another Internet issue that the WebRTC ICE server deals with is packet loss. Packets can be lost or delayed during transmission due to Internet congestion or errors. To handle this, ICE provides a real-time feedback mechanism, which allows it to detect when packets are lost and take corrective measures. ICE retransmits the lost packets, selects an alternative path with less congestion, and adjusts the media encoding parameters to reduce the packet size and minimize the chance of future packet loss.

Why is Signaling not Defined by RTC?

Have you ever wondered why WebRTC standards do not specify signaling methods and protocols? Well, the answer lies in the desire to minimize redundancy and maximize compatibility with existing technologies. By leaving the signaling plane up to the application, WebRTC allows for more flexibility in terms of protocol selection. This means that different apps can use their preferred protocols, such as the well-established SIP or Jingle call signaling protocols or custom protocols specific to their use cases. The method is described by the JavaScript Session Establishment Protocol (JSEP). Image for post

Image credit: Medium

In essence, the WebRTC approach is to fully control the media plane while leaving the signaling plane to the application as much as possible. The intent is to exchange the multimedia session description, which contains crucial transport and media configuration details for setting up the media plane. By doing so, WebRTC provides enough information for the application to facilitate signaling without mandating a particular protocol or method.

Furthermore, the WebRTC approach aligns with the spirit of innovation by allowing developers to create custom signaling solutions tailored to their specific needs. This is especially meaningful given the vast range of applications and use cases that WebRTC supports.

The bottom line is that WebRTC seeks to provide a standardized approach to media communication while allowing for flexibility in signaling. This approach helps minimize redundancy and maximize compatibility with existing technologies while promoting innovation and customizability.

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