The history of long-distance communication dates back to when ancient civilizations first used horns, drum beats, and smoke signals to send messages.
Over the centuries, finding better ways of communicating has driven human innovation. This led to the creation of the first acoustic telephone in the 1600s, followed by a wave of telecommunications advancements in the 19th century, which resulted in the retirement of the trusty postal pigeon.
Fast forward to 1990, and the internet as we know it was born. What began primarily as a system for the military and academia was democratized, becoming the universally accessible world wide web.
Since this revolution, significant technological advances in networking have enabled the world to communicate at greater speed, and with improved accuracy, reliability, and security.
Here we look at how telecommunications continues to advance, focusing on the latest communications protocols – SIP and WebRTC/Turn – and how WebRTC-CDN ushers in a new era of optimized network connections to support the next generation of real-time communication (RTC).
A Brief History of Telecommunications and Networking
The birth of the telecommunications industry was heralded by the creation of the first optical telegraph in 1790. This technology quickly advanced, and by 1858 the first trans-Atlantic telephone cable was completed.
The following century was a whirl of innovation, with the development of long-distance telephony, first via radio and later cable.
Telecommunication traveled into space in the 1960s thanks to the passing of the Communications Satellite Act in 1962, and within just two years AT&T had already put six telecommunication satellites into orbit.
In the same decade, fiber optics first appeared, as did the first computer network and internet’s predecessor; the Advanced Research Projects Agency Network (ARPANET).
When ARPANET switched off its old NCP network protocols in 1983 in favor of the Transmission Control Protocol and Internet Protocol (TCP/IP), researchers began to develop the ‘network of networks’ that become today’s modern internet. Then, in 2003, a way to transmit phone calls over a computer via internet protocols was discovered and Voice Over IP (VoIP) was born.
How the Internet’s First ‘Rules’ Were Introduced
Network protocols are sets of rules for formatting and processing data and were created to enable computers to communicate with each other.
Different protocols are used for different kinds of functions and are layered as a stack, with each layer addressing a set of requirements needed to communicate successfully.
The Core Protocols of the Internet
Most internet traffic is driven by these key network protocols.
Internet Protocol (IP)
The IP is responsible for routing data along the ‘information superhighway’ by indicating where data packets – small units of data – come from, and where they’re heading.
Transmission Control Protocol (TCP)
A transport protocol, TCP’s role is to ensure the smooth transfer of data packets across a network.
Hypertext Transfer Protocol (HTTP/HTTPS)
The Hypertext Transfer Protocol (HTTP) is used to transfer data between devices, and puts data into formats applications can use. Web content such as HMTL, CSS, and JavaScript are all transported using HTTP.
A weakness with HTTP was that messages weren’t encrypted. HTTPS was created to add this extra level of security.
Transport Layer Security (TLS)
The Transport Layer Security (TLS) protocol is used to encrypt HTTPS messages. It’s an enhanced version of the Secure Sockets Layer (SSL) protocol, but the terms continue to be used interchangeably.
SIP Creates Real-Time Telephony
As the use of real-time applications such as VoIP and video conferencing grew, the Session Initiation Protocol (SIP) was created to meet the real-time needs of IP-based communication.
Originally conceived by the telephony industry to improve the set-up and handling of telephone calls, technologists quickly adopted SIP as a way to simplify RTC communication over networks.
Created by the Internet Engineering Task Force in 1996, and standardized in 1999, this signaling application protocol is used to start, maintain and end real-time communications between two or more endpoints. Essentially, it is a network ‘switchboard operator,’ supporting a range of unified communications.
WebRTC Arrives to Enable Real-Time Communications
Web Real-Time Communication (WebRTC) arrived in 2011 as an alternative to SIP.
Before, websites and applications that used real-time communication – audio, video, and data sharing – often required plug-ins. WebRTC brought an end to that by enabling applications to run RTC with a JavaScript API.
At first, support was mixed, as the tech giants introduced competing solutions, but in 2021 WebRTC had become an official standard and today is used by all major browsers and 95% of web users.
Types of WebRTC servers
WebRTC requires four different servers in order to work.
- A WebRTC application server is essentially the piece of the puzzle that serves you the web page when you open the application’s website.
- WebRTC signaling servers find and connect users/endpoints. Within WebRTC, four primary signaling protocols are used. These are SIP: the dominant VoIP protocol, XMPP: a presence and messaging protocol, MQTT: used mainly for IoT communications, and proprietary: where people pick an alternative solution that simply works for them.
- WebRTC media servers support more complex RTC scenarios, processing media for features such as live streaming, group calling, cloud rendering, and recording.
- Network Address Translation (NAT) traversal servers are important to WebRTC as they ensure media can travel safely between applications. They work by establishing and maintaining IP connections through NATs and firewalls.
There are two types of NAT servers: STUN and TURN.
- The main purpose of Session Traversal Utilities for NAT (STUN) is to ask the question ‘what is my public IP address’ and then share this in order to transfer data/media directly to the other users/endpoints.
- Traversal Using Relays around NAT (TURN) is used to transmit media/data when STUN isn’t an option because the user cannot be connected directly.
As TURN uses more bandwidth and power, it’s rare to have a public TURN server, instead, organizations usually need to install and maintain their own, or pay for a hosted service.
WebRTC Spurs New Trends and Innovations
In response to emerging use cases, WebRTC is evolving. Through future improvements and additions to the standard, technologists aim to support live processing of audio and video feeds, internet of things (IoT), machine learning, virtual reality gaming, and one day, the metaverse.
Here we take a look at some of the latest trends spurring innovation.
WebRTC must handle bigger meetings
Hybrid working’s now part of our ‘new normal’ and requires reliable, secure RTC that can handle large numbers of meeting attendees. With this, and the growth of hybrid conferences, WebRTC is having to up its game, as it was designed to handle smaller groups.
Background blurring and noise replacement
As virtual meetings have become more mainstream users have begun to care about protecting their privacy by blurring their background and blocking out ambient noise (particularly useful when you live with other people). Originally ‘nice to haves’, they’re now becoming expected features and with this, will need to be better incorporated in WebRTC.
Security
WebRTC is considered highly secure, but as features advance, it’s important security does the same. One of the latest advancements is E2EE enablement in media servers. This provides end-to-end encryption in a group video call and has been made possible due to the introduction of insertable streams to WebRTC.
Investments in different video codecs
As technology firms look into new use cases for the latest video codecs – VP9 and AV1 – WebRTC, which primarily uses VP8, will need to ensure it can support these new formats.
More niche products
In response to the need for off-the-shelf solutions when Zoom won’t suffice, we can expect to see new and improved niche products that provide WebRTC solutions for specific business use cases, such as telemedicine.
These are exciting times, and among technology firms, there’s a huge propensity to move fast and deliver big changes.
“But in many cases, the networks themselves aren’t ready for the next generation of apps built on WebRTC. Without considering how we support real-time apps, there’s a risk that all of this energy and innovation will be hampered by the network that transports these streams,” notes Mo Nezarati (VP of Voice, Subspace).
What’s Next? Network Readiness Demands WebRTC-CDN
This challenge is being addressed by WebRTC-CDN, which provides global WebRTC acceleration and NAT traversal.
By eliminating the need for TURN servers, and side-stepping the initial negotiations required to establish connectivity, WebRTC-CDN does away with latency issues that often slow real-time apps.
“This is resolving some of the inherent challenges of WebRTC, and helping to pave the way for this next chapter,” says Nezarati.
Planet-scale turn provisioning
With Subspace’s global network, users bypass regional limitations and can rely on lag-free RTC.
Security without sacrificing speed
End-to-end security and always-on DDoS protection reduce attack services without affecting network speed.
Simple and effective APIs
A global IP level proxy you can connect to simple APIs. Just provision, point your traffic, and go.
Step into the Era of WebRTC-CDN
WebRTC-CDN is a truly disruptive solution that can support the exciting new era of WebRTC.
Providing the quality of experience (QoE) real-time users demand, there’s no compromise on performance, just an optimized network that can support the next generation of RTC services.
Skip the current challenges of WebRTC and prepare for the future with WebRTC-CDN.
