Practise vocabulary for WebRTC architecture: peer-to-peer vs. SFU vs. MCU, ICE framework, TURN/STUN servers, signaling channel, and NAT traversal.
0 / 10 completed
1 / 10
A team is building a video call for 2 participants. They choose a peer-to-peer (P2P) topology. What is the defining characteristic of P2P WebRTC?
In P2P WebRTC, media travels directly browser-to-browser once the ICE connection is established. For 2 participants this is optimal — no server cost, low latency. For N participants, each peer must upload N-1 streams, making P2P impractical beyond 4–6 participants. For larger calls, an SFU (Selective Forwarding Unit) is used instead.
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What does an SFU (Selective Forwarding Unit) do in a WebRTC architecture?
An SFU (e.g., Mediasoup, Janus, LiveKit) forwards RTP packets without touching the payload — no decode/re-encode, so CPU cost is low. Each participant uploads once to the SFU; the SFU selectively forwards streams to each subscriber. This scales to dozens of participants. Contrast with MCU (Multipoint Control Unit), which mixes streams server-side — heavier CPU, but clients receive a single mixed stream.
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What is the ICE (Interactive Connectivity Establishment) framework in WebRTC?
ICE is WebRTC's connection establishment engine. It gathers candidate types: host candidates (local IPs), server-reflexive candidates (public IP/port discovered via STUN), and relayed candidates (via TURN). ICE then performs connectivity checks on candidate pairs — both peers test all combinations — and selects the highest-priority working pair. This handles NAT, firewalls, and complex network topologies automatically.
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What is the difference between STUN and TURN servers in WebRTC?
STUN (Session Traversal Utilities for NAT) is a lightweight server that simply tells the peer 'your public IP:port is X' — the peer includes this as a server-reflexive ICE candidate. No media flows through STUN. TURN (Traversal Using Relays around NAT) is used when direct connectivity fails: both peers connect to TURN, which relays all media. TURN is expensive (bandwidth cost) but guarantees connectivity — a production WebRTC app always configures TURN as a fallback.
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What is the distinction between the 'media plane' and the 'signaling plane' in WebRTC architecture?
Signaling plane: SDP offer/answer exchange, ICE candidate trickling — all via the app's signaling server (typically WebSocket). This is how two peers negotiate what to connect and how. Media plane: once ICE succeeds and DTLS handshake completes, encrypted RTP/RTCP flows directly P2P (or via SFU/TURN) — the signaling server is completely out of the media path. This separation is fundamental to WebRTC's architecture and why signaling is not specified by the standard.
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During a code review of the WebRTC signaling server, Sarah asks David: 'How does the browser determine if it can directly connect to the remote peer without going through a TURN server?' David replies, 'We use ice.candidates to initiate that connection.' Which of the following best describes what David is referring to?
David is referring to the ICE framework's initial candidate exchange. This involves exchanging IP addresses and port numbers directly between peers – the first step in attempting a P2P connection. The other options describe later stages of negotiation or fallback mechanisms (like TURN) that occur if a direct connection isn't possible. The ice.candidates object is crucial for this initial exchange.
7 / 10
In a Slack channel discussing the architecture of their new WebRTC app, Ben writes: 'We're using an SFU to handle multiple concurrent calls. It's essentially acting as a centralized hub for all incoming media streams.' Which statement best explains the primary function of the SFU in this context?
The SFU (Selective Forwarding Unit) is a critical component for handling multiple concurrent calls. Its core function is to receive media streams from numerous active sessions and forward only the relevant streams to the intended recipients – this avoids overwhelming individual peers' bandwidth and processing capabilities. While encryption, bandwidth management, and session management are important aspects of WebRTC apps, they aren't the primary role of an SFU.
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During a standup meeting, Maria explains their team's WebRTC architecture. She says: 'We utilize STUN servers to discover our public IP address and port. This information is then used in the SDP offer to establish a connection.' What is the primary purpose of using STUN servers in this scenario?
STUN (Session Traversal Utilities for NAT) servers are specifically designed to help WebRTC peers discover their public IP address and port. This is essential because many devices are behind Network Address Translators (NATs), which hide their internal IP addresses from the outside world. The SDP offer relies on this publicly reachable information to initiate a connection. The other options describe functionalities of TURN servers or other related technologies.
9 / 10
You are reviewing a PR that implements the TURN server logic in your WebRTC application. The PR description states: 'The TURN server dynamically redirects media traffic based on network conditions and available bandwidth.' What is the *primary* role of the TURN server in this context?
TURN (Traversal Using Relays around NAT) servers function as relays when a direct P2P connection between peers cannot be established due to NAT restrictions. They receive media data from one peer and forward it to the other, effectively bridging the communication gap. The PR description accurately reflects this core functionality. The other options describe different services that might utilize TURN but aren't its primary purpose.
10 / 10
In a code review comment for the WebRTC application, Alex writes: 'We need to ensure we properly separate the signaling plane from the media plane.' What is the key distinction between these two planes in the WebRTC architecture?
The WebRTC architecture is fundamentally divided into two distinct planes. The *signaling plane* handles the exchange of control information – SDP offers and answers, ICE candidates – necessary for establishing a connection. The *media plane* then utilizes this information to transmit the actual audio and video data itself. Separating these planes improves modularity and simplifies development.
What does this WebRTC & Real-Time Language exercise cover?
This exercise, "WebRTC Architecture Vocabulary", tests your understanding of webrtc & real-time language vocabulary and phrasing through 10 multiple-choice questions drawn from real workplace scenarios.
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Who is this WebRTC & Real-Time Language exercise for?
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How is this different from reading a glossary or blog article?
Exercises like this one are active recall drills — you have to choose the correct term or phrasing yourself, which builds retention faster than passively reading a definition.
Where can I find more WebRTC & Real-Time Language exercises?
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