Quick answer: TCP is reliable and ordered but slower. UDP is fast but may lose packets. TCP is used for web browsing, email, and file transfers. UDP is used for video calls, live streaming, and gaming. Modern protocols like WebRTC and QUIC blur the line by building reliability on top of UDP. Read on for the full comparison.
If you've ever wondered why a webpage loads flawlessly every time but a video call occasionally glitches, the answer lies in two transport layer protocols: TCP and UDP. They're the backbone of almost everything that happens on the internet, and understanding how they work helps you make better decisions about the technology you build and use.
This article covers what TCP and UDP are, how they differ, when to use each one, and how modern protocols like QUIC and WebRTC are changing the picture in 2026.
Table of contents
The Transmission Control Protocol (TCP) is a connection-oriented protocol in the transport layer. Before any data is sent, TCP establishes a connection between sender and receiver through a process called the three-way handshake (SYN, SYN-ACK, ACK). Only after this handshake is complete does data start flowing.
Think of TCP as a tracked parcel delivery. Every packet is numbered, the receiver confirms each one, and if anything goes missing, it gets resent. This makes TCP highly reliable but adds overhead.
The User Datagram Protocol (UDP) is a connectionless protocol in the transport layer. Unlike TCP, there's no handshake, no connection setup, and no acknowledgement of received packets. UDP simply sends data and moves on.
Think of UDP as shouting across a room. It's fast and direct, but if someone doesn't hear a word, you don't go back and repeat it. For many real-time applications, this trade-off is worth it.
Here's a side-by-side breakdown of the key differences.
| Feature | TCP | UDP |
|---|---|---|
| Connection | Connection-oriented (three-way handshake) | Connectionless (no handshake) |
| Reliability | Guaranteed delivery with acknowledgements | No delivery guarantee |
| Ordering | In-order delivery guaranteed | No ordering guarantee |
| Speed | Slower (connection setup + acknowledgements) | Faster (no overhead) |
| Overhead | Higher (headers, handshake, acknowledgements) | Lower (minimal headers) |
| Error checking | Yes (checksums + retransmission) | Checksum only (no retransmission) |
| Congestion control | Yes (adapts to network conditions) | No (application-controlled) |
| Broadcasting | Not supported | Supported |
| Data retransmission | Yes (lost packets are resent) | No |
| Best for | Web, email, file transfer, banking | Video calls, streaming, gaming, DNS |
| Used by | HTTP/1.1, HTTP/2, FTP, SMTP, SSH | WebRTC, QUIC/HTTP/3, DNS, VoIP, DHCP |
The key trade-off: TCP sacrifices speed for reliability. UDP sacrifices reliability for speed. Your choice depends on whether your application needs every byte to arrive perfectly (TCP) or needs data to arrive as fast as possible (UDP).
If TCP vs UDP were the only story, networking would be simpler. But modern protocols have evolved beyond this binary choice.
QUIC (originally "Quick UDP Internet Connections") is a transport protocol developed by Google and now standardised by the IETF. It's built on top of UDP but adds many of TCP's reliability features – ordered delivery, congestion control, and encryption – without TCP's performance penalties.
The takeaway: the old "TCP = reliable, UDP = fast" framework still holds at the transport layer, but application-layer protocols like QUIC are increasingly building TCP-like reliability on top of UDP's speed. The boundary between the two is becoming less rigid.
WebRTC (Web Real-Time Communication) is the technology behind browser-based video calls, voice chat, and real-time data sharing. Rather than choosing between TCP and UDP, it uses both – each for what they do best. (For a related comparison, see WebRTC vs WebSocket.)
This hybrid approach is why modern video conferencing works so well. You get the real-time performance of UDP for the media that matters most, with the reliability of TCP for the control messages that set everything up.
You don't have to implement any of this transport logic yourself — Digital Samba's embedded video SDK manages the real-time connection for you.
The right protocol depends on what your application needs. Here's a practical decision framework.
In practice, many modern applications don't pick one protocol exclusively. WebRTC uses both. QUIC builds on UDP with TCP-like features. HTTP/3 runs on QUIC while older HTTP versions run on TCP.
If you're building a real-time application – especially anything involving video, voice, or live interaction – a hybrid approach using WebRTC gives you the speed of UDP and the reliability of TCP without having to manage the protocol complexity yourself.
Both protocols have security strengths and vulnerabilities worth understanding.
TCP's three-way handshake provides a basic level of connection verification – both sides confirm they're ready before data flows. Combined with TLS (Transport Layer Security), TCP connections can be encrypted and authenticated.
However, TCP is vulnerable to:
UDP lacks built-in security mechanisms. There's no handshake to verify the sender, and packets can be easily spoofed. This makes UDP susceptible to:
DTLS (Datagram Transport Layer Security) can add encryption and authentication to UDP communications. WebRTC, for example, mandates DTLS for all peer connections, ensuring that UDP-based media streams are encrypted.
UDP is faster. It has no handshake, no acknowledgements, and minimal headers, so data starts flowing immediately. TCP is slower because it sets up a connection and confirms every packet, but that overhead is what makes it reliable.
Real-time video and audio prioritise low latency over perfect delivery. If a packet is lost, retransmitting it (as TCP would) means it arrives too late to be useful. UDP lets the application skip the lost data and keep the stream moving, which keeps calls smooth.
HTTP/3 runs on QUIC, which is built on top of UDP. QUIC adds reliability, ordered delivery, and built-in encryption, so HTTP/3 behaves reliably at the application layer while using UDP at the transport layer. Earlier versions (HTTP/1.1 and HTTP/2) run on TCP.
Yes. UDP itself offers no delivery guarantees, but applications can add reliability on top of it. QUIC and WebRTC both do this, implementing their own acknowledgement, ordering, and congestion-control logic while keeping UDP's speed.
WebRTC uses both. Media streams travel over UDP for low latency, signalling typically uses TCP-based protocols like WebSocket for reliability, and connectivity checks use UDP first with a TCP fallback when UDP is blocked.
UDP has no built-in connection verification, which makes spoofing and amplification attacks easier. However, security is usually handled at a higher layer: DTLS encrypts UDP communications (as WebRTC requires), just as TLS secures TCP. Neither protocol is inherently safe without these layers.
For a wider view of how these protocols compare in real-time communication, see our guide to comparing WebRTC with other real-time communication protocols.