UDP vs TCP, how much faster is it? [closed]

Question

People say that the major thing TCP gives you is reliability. But that's not really true. The most important thing TCP gives you is congestion control: you can run 100 TCP connections across a DSL link all going at max speed, and all 100 connections will be productive, because they all "sense" the available bandwidth. Try that with 100 different UDP applications, all pushing packets as fast as they can go, and see how well things work out for you.

On a larger scale, this TCP behavior is what keeps the Internet from locking up into "congestion collapse".

Things that tend to push applications towards UDP:

• Group delivery semantics: it's possible to do reliable delivery to a group of people much more efficiently than TCP's point-to-point acknowledgement.

• Out-of-order delivery: in lots of applications, as long as you get all the data, you don't care what order it arrives in; you can reduce app-level latency by accepting an out-of-order block.

• Unfriendliness: on a LAN party, you may not care if your web browser functions nicely as long as you're blitting updates to the network as fast as you possibly can.

But even if you care about performance, you probably don't want to go with UDP:

• You're on the hook for reliability now, and a lot of the things you might do to implement reliability can end up being slower than what TCP already does.

• Now you're network-unfriendly, which can cause problems in shared environments.

• Most importantly, firewalls will block you.

You can potentially overcome some TCP performance and latency issues by "trunking" multiple TCP connections together; iSCSI does this to get around congestion control on local area networks, but you can also do it to create a low-latency "urgent" message channel (TCP's "URGENT" behavior is totally broken).

In some applications TCP is faster (better throughput) than UDP.

This is the case when doing lots of small writes relative to the MTU size. For example, I read an experiment in which a stream of 300 byte packets was being sent over Ethernet (1500 byte MTU) and TCP was 50% faster than UDP.

The reason is because TCP will try and buffer the data and fill a full network segment thus making more efficient use of the available bandwidth.

UDP on the other hand puts the packet on the wire immediately thus congesting the network with lots of small packets.

You probably shouldn't use UDP unless you have a very specific reason for doing so. Especially since you can give TCP the same sort of latency as UDP by disabling the Nagle algorithm (for example if you're transmitting real-time sensor data and you're not worried about congesting the network with lot's of small packets).

UDP is faster than TCP, and the simple reason is because its non-existent acknowledge packet (ACK) that permits a continuous packet stream, instead of TCP that acknowledges a set of packets, calculated by using the TCP window size and round-trip time (RTT).

For more information, I recommend the simple, but very comprehensible Skullbox explanation (TCP vs. UDP)

with loss tolerant

Do you mean "with loss tolerance" ?

Basically, UDP is not "loss tolerant". You can send 100 packets to someone, and they might only get 95 of those packets, and some might be in the wrong order.

For things like video streaming, and multiplayer gaming, where it is better to miss a packet than to delay all the other packets behind it, this is the obvious choice

For most other things though, a missing or 'rearranged' packet is critical. You'd have to write some extra code to run on top of UDP to retry if things got missed, and enforce correct order. This would add a small bit of overhead in certain places.

Thankfully, some very very smart people have done this, and they called it TCP.

Think of it this way: If a packet goes missing, would you rather just get the next packet as quickly as possible and continue (use UDP), or do you actually need that missing data (use TCP). The overhead won't matter unless you're in a really edge-case scenario.

When speaking of "what is faster" - there are at least two very different aspects: throughput and latency.

If speaking about throughput - TCP's flow control (as mentioned in other answers), is extremely important and doing anything comparable over UDP, while certainly possible, would be a Big Headache(tm). As a result - using UDP when you need throughput, rarely qualifies as a good idea (unless you want to get an unfair advantage over TCP).

However, if speaking about latencies - the whole thing is completely different. While in the absence of packet loss TCP and UDP behave extremely similar (any differences, if any, being marginal) - after the packet is lost, the whole pattern changes drastically.

After any packet loss, TCP will wait for retransmit for at least 200ms (1sec per paragraph 2.4 of RFC6298, but practical modern implementations tend to reduce it to 200ms). Moreover, with TCP, even those packets which did reach destination host - will not be delivered to your app until the missing packet is received (i.e., the whole communication is delayed by ~200ms) - BTW, this effect, known as Head-of-Line Blocking, is inherent to all reliable ordered streams, whether TCP or reliable+ordered UDP. To make things even worse - if the retransmitted packet is also lost, then we'll be speaking about delay of ~600ms (due to so-called exponential backoff, 1st retransmit is 200ms, and second one is 200*2=400ms). If our channel has 1% packet loss (which is not bad by today's standards), and we have a game with 20 updates per second - such 600ms delays will occur on average every 8 minutes. And as 600ms is more than enough to get you killed in a fast-paced game - well, it is pretty bad for gameplay. These effects are exactly why gamedevs often prefer UDP over TCP.

However, when using UDP to reduce latencies - it is important to realize that merely "using UDP" is not sufficient to get substantial latency improvement, it is all about HOW you're using UDP. In particular, while RUDP libraries usually avoid that "exponential backoff" and use shorter retransmit times - if they are used as a "reliable ordered" stream, they still have to suffer from Head-of-Line Blocking (so in case of a double packet loss, instead of that 600ms we'll get about 1.5*2*RTT - or for a pretty good 80ms RTT, it is a ~250ms delay, which is an improvement, but it is still possible to do better). On the other hand, if using techniques discussed in http://gafferongames.com/networked-physics/snapshot-compression/ and/or http://ithare.com/udp-from-mog-perspective/#low-latency-compression , it IS possible to eliminate Head-of-Line blocking entirely (so for a double-packet loss for a game with 20 updates/second, the delay will be 100ms regardless of RTT).

And as a side note - if you happen to have access only to TCP but no UDP (such as in browser, or if your client is behind one of 6-9% of ugly firewalls blocking UDP) - there seems to be a way to implement UDP-over-TCP without incurring too much latencies, see here: http://ithare.com/almost-zero-additional-latency-udp-over-tcp/ (make sure to read comments too(!)).