If you start with the problem of how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP. It just is the right solution for the job.
The three drawbacks of the original TCP algorithm were the window size (the maximum value is just too small for today's speeds), poor handling of missing packets (addressed by extensions such as selective-ACK), and the fact that it only manages one stream at a time, and some applications want multiple streams that don't block each other. You could use multiple TCP connections, but that adds its own overhead, so SCTP and QUIC were designed to address those issues.
The congestion control algorithm is not part of the on-the-wire protocol, it's just some code on each side of the connection that decides when to (re)send packets to make the best use of the available bandwidth. Anything that implements a reliable stream on top of datagrams needs to implement such an algorithm. The original ones (Reno, Vegas, etc) were very simple but already did a good job, although back then network equipment didn't have large buffers. A lot of research is going into making better algorithms that handle large buffers, large roundtrip times, varying bandwidth needs and also being fair when multiple connections share the same bandwidth.
I'll take flak for saying it, but I feel web developers are partially at fault for laziness on this one. I've often seen them trigger a swath of connections (e.g. for uncoordinated async events), when carefully managed multiplexing over one or a handful will do just fine.
Eg. In prehistoric times I wrote a JavaScript library that let you queue up several downloads over one stream, with control over prioritization and cancelability.
It was used in a GreaseMonkey script on a popular dating website, to fetch thumbnails and other details of all your matches in the background. Hovering over a match would bring up all their photos, and if some hadn't been retrieved yet they'd immediately move to the top of the queue. I intentionally wanted to limit the number of connections, to avoid oversaturating the server or the user's bandwidth. Idle time was used to prefetch all matches on the page (IIRC in a sensible order responsive to your scroll location). If you picked a large enough pagination, then stepped away to top up your coffee, by the time you got back you could browse through all of your recent matches instantly, without waiting for any server roundtrip lag.
It was pretty slick. I realize these days modern stacks give you multiplexing for free, but to put in context this was created in the era before even JQuery was well-known.
Funny story, I shared it with one of my matches and she found it super useful but was a bit surprised that, in a way, I was helping my competition. Turned out OK... we're still together nearly two decades later and now she generously jokes I invented Tinder before it was a thing.
Sure, you can reimplement multiplexing on the application level, but it just makes more sense to do it on the transport level, so that people don't have to do it in JavaScript.
This is wonderful to hear.
I have a naive question. Is this the reason most websites/web servers absolutely need CDNs (apart from their edge capabilities) because they understand caching much more than a web developer does? But I would think the person more closer to the user access pattern would know the optimal caching strategy.
[Not a web dev but] I thought each site gets a handful of connections (4) to each host and more requests would have to wait to use one of them. That's pretty close to what I'd want with a reasonably fast connection.
> If you start with the problem of how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP.
I'll add that at the time of TCP's writing, the telephone people far outnumbered everyone else in the packet switching vs circuit switching debate. TCP gives you a virtual circuit over a packet switched network as a pair of reliable-enough independent byte streams over IP. This idea, that the endpoints could implement reliability through retransmission came from an earlier French network, Cylades, and ends up being a core principle of IP networks.
We're still "suffering" from the latency and jitter effects of the packet switching victory. (The debate happened before my time and I don't know if I would have really agreed with circuit switching.) Latency and jitter on the modern Internet are very best effort emphasis on "effort".
The telephone people were basically right with their criticisms of TCP/IP such as:
What about QoS? Jitter, bandwidth, latency, fairness guarantees? What about queuing delay? What about multiplexing and tunneling? Traffic shaping and engineering? What about long-haul performance? Easy integration with optical circuit networks? etc. ATM addressed these issues, but TCP/IP did not.
All of these things showed up again once you tried to do VOIP and video conferencing, and in core ISPs as well as access networks, and they weren't (and in many cases still aren't) easy to solve.
TCP has another unfixable flaw - it cannot be properly secured. Writing a security layer on top of TCP can at most detect, not avoid, attacks.
It is very easy for a malicious actor anywhere in the network to inject data into a connection. By contrast, it is much harder for a malicious actor to break the legitimate traffic flow ... except for the fact that TCP RST grants any rando the power to upgrade "inject" to "break". This is quite common in the wild for any traffic that does not look like HTTP, even when both endpoints are perfectly healthy.
Blocking TCP RST packets using your firewall will significantly improve reliability, but this still does not project you from more advanced attackers which cause a desynchronization due to forged sequence numbers with nonempty payload.
As a result, it is mandatory for every application to support a full-blown "resume on a separate connection" operation, which is complicated and hairy and also immediately runs into the additional flaw that TCP is very slow to start.
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While not an outright flaw, I also think it has become clear by now that it is highly suboptimal for "address" and "port" to be separate notions.
"... some applications want multiple streams that don't block each other. You could use multiple TCP connections, but that adds its own overhead, so SCTP and QUIC were designed to address those issues."
Other applications work just fine with a single TCP connection
If I am using TCP for DNS, for example, and I am retrieving data from a single host such as a DNS cache, I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order. No blocking.^1 If the cache (application) supports it, this is much faster than receiving answers sequentially and it's more efficient and polite than opening multiple TCP connections
1. I do this every day outside the browser with DNS over TLS (DoT) using something like streamtcp from NLNet Labs. I'm not sure that QUIC is faster, server support for QUIC is much more limited, but QUIC may have other advantages
I also do it with DNS over HTTPS (DoH), outside the browser, using HTTP/1.1 pipelining, but there I receive answers sequentially. I'm still not convinced that HTTP/2 is faster for this particular use case, i.e., downloading data from a single host using multiple HTTP requests (compared to something like integrating online advertising into websites, for example)
> I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order.
This is because DoT allows the DNS server to resolve queries concurrently and send query responses out of order.
However, this is an application layer feature, not a transport layer one. The underlying TCP packets still have to arrive in order and therefore are subject to blocking.
> I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order. No blocking.
You're missing the point. You have one TCP connection, and the sever sends you response1 and then response2. Now if response1 gets lost or delayed due to network conditions, you must wait for response1 to be retransmitted before you can read response2. That is blocking, no way around it. It has nothing to do with advertising(?), and the other protocols mentioned don't have this drawback.
> how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP. It just is the right solution for the job
A stream of bytes made sense in the 1970s for remote terminal emulation. It still sort of makes sense for email, where a partial message is useful (though downloading headers in bulk followed by full message on demand probably makes more sense.)
But in 2025 much of communication involves messages that aren't useful if you only get part of them. It's also a pain to have to serialize messages into a byte stream and then deserialize the byte stream into messages (see: gRPC etc.) and the byte stream ordering is costly, doesn't work well with multipathing, and doesn't provide much benefit if you are only delivering complete messages.
TCP without congestion control isn't particularly useful. As you note traditional TCP congestion control doesn't respond well to reordering. Also TCP's congestion control traditionally doesn't distinguish between intentional packet drops (e.g. due to buffer overflow) and packet loss (e.g. due to corruption.) This means, for example that it can't be used directly over networks with wireless links (which is why wi-fi has its own link layer retransmission).
TCP's traditional congestion control is designed to fill buffers up until packets are dropped, leading to undesirable buffer bloat issues.
TCP's traditional congestion control algorithms (additive increase/multiplicative decrease on drop) also have the poor property that your data rate tends to drop as RTT increases.
TCP wasn't designed for hardware offload, which can lead to software bottlenecks and/or increased complexity when you do try to offload it to hardware.
TCP's three-way handshake is costly for one-shot RPCs, and slow start means that short flows may never make it out of slow start, neutralizing benefits from high-speed networks.
TCP is also poor for mobility. A connection breaks when your IP address changes, and there is no easy way to migrate it. Most TCP APIs expose IP addresses at the application layer, which causes additional brittleness.
Additionally, TCP is poorly suited for optical/WDM networks, which support dedicated bandwidth (signal/channel bandwidth as well as data rate), and are becoming more important in datacenters and as interconnects for GPU clusters.
Yeah the fact that the congestion control algorithm isn’t part of the wire protocol is very ahead of its time and gave the protocol flexibility that’s much needed in retrospective. OTOH a lot of college courses about TCP don’t really emphasize this fact and still many people I interacted with thought that TCP had a single defined congestion control algorithm.
Poor for high speed connections () or very unreliable connections.
) compared to when TCP was invented.
When I started at university the ftp speed from the US during daytime was 500 bytes per second! You don't have many unacknowledged packages in such a connection.
Back then even a 1 megabits/sec connection was super high speed and very expensive.
Might be obvious in hindsight, but it was not clear at all back then, that the congestion is manageable this way. There were legitimate concerns that it will all just melt down.
There are a lot of design alternatives possible to TCP within the "create a reliable stream of data on top of an unreliable datagram layer" space:
• Full-duplex connections are probably a good idea, but certainly are not the only way, or the most obvious way, to create a reliable stream of data on top of an unreliable datagram layer. TCP's predecessor NCP was half-duplex.
• TCP itself also supports a half-duplex mode—even if one end sends FIN, the other end can keep transmitting as long as it wants. This was probably also a good idea, but it's certainly not the only obvious choice.
• Sequence numbers on messages or on bytes?
• Wouldn't it be useful to expose message boundaries to applications, the way 9P, SCTP, and some SNA protocols do?
• If you expose message boundaries to applications, maybe you'd also want to include a message type field? Protocol-level message-type fields have been found to be very useful in Ethernet and IP, and in a sense the port-number field in UDP is also a message-type field.
• Do you really need urgent data?
• Do servers need different port numbers? TCPMUX is a straightforward way of giving your servers port names, like in CHAOSNET, instead of port numbers. It only creates extra overhead at connection-opening time, assuming you have the moral equivalent of file descriptor passing on your OS. The only limitation is that you have to use different client ports for multiple simultaneous connections to the same server host. But in TCP everyone uses different client ports for different connections anyway. TCPMUX itself incurs an extra round-trip time delay for connection establishment, because the requested server name can't be transmitted until the client's ACK packet, but if you incorporated it into TCP, you'd put the server name in the SYN packet. If you eliminate the server port number in every TCP header, you can expand the client port number to 24 or even 32 bits.
• Alternatively, maybe network addresses should be assigned to server processes, as in Appletalk (or IP-based virtual hosting before HTTP/1.1's Host: header, or, for TLS, before SNI became widespread), rather than assigning network addresses to hosts and requiring port numbers or TCPMUX to distinguish multiple servers on the same host?
• Probably SACK was actually a good idea and should have always been the default? SACK gets a lot easier if you ack message numbers instead of byte numbers.
• Why is acknowledgement reneging allowed in TCP? That was a terrible idea.
• It turns out that measuring round-trip time is really important for retransmission, and TCP has no way of measuring RTT on retransmitted packets, which can pose real problems for correcting a ridiculously low RTT estimate, which results in excessive retransmission.
• Do you really need a PUSH bit? C'mon.
• A modest amount of overhead in the form of erasure-coding bits would permit recovery from modest amounts of packet loss without incurring retransmission timeouts, which is especially useful if your TCP-layer protocol requires a modest amount of packet loss for congestion control, as TCP does.
• Also you could use a "congestion experienced" bit instead of packet loss to detect congestion in the usual case. (TCP did eventually acquire CWR and ECE, but not for many years.)
• The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network.
• TCP's hardcoded timeout of 5 minutes is also a major flaw. Wouldn't it be better if the application could set that to 1 hour, 90 minutes, 12 hours, or a week, to handle intermittent connectivity, such as with communication satellites? Similarly for very-long-latency datagrams, such as those relayed by single LEO satellites. Together this and the previous flaw have resulted in TCP largely being replaced for its original session-management purpose with new ad-hoc protocols such as HTTP magic cookies, protocols which use TCP, if at all, merely as a reliable datagram protocol.
• Initial sequence numbers turn out not to be a very good defense against IP spoofing, because that wasn't their original purpose. Their original purpose was preventing the erroneous reception of leftover TCP segments from a previous incarnation of the connection that have been bouncing around routers ever since; this purpose would be better served by using a different client port number for each new connection. The ISN namespace is far too small for current LFNs anyway, so we had to patch over the hole in TCP with timestamps and PAWS.
• Full-duplex connections are probably a good idea, but certainly are not the only way, or the most obvious way, to create a reliable stream of data on top of an unreliable datagram layer. TCP itself also supports a half-duplex mode—even if one end sends FIN, the other end can keep transmitting as long as it wants. This was probably also a good idea, but it's certainly not the only obvious choice.
Much of that comes from the original applications being FTP and TELNET.
• Sequence numbers on messages or on bytes?
Bytes, because the whole TCP message might not fit in an IP packet. This is the MTU problem.
• Wouldn't it be useful to expose message boundaries to applications, the way 9P, SCTP, and some SNA protocols do?
Early on, there were some message-oriented, rather than stream-oriented, protocols on top of IP. Most of them died out. RDP was one such.
Another was QNet.[2]
Both still have assigned IP protocol numbers, but I doubt that a RDP packet would get very far across today's internet.
This was a lack. TCP is not a great message-oriented protocol.
• Do you really need urgent data?
The purpose of urgent data is so that when your slow Teletype is typing away, and the recipient wants it to stop, there's a way to break in. See [1], p. 8.
• It turns out that measuring round-trip time is really important for retransmission, and TCP has no way of measuring RTT on retransmitted packets, which can pose real problems for correcting a ridiculously low RTT estimate, which results in excessive retransmission.
Yes, reliable RTT is a problem.
• Do you really need a PUSH bit? C'mon.
It's another legacy thing to make TELNET work on slow links. Is it even supported any more?
• A modest amount of overhead in the form of erasure-coding bits would permit recovery from modest amounts of packet loss without incurring retransmission timeouts, which is especially useful if your TCP-layer protocol requires a modest amount of packet loss for congestion control, as TCP does.
• Also you could use a "congestion experienced" bit instead of packet loss to detect congestion in the usual case. (TCP did eventually acquire CWR and ECE, but not for many years.)
Originally, there was ICMP Source Quench for that, but Berkley didn't put it in BSD, so nobody used it. Nobody was sure when to send it or what to do when it was received.
• The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network.
That would require a security system to prevent hijacking sessions.
> The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network
This 100% !! And basically the reason mosh had to be created in the first place (and it probably wasn't easy.) Unfortunately mosh only solves the problem for ssh. Exposing fixed IP addresses to the application layer probably doesn't help either.
So annoying that TCP tends to break whenever you switch wi-fi networks or switch from wi-fi to cellular. (On iPhones at least you have MPTCP, but that requires server-side support.)
AppleTalk didn't get much love for its broadcast (or possibly multicast?) based service discovery protocol - but of course that is what inspired mDNS. I believe AppleTalk's LAN addresses were always dynamic (like 169.x IP addresses), simplifying administration and deployment.
I tend to think that one of the reasons linux containers are needed for network services is that DNS traditionally only returns an IP address (rather than address + port) so each service process needs to have its own IP address, which in linux requires a container or at least a network namespace.
AppleTalk also supported a reliable transaction (basically request-response RPC) protocol (ATP) and a session protocol, which I believe were used for Mac network services (printing, file servers, etc.) Certainly easier than serializing/deserializing byte streams.
> The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the protocol provides the message-oriented feature of the User Datagram Protocol (UDP) while ensuring reliable, in-sequence transport of messages with congestion control like the Transmission Control Protocol (TCP). Unlike UDP and TCP, the protocol supports multihoming and redundant paths to increase resilience and reliability.
[…]
> SCTP may be characterized as message-oriented, meaning it transports a sequence of messages (each being a group of bytes), rather than transporting an unbroken stream of bytes as in TCP. As in UDP, in SCTP a sender sends a message in one operation, and that exact message is passed to the receiving application process in one operation. In contrast, TCP is a stream-oriented protocol, transporting streams of bytes reliably and in order. However TCP does not allow the receiver to know how many times the sender application called on the TCP transport passing it groups of bytes to be sent out. At the sender, TCP simply appends more bytes to a queue of bytes waiting to go out over the network, rather than having to keep a queue of individual separate outbound messages which must be preserved as such.
> The term multi-streaming refers to the capability of SCTP to transmit several independent streams of chunks in parallel, for example transmitting web page images simultaneously with the web page text. In essence, it involves bundling several connections into a single SCTP association, operating on messages (or chunks) rather than bytes.
Wait, can you actually just use IP? Can I just make up a packet and send it to a host across the Internet? I'd think that all the intermediate routers would want to have an opinion about my packet, caring, at the very least, that it's either TCP or UDP.
Core routers don't inspect that field, NAT/ISP boxes can. I believe that with two suitable dedicated linux servers it is very possible to send and receive single custom IP packet between them even using 253 or 254 (= Use for experimentation and testing [RFC3692]) as the protocol number
This is an interesting list; it makes you appreciate just how many obscure protocols have died out in practice. Evolution in networks seems to mimic evolution in nature quite well.
> caring, at the very least, that it's either TCP or UDP.
You left out ICMP, my favourite! (And a lot more important in IPv6 than in v4.)
Another pretty well known protocol that is neither TCP nor UDP is IPsec. (Which is really two new IP protocols.) People really did design proper IP protocols still in the 90s.
> Can I just make up a packet and send it to a host across the Internet?
You should be able to. But if you are on a corporate network with a really strict firewalling router that only forwards traffic it likes, then likely not. There are also really crappy home routers which gives similar problems from the other end of enterpriseness.
NAT also destroyed much of the end-to-end principle. If you don't have a real IP address and relies on a NAT router to forward your data, it needs to be in a protocol the router recognizes.
Anyway, for the past two decades people have grown tired of that and just piles hacks on top of TCP or UDP instead. That's sad. Or who am I kidding? Really it's on top of HTTP. HTTP will likely live on long past anything IP.
There is little point in inventing new protocols, given how low the overhead of UDP is. That's just 8 bytes per packet, and it enables going through NAT. Why come up with a new transport layer protocol, when you can just use UDP framing?
> NAT also destroyed much of the end-to-end principle. If you don't have a real IP address and relies on a NAT router to forward your data, it needs to be in a protocol the router recognizes.
Not necessarily. Many protocols can survive being NATed if they don't carry IP/port related information inside their payload. FTP is a famous counterexample - it uses a control channel (TCP21) which contains commands to open data channels (TCP20), and those commands specify IP:port pairs, so, depending on the protocol, a NAT router has to rewrite them and/or open ports dynamically and/or create NAT entries on the fly. A lot of other stuff has no need for that and will happily go through without any rewriting.
If there's no form of NAT or transport later processing along your path between endpoints you shouldn't have an issue. But NAT and transport and application layer load balancing are very common on the net these days so YMMV.
> I'd think that all the intermediate routers would want to have an opinion about my packet, caring, at the very least, that it's either TCP or UDP.
They absolutely don't. Routers are layer 3 devices; TCP & UDP are layer 4. The only impact is that the ECMP flow hashes will have less entropy, but that's purely an optimization thing.
Note TCP, UDP and ICMP are nowhere near all the protocols you'll commonly see on the internet — at minimum, SCTP, GRE, L2TP and ESP are reasonably widespread (even a tiny fraction of traffic is still a giant number considering internet scales).
You can send whatever protocol number with whatever contents your heart desires. Whether the other end will do anything useful with it is another question.
> They absolutely don't. Routers are layer 3 devices;
Idealized routers are, yes.
Actual IP paths these days usually involve at least one NAT, and these will absolutely throw away anything other than TCP, UDP, and if you're lucky ICMP.
As far as I'm aware, sure you can. TCP packets and UDP datagrams are wrapped in IP datagrams, and it's the job of an IP network to ship your data from point A (sender) to point B (receiver). Nodes along the way might do so-called "deep packet inspection" to snoop on the payload of your IP datagrams (for various reasons, not all nefarious), but they don't need to do that to do the basic job of routing. From a semantic standpoint, the information in the TCP and UDP headers (as part of the IP payload) is only there to govern interactions between the two endpoint parties. (For instance, the "port" of a TCP or UDP packet is a node-local identifier for one of many services that might exist at the IP address the packet was routed to, allowing many services to coexist at the same node.)
Hmm, I thought intermediate routers use the TCP packet's bits for congestion control, no? Though I guess they can probably just use the destination IP for that.
Yep it's full of IP protocols other than the well-known TCP, UDP and ICMP (and, if you ever had the displeasure of learning IPSEC, its AH and ESP).
A bunch of multicast stuff (IGMP, PIM)
A few routing protocols (OSPF, but notably not BGP which just uses TCP, and (usually) not MPLS which just goes over the wire - it sits at the same layer as IP and not above it)
A few VPN/encapsulation solutions like GRE, IP-in-IP, L2TP and probably others I can't remember
The reason you wouldn't do that is IP doesn't give you a mechanism to share an IP address with multiple processes on a host, it just gets your packets to a particular host.
As soon as you start thinking about having multiple services on a host you end up with the idea of having a service id or "port"
UDP or UDP Lite gives you exactly that at the cost of 8 bytes, so there's no real value in not just putting everything on top of UDP
They shouldn't; the whole point is that the IP header is enough to route packets between endpoints, and only the endpoints should care about any higher layer protocols. But unfortunately some routers do, and if you have NAT then the NAT device needs to examine the TCP or UDP header to know how to forward those packets.
That would be IP over some lower level physical layer, not some custom content stuffed into an IP packet :)
(It's absolutely worth reading some of those old April Fools' RFCs, by the way [0]. I'm a big fan of RFC 7168, which introduced HTTP response code 418 "I'm a teapot".)
TCP being the “default” meant it was chosen when the need for ordering and uniform reliability wasn’t there. That was fine but left systems working less well than they could have with more carefully chosen underpinnings. With HTTP/3 gaining traction, and HTTP being the “next level up default choice” things potentially get better. The issue I see is that QUIC is far more complex, and the new power is fantastic for a few but irrelevant to most.
UDP has its place as well, and if we have more simple and effective solutions like WireGuard’s handshake and encryption on top of it we’d be better off as an industry.
The congestion control algorithm in TCP has some interesting effects on throughput that a lot of developers aren’t aware of.
For example, sending some data on a fresh TCP connection is slow, and the “ramp up time” to the bandwidth of the network is almost entirely determined by the latency.
Amazing speed ups can be achieved in a data centre network by shaving microseconds off the round trip time!
Similarly, many (all?) TCP stacks count segments, not bytes, when determining this ramp up rate. This means that jumbo frames can provide 6x the bandwidth during this period!
If you read about the network design of AWS, they put a lot of effort into low switching latency and enabling jumbo frames.
The real pros do this kind of network tuning, everyone else wonders why they don’t get anywhere near 10 Gbps through a 10 Gbps link.
No. TCP likes zero packet loss (connected), and it understands 100% packet loss (disconnected). Its weakness is scenarios (semiconnected) in which packet loss is constantly fluctuating between substantial and nearly-total. It doesn't know what is going on, and it may cope or it may not, because its designers did not envision a future in which most networks have a semiconnected last mile; but that is where we are. Without things like forward error correction, TCP would be nearly useless over wireless. It is interesting to envision a layer-4 protocol that would incorporate FEC-like capabilities.
if you went back to 1981 and said 'yeah, this is great. but what we really want to do is not have an internet, but kind of a piecewise internet. instead of a global address we'll use addresses that have a narrower scope. and naturally as consequence of this we'll need to start distinguishing between nodes that everyone can reach, service nodes, and nodes that no one can reach - client nodes. and as a consequence of this we'll start building links that are asymmetric in bandwidth, since one direction is only used for requests and acks and not any data volume.'
they would have looked at you and asked straight out what you hoped to gain by making these things distinguished, because it certainly complicates things.
Its trivial to develop your own protocols on top of IP.
It was trivial like 15 years ago in python (without any libraries) just handcrafted packets (arp, ip etc).
> The internet is incredible. It’s nearly impossible to keep people away from.
Well ... he seems very motivated. I am more skeptical.
For instance, Google via chrome controls a lot of the internet, even more so via its search engine, AI, youtube and so forth.
Even aside from this people's habits changed. In the 1990s everyone and their Grandma had a website. Nowadays ... it is a bit different. We suddenly have horrible blogging sites such as medium.com, pestering people with popups. Of course we also had popups in the 1990s, but the diversity was simply higher. Everything today is much more streamlined it seems. And top-down controlled. Look at Twitter, owned by a greedy and selfish billionaire. And the US president? Super-selfish too. We lost something here in the last some 25 years.
You're talking about the web, which is merely an app with the internet as its platform. We can scrap it and still use the internet to build a different one.
It’s worth considering how the tiny computers of the era forced a simple clean design. IPv6 was designed starting in the early 90s and they couldn’t resist loading it up with extensions, though the core protocol remains fine and is just IP with more bits. (Many of the extensions are rarely if ever used.)
If the net were designed today it would be some complicated monstrosity where every packet was reminiscent of X.509 in terms of arcane complexity. It might even have JSON in it. It would be incredibly high overhead and we’d see tons of articles about how someone made it fast by leveraging CPU vector instructions or a GPU to parse it.
This is called Eroom’s law, or Moore’s law backwards, and it is very real. Bigger machines let programmers and designers loose to indulge their desire to make things complicated.
IPSec was a big one that’s now borderline obsolete, though it is still used for VPNs and was back ported to IPv4.
Many networking folks including myself consider IPv6 router advertisements and SLAAC to be inferior, in practice, to DHCPv6, and that it would be better if we’d just left IP assignment out of the spec like it was in V4. Right now we have this mess where a lot of nets prefer or require DHCPv6 but some vendors, like apparently Android, refuse to support it.
The rules about how V6 addresses are chopped up and assigned are wasteful and dumb. The entire V4 space could have been mapped onto /32 and an encapsulation protocol made to allow V4 to carry V6, providing a seamless upgrade path that does not require full upgrade of the whole core, but that would have been too logical. Every machine should get like a /96 so it can use 32 bits of space to address apps, VMs, containers, etc. As it stands we waste 64 bits of the space to make SLAAC possible, as near as I can tell. The SLAAC tail must have wagged the dog in that people thought this feature was cool enough to waste 8 bytes per packet.
The V6 header allows extension bits that are never used and blocked by most firewalls. There’s really no point in them existing since middle boxes effectively freeze the base protocol in stone.
Those are some of the big ones.
Basically all they should have done was make IPs 64 or 128 bits and left everything else alone. But I think there was a committee.
As it stands we have what we have and we should just treat V6 as IP128 and ignore the rest. I’m still in favor of the upgrade. V4 is too small, full stop. If we don’t enlarge the addresses we will completely lose end to end connectivity as a supported feature of the network.
I can easily spot it's an AI written article, because it actually explains the technology in understandable human language. A human would have written it the way it was either presented to them in university or in bloated IT books: absolutely useless.
I can easily spot it's an AI written comment, because it actually explains their idea in understandable human language and brings nothing to the discussion. A human would have written it the way they understand it and bring their opinions along: absolutely useless.
At first wanted to give the benefit of the doubt that this is sarcasm but gave a skim through history and I guess it's just a committed anti-AI agenda.
Personally I found the tone of the article quite genuine and the video at the end made a compelling case for it. Well I figure you commented having actually read it.
Edit: I can't downvote but if I could it probably would have been better than this comment!
If you start with the problem of how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP. It just is the right solution for the job.
The three drawbacks of the original TCP algorithm were the window size (the maximum value is just too small for today's speeds), poor handling of missing packets (addressed by extensions such as selective-ACK), and the fact that it only manages one stream at a time, and some applications want multiple streams that don't block each other. You could use multiple TCP connections, but that adds its own overhead, so SCTP and QUIC were designed to address those issues.
The congestion control algorithm is not part of the on-the-wire protocol, it's just some code on each side of the connection that decides when to (re)send packets to make the best use of the available bandwidth. Anything that implements a reliable stream on top of datagrams needs to implement such an algorithm. The original ones (Reno, Vegas, etc) were very simple but already did a good job, although back then network equipment didn't have large buffers. A lot of research is going into making better algorithms that handle large buffers, large roundtrip times, varying bandwidth needs and also being fair when multiple connections share the same bandwidth.
it only manages one stream at a time
I'll take flak for saying it, but I feel web developers are partially at fault for laziness on this one. I've often seen them trigger a swath of connections (e.g. for uncoordinated async events), when carefully managed multiplexing over one or a handful will do just fine.
Eg. In prehistoric times I wrote a JavaScript library that let you queue up several downloads over one stream, with control over prioritization and cancelability.
It was used in a GreaseMonkey script on a popular dating website, to fetch thumbnails and other details of all your matches in the background. Hovering over a match would bring up all their photos, and if some hadn't been retrieved yet they'd immediately move to the top of the queue. I intentionally wanted to limit the number of connections, to avoid oversaturating the server or the user's bandwidth. Idle time was used to prefetch all matches on the page (IIRC in a sensible order responsive to your scroll location). If you picked a large enough pagination, then stepped away to top up your coffee, by the time you got back you could browse through all of your recent matches instantly, without waiting for any server roundtrip lag.
It was pretty slick. I realize these days modern stacks give you multiplexing for free, but to put in context this was created in the era before even JQuery was well-known.
Funny story, I shared it with one of my matches and she found it super useful but was a bit surprised that, in a way, I was helping my competition. Turned out OK... we're still together nearly two decades later and now she generously jokes I invented Tinder before it was a thing.
Sure, you can reimplement multiplexing on the application level, but it just makes more sense to do it on the transport level, so that people don't have to do it in JavaScript.
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This is wonderful to hear. I have a naive question. Is this the reason most websites/web servers absolutely need CDNs (apart from their edge capabilities) because they understand caching much more than a web developer does? But I would think the person more closer to the user access pattern would know the optimal caching strategy.
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[Not a web dev but] I thought each site gets a handful of connections (4) to each host and more requests would have to wait to use one of them. That's pretty close to what I'd want with a reasonably fast connection.
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> If you start with the problem of how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP.
I'll add that at the time of TCP's writing, the telephone people far outnumbered everyone else in the packet switching vs circuit switching debate. TCP gives you a virtual circuit over a packet switched network as a pair of reliable-enough independent byte streams over IP. This idea, that the endpoints could implement reliability through retransmission came from an earlier French network, Cylades, and ends up being a core principle of IP networks.
We're still "suffering" from the latency and jitter effects of the packet switching victory. (The debate happened before my time and I don't know if I would have really agreed with circuit switching.) Latency and jitter on the modern Internet are very best effort emphasis on "effort".
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The telephone people were basically right with their criticisms of TCP/IP such as:
What about QoS? Jitter, bandwidth, latency, fairness guarantees? What about queuing delay? What about multiplexing and tunneling? Traffic shaping and engineering? What about long-haul performance? Easy integration with optical circuit networks? etc. ATM addressed these issues, but TCP/IP did not.
All of these things showed up again once you tried to do VOIP and video conferencing, and in core ISPs as well as access networks, and they weren't (and in many cases still aren't) easy to solve.
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TCP has another unfixable flaw - it cannot be properly secured. Writing a security layer on top of TCP can at most detect, not avoid, attacks.
It is very easy for a malicious actor anywhere in the network to inject data into a connection. By contrast, it is much harder for a malicious actor to break the legitimate traffic flow ... except for the fact that TCP RST grants any rando the power to upgrade "inject" to "break". This is quite common in the wild for any traffic that does not look like HTTP, even when both endpoints are perfectly healthy.
Blocking TCP RST packets using your firewall will significantly improve reliability, but this still does not project you from more advanced attackers which cause a desynchronization due to forged sequence numbers with nonempty payload.
As a result, it is mandatory for every application to support a full-blown "resume on a separate connection" operation, which is complicated and hairy and also immediately runs into the additional flaw that TCP is very slow to start.
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While not an outright flaw, I also think it has become clear by now that it is highly suboptimal for "address" and "port" to be separate notions.
"... some applications want multiple streams that don't block each other. You could use multiple TCP connections, but that adds its own overhead, so SCTP and QUIC were designed to address those issues."
Other applications work just fine with a single TCP connection
If I am using TCP for DNS, for example, and I am retrieving data from a single host such as a DNS cache, I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order. No blocking.^1 If the cache (application) supports it, this is much faster than receiving answers sequentially and it's more efficient and polite than opening multiple TCP connections
1. I do this every day outside the browser with DNS over TLS (DoT) using something like streamtcp from NLNet Labs. I'm not sure that QUIC is faster, server support for QUIC is much more limited, but QUIC may have other advantages
I also do it with DNS over HTTPS (DoH), outside the browser, using HTTP/1.1 pipelining, but there I receive answers sequentially. I'm still not convinced that HTTP/2 is faster for this particular use case, i.e., downloading data from a single host using multiple HTTP requests (compared to something like integrating online advertising into websites, for example)
> I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order.
This is because DoT allows the DNS server to resolve queries concurrently and send query responses out of order.
However, this is an application layer feature, not a transport layer one. The underlying TCP packets still have to arrive in order and therefore are subject to blocking.
> I can send multiple queries over a single TCP connection and receive multiple responses over the same single TCP single connection, out of order. No blocking.
You're missing the point. You have one TCP connection, and the sever sends you response1 and then response2. Now if response1 gets lost or delayed due to network conditions, you must wait for response1 to be retransmitted before you can read response2. That is blocking, no way around it. It has nothing to do with advertising(?), and the other protocols mentioned don't have this drawback.
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> how to create a reliable stream of data on top of an unreliable datagram layer, then the solution that comes out will look virtually identical to TCP. It just is the right solution for the job
A stream of bytes made sense in the 1970s for remote terminal emulation. It still sort of makes sense for email, where a partial message is useful (though downloading headers in bulk followed by full message on demand probably makes more sense.)
But in 2025 much of communication involves messages that aren't useful if you only get part of them. It's also a pain to have to serialize messages into a byte stream and then deserialize the byte stream into messages (see: gRPC etc.) and the byte stream ordering is costly, doesn't work well with multipathing, and doesn't provide much benefit if you are only delivering complete messages.
TCP without congestion control isn't particularly useful. As you note traditional TCP congestion control doesn't respond well to reordering. Also TCP's congestion control traditionally doesn't distinguish between intentional packet drops (e.g. due to buffer overflow) and packet loss (e.g. due to corruption.) This means, for example that it can't be used directly over networks with wireless links (which is why wi-fi has its own link layer retransmission).
TCP's traditional congestion control is designed to fill buffers up until packets are dropped, leading to undesirable buffer bloat issues.
TCP's traditional congestion control algorithms (additive increase/multiplicative decrease on drop) also have the poor property that your data rate tends to drop as RTT increases.
TCP wasn't designed for hardware offload, which can lead to software bottlenecks and/or increased complexity when you do try to offload it to hardware.
TCP's three-way handshake is costly for one-shot RPCs, and slow start means that short flows may never make it out of slow start, neutralizing benefits from high-speed networks.
TCP is also poor for mobility. A connection breaks when your IP address changes, and there is no easy way to migrate it. Most TCP APIs expose IP addresses at the application layer, which causes additional brittleness.
Additionally, TCP is poorly suited for optical/WDM networks, which support dedicated bandwidth (signal/channel bandwidth as well as data rate), and are becoming more important in datacenters and as interconnects for GPU clusters.
etc.
Yeah the fact that the congestion control algorithm isn’t part of the wire protocol is very ahead of its time and gave the protocol flexibility that’s much needed in retrospective. OTOH a lot of college courses about TCP don’t really emphasize this fact and still many people I interacted with thought that TCP had a single defined congestion control algorithm.
> If you start with the problem of how to create a reliable stream of data on top of an unreliable datagram layer
> poor handling of missing packets
so it was poor at exact thing it was designed for?
Poor for high speed connections () or very unreliable connections.
) compared to when TCP was invented.
When I started at university the ftp speed from the US during daytime was 500 bytes per second! You don't have many unacknowledged packages in such a connection.
Back then even a 1 megabits/sec connection was super high speed and very expensive.
It was a trade off at the time. Selective acknowledgments require more resources.
Might be obvious in hindsight, but it was not clear at all back then, that the congestion is manageable this way. There were legitimate concerns that it will all just melt down.
There are a lot of design alternatives possible to TCP within the "create a reliable stream of data on top of an unreliable datagram layer" space:
• Full-duplex connections are probably a good idea, but certainly are not the only way, or the most obvious way, to create a reliable stream of data on top of an unreliable datagram layer. TCP's predecessor NCP was half-duplex.
• TCP itself also supports a half-duplex mode—even if one end sends FIN, the other end can keep transmitting as long as it wants. This was probably also a good idea, but it's certainly not the only obvious choice.
• Sequence numbers on messages or on bytes?
• Wouldn't it be useful to expose message boundaries to applications, the way 9P, SCTP, and some SNA protocols do?
• If you expose message boundaries to applications, maybe you'd also want to include a message type field? Protocol-level message-type fields have been found to be very useful in Ethernet and IP, and in a sense the port-number field in UDP is also a message-type field.
• Do you really need urgent data?
• Do servers need different port numbers? TCPMUX is a straightforward way of giving your servers port names, like in CHAOSNET, instead of port numbers. It only creates extra overhead at connection-opening time, assuming you have the moral equivalent of file descriptor passing on your OS. The only limitation is that you have to use different client ports for multiple simultaneous connections to the same server host. But in TCP everyone uses different client ports for different connections anyway. TCPMUX itself incurs an extra round-trip time delay for connection establishment, because the requested server name can't be transmitted until the client's ACK packet, but if you incorporated it into TCP, you'd put the server name in the SYN packet. If you eliminate the server port number in every TCP header, you can expand the client port number to 24 or even 32 bits.
• Alternatively, maybe network addresses should be assigned to server processes, as in Appletalk (or IP-based virtual hosting before HTTP/1.1's Host: header, or, for TLS, before SNI became widespread), rather than assigning network addresses to hosts and requiring port numbers or TCPMUX to distinguish multiple servers on the same host?
• Probably SACK was actually a good idea and should have always been the default? SACK gets a lot easier if you ack message numbers instead of byte numbers.
• Why is acknowledgement reneging allowed in TCP? That was a terrible idea.
• It turns out that measuring round-trip time is really important for retransmission, and TCP has no way of measuring RTT on retransmitted packets, which can pose real problems for correcting a ridiculously low RTT estimate, which results in excessive retransmission.
• Do you really need a PUSH bit? C'mon.
• A modest amount of overhead in the form of erasure-coding bits would permit recovery from modest amounts of packet loss without incurring retransmission timeouts, which is especially useful if your TCP-layer protocol requires a modest amount of packet loss for congestion control, as TCP does.
• Also you could use a "congestion experienced" bit instead of packet loss to detect congestion in the usual case. (TCP did eventually acquire CWR and ECE, but not for many years.)
• The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network.
• TCP's hardcoded timeout of 5 minutes is also a major flaw. Wouldn't it be better if the application could set that to 1 hour, 90 minutes, 12 hours, or a week, to handle intermittent connectivity, such as with communication satellites? Similarly for very-long-latency datagrams, such as those relayed by single LEO satellites. Together this and the previous flaw have resulted in TCP largely being replaced for its original session-management purpose with new ad-hoc protocols such as HTTP magic cookies, protocols which use TCP, if at all, merely as a reliable datagram protocol.
• Initial sequence numbers turn out not to be a very good defense against IP spoofing, because that wasn't their original purpose. Their original purpose was preventing the erroneous reception of leftover TCP segments from a previous incarnation of the connection that have been bouncing around routers ever since; this purpose would be better served by using a different client port number for each new connection. The ISN namespace is far too small for current LFNs anyway, so we had to patch over the hole in TCP with timestamps and PAWS.
• Full-duplex connections are probably a good idea, but certainly are not the only way, or the most obvious way, to create a reliable stream of data on top of an unreliable datagram layer. TCP itself also supports a half-duplex mode—even if one end sends FIN, the other end can keep transmitting as long as it wants. This was probably also a good idea, but it's certainly not the only obvious choice.
Much of that comes from the original applications being FTP and TELNET.
• Sequence numbers on messages or on bytes?
Bytes, because the whole TCP message might not fit in an IP packet. This is the MTU problem.
• Wouldn't it be useful to expose message boundaries to applications, the way 9P, SCTP, and some SNA protocols do?
Early on, there were some message-oriented, rather than stream-oriented, protocols on top of IP. Most of them died out. RDP was one such. Another was QNet.[2] Both still have assigned IP protocol numbers, but I doubt that a RDP packet would get very far across today's internet.
This was a lack. TCP is not a great message-oriented protocol.
• Do you really need urgent data?
The purpose of urgent data is so that when your slow Teletype is typing away, and the recipient wants it to stop, there's a way to break in. See [1], p. 8.
• It turns out that measuring round-trip time is really important for retransmission, and TCP has no way of measuring RTT on retransmitted packets, which can pose real problems for correcting a ridiculously low RTT estimate, which results in excessive retransmission.
Yes, reliable RTT is a problem.
• Do you really need a PUSH bit? C'mon.
It's another legacy thing to make TELNET work on slow links. Is it even supported any more?
• A modest amount of overhead in the form of erasure-coding bits would permit recovery from modest amounts of packet loss without incurring retransmission timeouts, which is especially useful if your TCP-layer protocol requires a modest amount of packet loss for congestion control, as TCP does.
• Also you could use a "congestion experienced" bit instead of packet loss to detect congestion in the usual case. (TCP did eventually acquire CWR and ECE, but not for many years.)
Originally, there was ICMP Source Quench for that, but Berkley didn't put it in BSD, so nobody used it. Nobody was sure when to send it or what to do when it was received.
• The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network.
That would require a security system to prevent hijacking sessions.
[1] https://archive.org/stream/rfc854/rfc854.txt_djvu.txt
[2] https://en.wikipedia.org/wiki/List_of_IP_protocol_numbers
> The fact that you can't resume a TCP connection from a different IP address, the way you can with a Mosh connection, is a serious flaw that seriously impedes nodes from moving around the network
This 100% !! And basically the reason mosh had to be created in the first place (and it probably wasn't easy.) Unfortunately mosh only solves the problem for ssh. Exposing fixed IP addresses to the application layer probably doesn't help either.
So annoying that TCP tends to break whenever you switch wi-fi networks or switch from wi-fi to cellular. (On iPhones at least you have MPTCP, but that requires server-side support.)
AppleTalk didn't get much love for its broadcast (or possibly multicast?) based service discovery protocol - but of course that is what inspired mDNS. I believe AppleTalk's LAN addresses were always dynamic (like 169.x IP addresses), simplifying administration and deployment.
I tend to think that one of the reasons linux containers are needed for network services is that DNS traditionally only returns an IP address (rather than address + port) so each service process needs to have its own IP address, which in linux requires a container or at least a network namespace.
AppleTalk also supported a reliable transaction (basically request-response RPC) protocol (ATP) and a session protocol, which I believe were used for Mac network services (printing, file servers, etc.) Certainly easier than serializing/deserializing byte streams.
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I was excited about SCTP over 10 years ago but getting it to work was hard.
The Linux kernel supports it but at least when I had tried this those modules were disabled on most distros.
Any love for SCTP?
> The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the protocol provides the message-oriented feature of the User Datagram Protocol (UDP) while ensuring reliable, in-sequence transport of messages with congestion control like the Transmission Control Protocol (TCP). Unlike UDP and TCP, the protocol supports multihoming and redundant paths to increase resilience and reliability.
[…]
> SCTP may be characterized as message-oriented, meaning it transports a sequence of messages (each being a group of bytes), rather than transporting an unbroken stream of bytes as in TCP. As in UDP, in SCTP a sender sends a message in one operation, and that exact message is passed to the receiving application process in one operation. In contrast, TCP is a stream-oriented protocol, transporting streams of bytes reliably and in order. However TCP does not allow the receiver to know how many times the sender application called on the TCP transport passing it groups of bytes to be sent out. At the sender, TCP simply appends more bytes to a queue of bytes waiting to go out over the network, rather than having to keep a queue of individual separate outbound messages which must be preserved as such.
> The term multi-streaming refers to the capability of SCTP to transmit several independent streams of chunks in parallel, for example transmitting web page images simultaneously with the web page text. In essence, it involves bundling several connections into a single SCTP association, operating on messages (or chunks) rather than bytes.
* https://en.wikipedia.org/wiki/Stream_Control_Transmission_Pr...
No, SCTP only fixes half of a problem, but also gratuitously introduces several additional flaws, even ignoring the "router support" problem.
The only good answer is "a reliability layer on top of UDP"; fortunately everybody is now rallying around QUIC as the choice for that.
As a BSD enjoyer and paid to write Erlang, I have nothing but love for SCTP.
Wait, can you actually just use IP? Can I just make up a packet and send it to a host across the Internet? I'd think that all the intermediate routers would want to have an opinion about my packet, caring, at the very least, that it's either TCP or UDP.
You can definitely craft an IP packet by hand and send it. If it's IPv4, you need to put a number between 0 and 255 to the protocol field from this list: https://www.iana.org/assignments/protocol-numbers/protocol-n...
Core routers don't inspect that field, NAT/ISP boxes can. I believe that with two suitable dedicated linux servers it is very possible to send and receive single custom IP packet between them even using 253 or 254 (= Use for experimentation and testing [RFC3692]) as the protocol number
> If it's IPv4, you need to put a number between 0 and 255 to the protocol field from this list:
To save a skim (though it's an interesting list!), protocol codes 253 and 254 are suitable "for experimentation and testing".
Playing with protocol number change usually results in “Protocol Unreachable” or “Malformed Packet” from your OS.
What happens when the remaining 104 unassigned protocol numbers are exhausted?
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This is an interesting list; it makes you appreciate just how many obscure protocols have died out in practice. Evolution in networks seems to mimic evolution in nature quite well.
Very interesting, thanks!
> caring, at the very least, that it's either TCP or UDP.
You left out ICMP, my favourite! (And a lot more important in IPv6 than in v4.)
Another pretty well known protocol that is neither TCP nor UDP is IPsec. (Which is really two new IP protocols.) People really did design proper IP protocols still in the 90s.
> Can I just make up a packet and send it to a host across the Internet?
You should be able to. But if you are on a corporate network with a really strict firewalling router that only forwards traffic it likes, then likely not. There are also really crappy home routers which gives similar problems from the other end of enterpriseness.
NAT also destroyed much of the end-to-end principle. If you don't have a real IP address and relies on a NAT router to forward your data, it needs to be in a protocol the router recognizes.
Anyway, for the past two decades people have grown tired of that and just piles hacks on top of TCP or UDP instead. That's sad. Or who am I kidding? Really it's on top of HTTP. HTTP will likely live on long past anything IP.
There is little point in inventing new protocols, given how low the overhead of UDP is. That's just 8 bytes per packet, and it enables going through NAT. Why come up with a new transport layer protocol, when you can just use UDP framing?
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> NAT also destroyed much of the end-to-end principle. If you don't have a real IP address and relies on a NAT router to forward your data, it needs to be in a protocol the router recognizes.
Not necessarily. Many protocols can survive being NATed if they don't carry IP/port related information inside their payload. FTP is a famous counterexample - it uses a control channel (TCP21) which contains commands to open data channels (TCP20), and those commands specify IP:port pairs, so, depending on the protocol, a NAT router has to rewrite them and/or open ports dynamically and/or create NAT entries on the fly. A lot of other stuff has no need for that and will happily go through without any rewriting.
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> You left out ICMP, my favourite!
Even ICMP has a hard time traversing NATs and firewalls these days, for largely bad reasons. Try pinging anything in AWS, for example...
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If there's no form of NAT or transport later processing along your path between endpoints you shouldn't have an issue. But NAT and transport and application layer load balancing are very common on the net these days so YMMV.
You might have more luck with an IPv6 packet.
> I'd think that all the intermediate routers would want to have an opinion about my packet, caring, at the very least, that it's either TCP or UDP.
They absolutely don't. Routers are layer 3 devices; TCP & UDP are layer 4. The only impact is that the ECMP flow hashes will have less entropy, but that's purely an optimization thing.
Note TCP, UDP and ICMP are nowhere near all the protocols you'll commonly see on the internet — at minimum, SCTP, GRE, L2TP and ESP are reasonably widespread (even a tiny fraction of traffic is still a giant number considering internet scales).
You can send whatever protocol number with whatever contents your heart desires. Whether the other end will do anything useful with it is another question.
> They absolutely don't. Routers are layer 3 devices;
Idealized routers are, yes.
Actual IP paths these days usually involve at least one NAT, and these will absolutely throw away anything other than TCP, UDP, and if you're lucky ICMP.
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As far as I'm aware, sure you can. TCP packets and UDP datagrams are wrapped in IP datagrams, and it's the job of an IP network to ship your data from point A (sender) to point B (receiver). Nodes along the way might do so-called "deep packet inspection" to snoop on the payload of your IP datagrams (for various reasons, not all nefarious), but they don't need to do that to do the basic job of routing. From a semantic standpoint, the information in the TCP and UDP headers (as part of the IP payload) is only there to govern interactions between the two endpoint parties. (For instance, the "port" of a TCP or UDP packet is a node-local identifier for one of many services that might exist at the IP address the packet was routed to, allowing many services to coexist at the same node.)
Huh. So it's literally "TCP over IP" like the name suggests.
Hmm, I thought intermediate routers use the TCP packet's bits for congestion control, no? Though I guess they can probably just use the destination IP for that.
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Yep it's full of IP protocols other than the well-known TCP, UDP and ICMP (and, if you ever had the displeasure of learning IPSEC, its AH and ESP).
A bunch of multicast stuff (IGMP, PIM)
A few routing protocols (OSPF, but notably not BGP which just uses TCP, and (usually) not MPLS which just goes over the wire - it sits at the same layer as IP and not above it)
A few VPN/encapsulation solutions like GRE, IP-in-IP, L2TP and probably others I can't remember
As usual, Wikipedia has got you covered, much better than my own recollection: https://en.wikipedia.org/wiki/List_of_IP_protocol_numbers
To GPs point, though, most of these will unfortunately be dropped by most middleboxes for various reasons.
Behind a NA(P)T, you can obviously only use those protocols that the translator knows how to remap ports for.
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The reason you wouldn't do that is IP doesn't give you a mechanism to share an IP address with multiple processes on a host, it just gets your packets to a particular host.
As soon as you start thinking about having multiple services on a host you end up with the idea of having a service id or "port"
UDP or UDP Lite gives you exactly that at the cost of 8 bytes, so there's no real value in not just putting everything on top of UDP
You know I've always wondered if you could run Kermit*-over-IP, without having TCP inbetween.
*The protocol.
They shouldn't; the whole point is that the IP header is enough to route packets between endpoints, and only the endpoints should care about any higher layer protocols. But unfortunately some routers do, and if you have NAT then the NAT device needs to examine the TCP or UDP header to know how to forward those packets.
Notably, QUIC (and thus HTTP/3) uses UDP instead of a new protocol number for this reason.
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Can also NAT using IP protocol.
Probably not, loads of routers are even blocking parts of ICMP.
That's firewalls (or others), not routers. If it blocks things, it's by definition not a router anymore.
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something like this?
https://en.wikipedia.org/wiki/IP_over_Avian_Carriers
That would be IP over some lower level physical layer, not some custom content stuffed into an IP packet :)
(It's absolutely worth reading some of those old April Fools' RFCs, by the way [0]. I'm a big fan of RFC 7168, which introduced HTTP response code 418 "I'm a teapot".)
[0]: https://en.wikipedia.org/wiki/April_Fools%27_Day_Request_for...
Yes but not if you or they are behind NAT. It's a shame port numbers aren't in IP.
RUDP from Plan9 was a nice step between TCP and UDP - https://en.wikipedia.org/wiki/Reliable_User_Datagram_Protoco...
TCP being the “default” meant it was chosen when the need for ordering and uniform reliability wasn’t there. That was fine but left systems working less well than they could have with more carefully chosen underpinnings. With HTTP/3 gaining traction, and HTTP being the “next level up default choice” things potentially get better. The issue I see is that QUIC is far more complex, and the new power is fantastic for a few but irrelevant to most.
UDP has its place as well, and if we have more simple and effective solutions like WireGuard’s handshake and encryption on top of it we’d be better off as an industry.
https://news.ycombinator.com/newsguidelines.html
Strange, the title was definitely the original title earlier today
The congestion control algorithm in TCP has some interesting effects on throughput that a lot of developers aren’t aware of.
For example, sending some data on a fresh TCP connection is slow, and the “ramp up time” to the bandwidth of the network is almost entirely determined by the latency.
Amazing speed ups can be achieved in a data centre network by shaving microseconds off the round trip time!
Similarly, many (all?) TCP stacks count segments, not bytes, when determining this ramp up rate. This means that jumbo frames can provide 6x the bandwidth during this period!
If you read about the network design of AWS, they put a lot of effort into low switching latency and enabling jumbo frames.
The real pros do this kind of network tuning, everyone else wonders why they don’t get anywhere near 10 Gbps through a 10 Gbps link.
Crap, that thing, the code etc... has been posted thousand of times on the internet. The final quote "Oh I am so happy this works", ok thanks bye.
TCP is one of the great works of the human mind, but it did not envision the dominance of semiconnected networks.
Are you referring to NAT?
No. TCP likes zero packet loss (connected), and it understands 100% packet loss (disconnected). Its weakness is scenarios (semiconnected) in which packet loss is constantly fluctuating between substantial and nearly-total. It doesn't know what is going on, and it may cope or it may not, because its designers did not envision a future in which most networks have a semiconnected last mile; but that is where we are. Without things like forward error correction, TCP would be nearly useless over wireless. It is interesting to envision a layer-4 protocol that would incorporate FEC-like capabilities.
if you went back to 1981 and said 'yeah, this is great. but what we really want to do is not have an internet, but kind of a piecewise internet. instead of a global address we'll use addresses that have a narrower scope. and naturally as consequence of this we'll need to start distinguishing between nodes that everyone can reach, service nodes, and nodes that no one can reach - client nodes. and as a consequence of this we'll start building links that are asymmetric in bandwidth, since one direction is only used for requests and acks and not any data volume.'
they would have looked at you and asked straight out what you hoped to gain by making these things distinguished, because it certainly complicates things.
Wireless networks are always going to have asymmetries of transmit power. Everything flows from that. ALOHAnet was 1971.
Its trivial to develop your own protocols on top of IP. It was trivial like 15 years ago in python (without any libraries) just handcrafted packets (arp, ip etc).
> The internet is incredible. It’s nearly impossible to keep people away from.
Well ... he seems very motivated. I am more skeptical.
For instance, Google via chrome controls a lot of the internet, even more so via its search engine, AI, youtube and so forth.
Even aside from this people's habits changed. In the 1990s everyone and their Grandma had a website. Nowadays ... it is a bit different. We suddenly have horrible blogging sites such as medium.com, pestering people with popups. Of course we also had popups in the 1990s, but the diversity was simply higher. Everything today is much more streamlined it seems. And top-down controlled. Look at Twitter, owned by a greedy and selfish billionaire. And the US president? Super-selfish too. We lost something here in the last some 25 years.
You're talking about the web, which is merely an app with the internet as its platform. We can scrap it and still use the internet to build a different one.
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For the record I thought the TLD for this page was ‘cerfbound’, which sounds like the name for the race horse of the internet.
Do extraterrestrial civilizations also use TCP?
How much energy does the internet use?
I hate to think of the future of these nice blog posts, that need to struggle to convince the readers about the organic level of their content.
i have an idea for a new javascript framework
It’s worth considering how the tiny computers of the era forced a simple clean design. IPv6 was designed starting in the early 90s and they couldn’t resist loading it up with extensions, though the core protocol remains fine and is just IP with more bits. (Many of the extensions are rarely if ever used.)
If the net were designed today it would be some complicated monstrosity where every packet was reminiscent of X.509 in terms of arcane complexity. It might even have JSON in it. It would be incredibly high overhead and we’d see tons of articles about how someone made it fast by leveraging CPU vector instructions or a GPU to parse it.
This is called Eroom’s law, or Moore’s law backwards, and it is very real. Bigger machines let programmers and designers loose to indulge their desire to make things complicated.
What are some extensions? just curious.
IPSec was a big one that’s now borderline obsolete, though it is still used for VPNs and was back ported to IPv4.
Many networking folks including myself consider IPv6 router advertisements and SLAAC to be inferior, in practice, to DHCPv6, and that it would be better if we’d just left IP assignment out of the spec like it was in V4. Right now we have this mess where a lot of nets prefer or require DHCPv6 but some vendors, like apparently Android, refuse to support it.
The rules about how V6 addresses are chopped up and assigned are wasteful and dumb. The entire V4 space could have been mapped onto /32 and an encapsulation protocol made to allow V4 to carry V6, providing a seamless upgrade path that does not require full upgrade of the whole core, but that would have been too logical. Every machine should get like a /96 so it can use 32 bits of space to address apps, VMs, containers, etc. As it stands we waste 64 bits of the space to make SLAAC possible, as near as I can tell. The SLAAC tail must have wagged the dog in that people thought this feature was cool enough to waste 8 bytes per packet.
The V6 header allows extension bits that are never used and blocked by most firewalls. There’s really no point in them existing since middle boxes effectively freeze the base protocol in stone.
Those are some of the big ones.
Basically all they should have done was make IPs 64 or 128 bits and left everything else alone. But I think there was a committee.
As it stands we have what we have and we should just treat V6 as IP128 and ignore the rest. I’m still in favor of the upgrade. V4 is too small, full stop. If we don’t enlarge the addresses we will completely lose end to end connectivity as a supported feature of the network.
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I can easily spot it's an AI written article, because it actually explains the technology in understandable human language. A human would have written it the way it was either presented to them in university or in bloated IT books: absolutely useless.
I can easily spot it's an AI written comment, because it actually explains their idea in understandable human language and brings nothing to the discussion. A human would have written it the way they understand it and bring their opinions along: absolutely useless.
At first wanted to give the benefit of the doubt that this is sarcasm but gave a skim through history and I guess it's just a committed anti-AI agenda.
Personally I found the tone of the article quite genuine and the video at the end made a compelling case for it. Well I figure you commented having actually read it.
Edit: I can't downvote but if I could it probably would have been better than this comment!