WebRTC is a fantastic technology that allows users to communicate with each other in real-time. While on the surface, it looks like participants don’t need anything more than a browser and a good internet connection to start communicating, underneath the surface, there is a complex architecture of signalling servers that developers need to instruct in order to deliver data from one peer to another.
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Even if based on a peer-to-peer protocol, WebRTC connections can't run without a server. In order for two devices to locate one another, a discovery and negotiation mechanism called signalling must take place.
Since this process is not offered by the WebRTC API itself, developers need to establish a framework that allows WebRTC to overcome the complexities of real-time communication.
As mentioned before, exchanging real-time communication (video, audio, screen sharing and other data) requires devices to connect to a mutually agreed upon server. STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers are both used in the signalling process but for different reasons.
Before establishing a peer-to-peer connection, it is essential for servers to identify the IP address for each participant. WebRTC applications use STUN servers most of the time. They are straightforward, quick, and most importantly, they don’t create much load.
Additionally, they are applied only during the connection setup for discovering and exchanging external host:port-pairs. Once the session has been established, media will be exchanged directly between participants. Nevertheless, STUN is sometimes restricted by firewalls. Therefore, developers need to "turn" and utilize a TURN server when STUN fails.
Firewalls and closed corporate environments can limit access to the IP address. Without unique identifiers, it becomes unfeasible for two networks to find each other. This problem can be solved by using TURN servers as a middleman. Unlike STUN, a TURN server remains in the media path after the connection has been established.
However, using this walk-around can add some latency to your video conference. On top of that, it can also increase the operational costs of your IT infrastructure. The more traffic you direct to your TURN server, the more complex infrastructure you will need.
Understanding STUN and TURN protocols is an integral part of comprehending the complex networking landscape that supports real-time communication. However, when it comes to implementing these protocols into your software products or websites, you'll need a reliable and robust solution.
Digital Samba, a GDPR-compliant video conferencing API and SDK provider, excels in providing such solutions. Digital Samba's technology incorporates the use of STUN and TURN protocols to ensure a smooth, uninterrupted, and high-quality video conferencing experience. Whether your requirements involve peer-to-peer video streaming or conducting conferences with numerous participants from across the globe, our technology provides the framework to ensure seamless communication.
One of the primary advantages of Digital Samba lies in its compliance and data protection standards. Hosted in the EU and offering end-to-end encryption, our video conferencing solutions align with the stringent GDPR guidelines, ensuring maximum data privacy and security.
Moreover, our flexible API and SDK allow for effortless integration into your existing software products or websites. This enables a seamless transition towards more efficient and secure communication, fostering improved collaboration and productivity.
By leveraging Digital Samba, you can effectively navigate the complexities of real-time communication protocols, while ensuring compliance, data security, and an outstanding user experience. The perfect blend of STUN and TURN protocols with our video conferencing technology offers a comprehensive communication solution to meet your unique business needs.
Choose Digital Samba, your trusted partner for GDPR-compliant, EU-hosted, and end-to-end encrypted video conferencing.
Real-time video communication comes with a series of challenges. Even if WebRTC is designed to work peer-to-peer, it still requires building a whole signalling architecture behind it. This means that scaling up requires an additional investment in server infrastructure. One of the main key benefits of video APIs is freeing developers from dealing with this complexity.
CPaaS and VCaaS platforms can handle the signalling processes, accommodating large and frequent load changes. This way, developers can dedicate more time to developing their applications. Relying on video engines that already take care of the signalling process can be a game changer!