TCP vs UDP: Differences, Use Cases, and How to Choose

Every packet you send today rides on one of two protocols: TCP or UDP. Web pages, emails, video calls, DNS lookups, online games, scraping requests — all of it. The two protocols solve the same problem (moving data between two machines) in almost opposite ways, and picking the wrong one will cost you in either speed or reliability.

Here's how they actually work, where each one fits, and how your choice affects proxies and scraping infrastructure.

What TCP and UDP Actually Do

Both protocols sit at the transport layer of the network stack — one level above IP, one level below the application protocols you use directly (HTTP, DNS, RTP, and so on). Their job is to take your data, hand it to the network, and get it to the other side.

The difference is what happens in between.

TCP (Transmission Control Protocol) opens a connection, splits your data into ordered segments, confirms every segment arrived, and retransmits anything that got lost. It's a conversation with receipts.

UDP (User Datagram Protocol) wraps your data in datagrams and sprays them at the destination. No connection, no confirmation, no retransmission. If a packet vanishes, UDP doesn't know and doesn't care.

That's the whole disagreement. Everything else — speed, reliability, overhead, use cases — falls out of those design choices.

How TCP Works

A TCP connection starts with a three-way handshake:

1. The client sends a SYN packet ("I want to talk").
2. The server replies SYN-ACK ("Sure, I'm here").
3. The client replies ACK ("Great, starting now").

Only then does data start flowing. Each segment carries a sequence number, and the receiver ACKs the bytes it got. If an ACK doesn't come back in time, the sender assumes loss and retransmits. TCP also handles flow control (don't overwhelm a slow receiver) and congestion control (don't overwhelm the network).

This is why TCP feels heavy. You're paying for a handshake before any real data moves, and you're paying for every ACK after that. On a clean local network it doesn't matter. Over a busy mobile connection with 200ms of latency, it adds up fast.

There's also a quieter cost called head-of-line blocking. Because TCP delivers bytes in order, a single lost packet stalls everything behind it until the retransmission arrives — even if the later packets are already sitting in the receive buffer. For a file download nobody notices. For a video call, you feel it.

How UDP Works

UDP has no handshake. The client sends a datagram. The server (if it's listening) receives it. That's the entire protocol.

Each datagram is independent. There's no sequence number guaranteeing order, no ACK confirming arrival, no retransmission. If your application needs any of that, it has to build it on top of UDP itself.

That sounds reckless, and for some workloads it would be. For others, it's exactly what you want. A live video frame that arrives 400ms late is worse than useless — it's actively harmful, because it puts you behind real time. Dropping it and moving to the next frame is the right call. UDP makes that the default behavior.

When to Use TCP

Use TCP when you cannot tolerate missing or reordered data:

• HTTP and HTTPS. Web pages, REST APIs, GraphQL — anything that needs the complete response.
• Email (SMTP, IMAP, POP3). A truncated email is a broken email.
• File transfers (FTP, SFTP, SCP). One missing byte corrupts the file.
• Web scraping and data collection. Your scraper needs the full HTML, not 94% of it.
• SSH and remote access. Commands and output have to arrive intact and in order.
• Database connections. Postgres, MySQL, Redis — all TCP.

If you're building anything that talks to an HTTP endpoint, you're using TCP whether you think about it or not.

When to Use UDP

Use UDP when latency matters more than completeness:

• Live video and voice (WebRTC, RTP, VoIP). A dropped frame is invisible; a delayed frame is a stutter.
• Online gaming. Position updates have a shelf life of about 50ms. Old data is garbage.
• DNS queries. A DNS request is one packet. Setting up a TCP connection for that would be absurd.
• Real-time telemetry, IoT sensors, monitoring. High volume, loss-tolerant.
• Live streaming protocols (parts of HLS, most of RTSP, all of SRT and QUIC-based streams).

The pattern: if a late packet is worse than a missing packet, use UDP.

What About QUIC and HTTP/3?

This is where the old "TCP for web, UDP for streaming" framing starts to break down. QUIC is a transport protocol Google built on top of UDP that adds TCP's reliability features (and TLS encryption) at the application layer. HTTP/3 runs on QUIC, which means it runs on UDP.

So when you load a modern site over HTTP/3 — Google, Facebook, Cloudflare-fronted properties — your "web traffic" is technically UDP. QUIC sidesteps head-of-line blocking, handshakes faster than TCP+TLS, and survives network changes (Wi-Fi to cellular) without dropping the connection.

The protocol underneath HTTP is shifting, and a growing chunk of the web is no longer pure TCP. If you're scraping or proxying these targets, your stack needs to handle both.

How This Affects Proxies and Scraping

This is where protocol choice stops being abstract.

HTTP/HTTPS proxies run on TCP. They speak the HTTP protocol, which means they understand requests and responses, can rewrite headers, and can be used directly by any HTTP client. For web scraping, residential and datacenter proxy networks, and most data collection work, this is what you want. The overhead is invisible compared to the cost of getting blocked.

SOCKS5 proxies support both TCP and UDP. SOCKS operates lower in the stack — it doesn't care what application protocol you're running. That makes SOCKS5 the right choice for tunneling UDP-heavy workloads: gaming traffic, VoIP, QUIC-based scraping, or any custom protocol that isn't HTTP.

If you're collecting data from sites that have moved to HTTP/3, the proxy in front of your scraper needs to handle QUIC traffic — otherwise you're falling back to HTTP/2 over TCP and missing whatever the target serves only on the newer protocol.