In some cases, 4 packets are lost successively. This means the congestion caused byTCPsynchronization is not resolved for a duration in which UDP transmits 4 packets. Danitza Brando Pundit. What is TFTP used for? Itis used where user authentication and directory visibilityarenot required.
Aleksei Hijano Teacher. Why TCP is reliable? TCP Reliability. TCP provides fortherecovery of segments that get lost, are damaged, duplicatedorreceived out of their correct order.
TCP is described asa' reliable ' protocol because it attempts to recoverfromthese errors. TCP also requires that an acknowledgemessagebe returned after transmitting data. Celina Teacher. What is quic app? An experimental implementation is being put in place in Chromeby ateam of engineers at Google. Amor Bohnisch Teacher. Is UDP connection oriented? TCP is connectionoriented — once a connection isestablished, datacan be sent bidirectional.
UDP is asimpler, connectionlessInternet protocol. Multiple messages aresent as packets in chunksusing UDP. Nicolita Tiessen Teacher. What is the header size of UDP packet? Maissae Eisenhut Reviewer. Why TCP is connection oriented? For connection - oriented communications, each endpointmust be able to transmit so that it can communicate. Becausetheycan keep track of a conversation, connection - oriented protocols are sometimes describedas stateful.
Idania Motke Reviewer. Is IP connection oriented? Data istransmitted link by link; anend-to-end connection is neverset up during thecall. Chat in the GameDev. Back to Networking and Multiplayer. Networking and Multiplayer Programming. Started by ehmdjii May 18, AM.
Cancel Save. Evil Steve Quote: Original post by ehmdjii i read quite often that UDP packets may not arrive in order or even not arrive at all. TCP will resend the packet if it hasn't been acknowledged in a certain time, and will keep trying to resend it. UDP won't - if the packet is dropped, it's gone. As for how often it happens; no idea, sorry. Someone else will hopefult know. Quote: Original post by ehmdjii also, some network libraries torque, raknet provide "reliable" UDP.
If the packet isn't acknowledged within the timeout period, then the packet is resent by the API. Basically, it's another layer built on top of UDP. There are various methods of identifying and sequencing packets and sequencing is important or you can receive crossovers, where an ack is delayed or fails so the same packet is resent and received twice.
Sometimes this method is reversed, and packets are resent if a whole message fails - for example some packets are received for message ID 7. Non-ack NACKs completion requests are sent with a message ID 7 and list of non-received packets if the message isn't constructed within a certain time all packets for it received. Winterdyne Solutions Ltd is recruiting - this thread for details!
DrakeSlayer Differents IP packets ca take different routes between two computers, thus IP does not garantee packets will arrive in order, if they ever arrive. Packetlost will will occur, depending on network congestion. Well, based on these results, you probably will. Once you hit a certain congestion on a network or a network buffer fills up you suddenly things go a bit sideways. Conversely I've pondered using UDP packets as a "canary in the coal mine" for networks to monitor it's health.
This is how congestion works: queues are not infinitely deep, so the only way to deal with congestion is to at some point start dropping packets.
UDP has no such mechanism, hence you see firsthand the dropped packets. A well-written application should back off in such a scenario so as not to flood the network, but of course many won't. It's just not necessary to explain your observations.
It's often the opposite. In order to prevent the Internet from melting, TCP does an elaborate dance with all the TCP speakers to back off and avoid congesting the network.
Obviously this presumes that the UDP protocol in question has some mechanism for handling lost packets. But, for instance, if you're doing lossy video or forward error correction, end users do not deal directly with lost packets.
Also priority may be a function of the network configuration QoS etc not biased against UDP by default. Very much this.
It's a bad idea to measure UDP reliability on a quiet day and then make decisions based on those results. On a different day, everything will be happening at once - everyone trying to message their Mom, everyone trying to sell stock, everyone trying to cast a spell on the big monster - and that is when the most packets will be dropped.
TCP priority. You can use TCP as this canary as well by monitoring a counter of how many retransmissions have been performed. So perhaps try with 1, 2, or more TCP streams trying to max wire-speed, between the same machines at the same time.
Note also you can create arbitrarily-high loss rates by simply choosing to send more than the path and endpoints can handle. Let one side send X thousand in a tight loop; on the other side only check for 1 per second for X thousand seconds. Avoid timestamps: computer are not perfect, their clocks tends to drifts in non predictable ways even with NTP.
I suspect seen quite real bugs due to clock drifting and linux non monotonic timer which impacted python for a while but problem is solved resulting in packet drops because networks stack don't like to see packets arriving in the future when I was using pyzmq.
My intuition is we are close to relativistic problems. Can't prove it, I have to code a kikoo lol form for tomorrow. So perhaps try with 1, 2, or more TCP streams doing a max-speed copy at the same time. I agree. Having or not other traffic on the network will impact UDP throughput dramatically. RUDP or similar systems allow network messaging to ack when needed and unreliable by default which is fine for positional updates with some missing data using interpolation and extrapolation to simulate missing data.
With UDP and only some reliable calls you drastically improve real-time performance with less queueing. UDP allows saturation later. Pretty good reliability and lack of ordering are not really an issue, if out of order timestamp older than last discard and use next or predict until the next valid message too much of this leads to lag but normal UDP operations it is enough and actually smoother.
Two other reasons to use UDP for low-latency use cases, even if you cannot handle actual missed packets and thus are fully reliable: - If your bandwidth use is small, you can just spam multiple copies of each packet to decrease the chance that a laggy retransmission will be needed.
If you're sending packets at a constant high rate, you can instead include copies of the last N messages in each packet, rather than just the new data. Pretty obvious, but I only see one other mention of this fact in the thread, and it vastly increases the likelihood of being able to make a direct connection between two random consumer devices. Both are actually distinct and different protocols. Most game networking libraries will let the developer choose, when sending a datagram, whether it's reliable, ordered, both or neither.
The reason is that you rarely actually need both, but in the rare cases you do e. RUDP is simply a message-based connection that is both reliable and ordered. You might be able to get away with using it for games, but, why would you - RakNet is open source and is mostly the industry standard. IvyMike on Oct 16, prev next [—]. This is interesting, but in my experience, you can't use this data to usefully extrapolate anything. Slightly different hardware, topology, OSes, drivers, system load, network "weather conditions", etc, can radically change the results.
And they were right--the hardware wasn't dropping anything. This behavior isn't entirely unique to windows btw. Which unfortunately is all they can do, in the limit. Not good for a production system. This is really testing "how unreliable are the network routers, cables and network adapters and drivers!
How unreliable is UDP as a protocol? This is really a binary state, not a percentage-measurable value. If you need reliability in ordering or delivery you need to layer on top of it and unless your network usage has very specific constraints eg. Mithaldu on Oct 16, parent next [—]. JoeAltmaier on Oct 16, root parent next [—]. TCP stands for "Transmission control protocol" which started out as file transfer. These days it is exquisitely unsuitable for most things it get used for, even fetching web pages.
The delays, retries and congestion controls are set arbitrarily and rarely adjusted. In this modern world of wireless roaming and streaming media, TCP has little or nothing to offer, except that its there. I've long wished for a reliable protocol that was negotiated on a per-link basis. I mean: we have the processing power now. So, effectively, packets aren't removed from the router's buffer until it knows the next link has the packet in buffer.
Lots of implementation details to be gone over, though. It seems a mite When packets are being sent too fast the sender needs to be slowed down. Otherwise the buffers would just fill up.
And in traditional TCP, the only thing that tells the sender this is dropped packets. The smart solution you're looking for is TCP ECNs - a way for the routers to say "I'm buffering it for now, but you'd better slow down". If you're running linux it's a kernel setting you can enable they're disabled by default as some routers mishandle them.
Sorry, should have specified. One of the things this suggests is explicit cache management for individual links. Don't treat links as end-to-end. Treat them as a bucket chain - each link in the chain negotiates with its immediate neighbors only. Currently we do a game of "toss the bucket at the next guy and hope he catches it". That would cause abysmal performance because of the head of line blocking of this approach: If one of the outbound links becomes congested, sooner or later, all of the memory would be occupied by packets destined for that congested link, as all of the packets destined for other links would be transmitted, making room for some more inbound packets, of which again all the packets destined for the uncongested links would be transmitted right away, leaving in the buffer the ones destined for the congested link, JoeAltmaier on Oct 17, root parent prev next [—].
Carnegie Mellon has a blank-slate project to manage data center traffic similar to this. The idea is, buffer backup at the source is inevitable if subnet bandwidth is insufficient; you have to accept that.
What you CAN do is avoid pointlessly sending packets to where they can't be kept, because it wastes bandwidth; even worse signaling and retries use even more.
It seems like the routes would have to be a lot more static for that to work, negating the big advantage of the internet over traditional circuit switching. Right now each end-to-end link negotiates its own window size and can accept that many packets before acking, and it doesn't matter whether half of those packets go by one route and half of them by another, they just have to all arrive at the end. TheLoneWolfling on Oct 17, root parent next [—].
Not particularly. You can still do all the fancy and not-so-fancy tricks regarding packet routing. As long as each router knows a "closer" router to the destination, you're fine. This is identical to the current setup in that regard. As a matter of fact, it would probably be easier to make dynamic.
Router A gets a packet for router Z - router A wants to send it to router B, but router B is currently congested, and router A knows that router C is an alternate route, so it sends it to router C instead. Now, there are circumstances where this approach is not particularly valid. In particular, on wireless networks.
However, TCP over wireless networks isn't exactly great either. TCP and this approach both make the same assumption: namely that most packet loss is congestion as opposed to actual packet loss.
This approach is for the segment of the network that's wired routers with little to no packet loss disregarding packets being dropped due to no cache space. Router A knows that router B is congested - but this is actually due to congestion in the link between router K and router L.
How does it know which of router C or D would be using the same link? It has to have a global understanding of all the routing paths, no? Routing the packet to Z and telling you that the path to Z is congested are mirror images of each other; it makes sense to use the same mechanism for both.
TCP has a lot of behaviour you can't work around. UDP is what you write it to be. For games etc, you need to be able to hide seconds of lag for TCP. I thought it was pretty clear in the post you're responding to that there are situations in which UDP is a much better choice, I even called out games specifically.
The question is, do you need reliability as defined by guaranteed delivery and sequencing or not. If you do, UDP is not enough even if it is seems "mostly reliable" in some tests.
If you need the full reliability TCP offers, UDP isn't the answer unless you layer your own protocol on top of it, and while it is possible to layer your own protocol on top of it that will beat TCP for specific use cases, most people who aren't expert network programmers and don't even know about [let alone understand] the minefields of issues you can run into such as dealing with NATing, etc are much better off just using TCP, warts and all.
Animats on Oct 16, prev next [—]. The writer is testing over links between data centers. Those tend to have good bandwidth and no "deep packet" examination and munging.
IgorPartola on Oct 16, prev next [—]. So UDP is great.
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