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It is a common knowledge that transferring TCP packet inside a tunnel with TCP connection can create a devastating effect called TCP meltdown and degrade tunnel quality greatly. I somehow wondering if similar effect may happen in we try to transfer TCP data over a Quic connection. Even though Quic is UDP packets, but it need to have something similar to windowing for keeping track of received packets in order to provide a connection-oriented protocol. So I'm not sure if a similar effect will happen or not.
Any idea?
QUIC indeed uses a similar congestion control as TCP, see https://www.rfc-editor.org/rfc/rfc9002.html#name-congestion-control. So when tunnelling a TCP connection over a QUIC stream, I would say the same "meltdown" problems can occur (a QUIC stream has the same properties as a TCP connection: reliable ordered stream of byte, so the stream will stall if QUIC packets are lost).
However, a QUIC extensions is being defined for sending datagrams, https://datatracker.ietf.org/doc/html/draft-ietf-quic-datagram. That might provide a better way for transporting TCP packets, as these datagrams will never be retransmitted at QUIC level. However, it would require TCP packets to fit into a Datagram frame.
I had a strange idea. I heard of software which from my understanding uses UDP to transfer files decreasing the overhead found in TCP packets.
If my app requires TCP and my LAN has software set up to communicate with another datacenter on the other side of the coast with software setup on their end. Would it be possible to send the actual data via UDP but than simulating TCP on both ends?
Does anyone have any ideas or information about such projects?
If you're asking if you can use UDP as a Layer 2, then the answer is yes, sort of. There are various protocols that allow you to create a tunnel to another network using a UDP transport, such as L2TP and even IPsec (with NAT traversal). You could also do it at the application layer.
If you're asking if TCP can be implemented in UDP, the answer is no. First, TCP packets and UDP packets have an incompatible format. Second, TCP and UDP have different protocol numbers (seen in the IP header) which means that TCP traffic destined for a UDP port would not be passed to the correct upper-layer protocol.
Both TCP and UDP are built on top of the IP, but the TCP uses different packet structure and at the layer-2 it is not possible to mimic the TCP using UDP packets.
Of course, if you have the control on both the source and destination, then it is possible to create a reliable UDP tunnel for the TCP packets. This would require some internal information (packet number, ack/nack flags) in the body of the UDP packet.
There is an interesting project http://udt.sourceforge.net/
It is a broadcast-capable reliable file transfer mechanism built on top the UDP.
PseudoTCP is a protocol which implements TCP algorithms on top of the UDP. It was introduced since the NAT traversal for TCP is much more complicated than UDP. But some P2P applications do need a reliable data transfer among nodes.
So far as I know, there are two PseudoTCP variations: Libjingle and Libnice.Libjingle is an open source library from google which was initially for gtalk. You could take a look at file sharing example from libjingle: https://developers.google.com/talk/libjingle/file_share. Recently, Chrome desktop also use PseudoTCP implementation from libjingle for reliable connections.
Yes, you can develop a protocol on UDP that simulates TCP. However, if you simulated TCP fully, it would technically have more overhead. Because TCP is implement as the packet and your simulated TCP is implemented in the body of the packet.
If you only need one or two features of TCP (such as basic ordering), then implementing it in UDP is useful.
Halo uses 2-3 (IIRC) UDP protocols that simulate different features of TCP, then full fledged TCP for initializing game-states. I Shot You First Networking, GDC publication
For example, in one case, they send 3 duplicate UDP packets to overcome packet loss.
If you control the software on both ends, and it is cost-effective to build your own protocol, then UDP can be versatile.
One way to do it now on Linux-3.18+ is to use Foo over UDP (FOU) which implements Generic UDP Encapsulation (GUE). Here's a good introduction to FOU, and the man page for ip-fou.
Or if you want an [open source] UDP based file transfer system there are things like UDT, UFTP, Tsunami-UDP, and even Google's QUIC (Now deprecated in favour of IETF QUIC).
Update: The QUIC protocol now has been standardised by the IETF which provides for secure reliable and unreliable transport over UDP as an alternative to TCP. There's a wide range of QUIC implementations available. There is also a growing set of protocol mappings on to QUIC such as HTTP/3, DNS over QUIC, etc
If my app requires TCP and my LAN has software setup to communicate
with another datacenter on the other side of the coast with software
setup on their end. Would it be possible to send the actual data via
UDP but than simulating TCP on both ends?
No. A UDP socket is in a different namespace from a TCP socket. You will be unable to write UDP at one end and send or receive TCP at the other end. TCP and UDP are peer protocols; both exist at the layer above IP. You can't use one to spoof the other.
Hmm, I believe so. You'd need to use a proxy at both ends, but it should be possible.
The biggest problem you are going to run into is that UDP is designed with the idea that you don't care if some of the packets don't ever make it to the other end.
Here's a link with some more info:
http://www.cyberciti.biz/faq/key-differences-between-tcp-and-udp-protocols/
IMHO, it's not a good idea to transmit files via UDP.
TCP's problems are in its algorithms, not its headers.
You certainly could implement the TCP algorithms on top of UDP. That would effectively be the same as tunneling TCP datagrams inside of UDP datagrams. But all this accomplishes is to add a few more bytes of overhead to each packet, and require another endpoint to unwrap the packets.
UDP itself is just thin shim on top of IP: its a convenient way to access IP packet switched networking without having to dive into kernels or receive special handling from routers. The main reason to implement reliable transport on top of UDP is to get away from TCP algorithms in favor of something more efficient. FileCatalyst was mentioned above as one company which does this, and my own company Data Expedition, Inc. does so as well.
So you could implement TCP algorithms on top of UDP, but you wouldn't want to.
You can simulate something like a connection over UDP, and you as well can add reliability checks and ordering and retransmission and so on. - but then, it still isn't TCP, it just acts the way.
Of course, one of the ends can be a kind of "hub" or "proxy" which does an adaption. Then you don't have a 2-end solution, but in fact a 4 end solution - one pair with "real" TCP and the other with the "self-knitted" "TCP" - which you put together with an appropriately crafted program.
Are there any networking protocols that are not strictly TCP or UDP but can be used with either one?
For example, HTTP, FTP, STMP, RTMP are always TCP.
DNS, SNMP, DHCP, RIP are always UDP.
Is there anything that can be either TCP or UDP? Or am I wrong in the above assertions?
RTSP is one weird one I know of that uses both, TCP for the control port but UDP for audio/video/quality, but it has strict requirements of what gets sent of each.
I'm asking about standard, published, or at least commonly used protocols, not custom ones.
DNS can use either UDP or TCP; TCP is required when the response data exceeds 512 bytes.
If you examine a Windows' services file you will see a number of protocols registered for both TCP and UDP. Path: C:\Windows\System32\drivers\etc In fact, most of the listings in the services file use both TCP and UDP protocols.
As far as well known apps that use both, I would think that most chat applications use both. sms-chat definitely does but probably most others.
Edit:
From that file, here's a few of the protocols that can be sent over either TCP or UDP (there are exactly 100 listed protocols that use both in the file, many internal MS protocols):
echo
discard
daytime
qotd (Quote of the day)
chargen (Character generator)
time
SIP can use UDP, TCP or SCTP. Using a reliable transport becomes important in SIP if your messages get to be at all large (i.e., significantly larger than the smallest MTU in between user agents). A good example is shared- or bridged-line appearances, which use a form of presence with XML bodies. The larger the number of SIP clients in the shared-line group, the larger the packets are likely to be, making fragmentation and retransmission an issue.
SIP can be either UDP or TCP. However, the reality is that UDP is mostly used for this protocol.
SNMP almost always runs over UDP, but it can and does run over TCP. Theory says that it's a bad idea to do SNMP over an error-correcting transport because because some of the very errors that SNMP intends to detect are masked.
is it possible to send multiple tcp or udp packets on a single ip packet? are there any specifications in the protocol that do not allow this.
if it is allowed by the protocol but is generally not done by tcp/udp implementations could you point me to the relevant portion in the linux source code that proves this.
are there any implementations of tcp/udp on some os that do send multiple packets on a single ip packet. (if it is allowed).
It is not possible.
The TCP seqment header does not describe its length. The length of the TCP payload is derived from the length of the IP packet(s) minus the length of the IP and TCP headers. So only one TCP segment per IP packet.
Conversely, however, a single TCP segment can be fragmented over several IP packets by IP fragmentation.
Tcp doesn't send packets: it is a continuous stream. You send messages.
Udp, being packet based, will only send one packet at a time.
The protocol itself does not allow it. It won't break, it just won't happen.
The suggestion to use tunneling is valid, but so is the warning.
You might want to try tunneling tcp over tcp, although it's generally considered a bad idea. Depending on your needs, your mileage may vary.
You may want to take a look at the Stream Control Transmission Protocol which allows multiple data streams across a single TCP connection.
EDIT - I wasn't aware that TCP doesn't have it's own header field so there would be no way of doing this without writing a custom TCP equivalent that contains this info. SCTP may still be of use though so I'll leave that link.
TCP is a public specification, why not just read it?
RFC4164 is the roadmap document, RFC793 is TCP itself, and RFC1122 contains some errata and shows how it fits together with the rest of the (IPv4) universe.
But in short, because the TCP header (RFC793 section 3.1) does not have a length field, TCP data extends from the end of the header padding to the end of the IP packet. There is nowhere to put another data segment in the packet.
You cannot pack several TCP packets into one IP packet - that is a restriction of specification as mentioned above. TCP is the closest API which is application-oriented. Or you want to program sending of raw IP messages? Just tell us, what problem do you want to solve. Think about how you organize the delivery of the messages from one application to another, or mention that you want to hook into TCP/IP stack. What I can suggest you:
Consider packing whatever you like into UDP packet. I am not sure, how easy is to initiate routing of "unpacked" TCP packages on remote side.
Consider using PPTP or similar tunnelling protocol.
This might be a silly question:
Does HTTP ever use the User Datagram Protocol?
For example:
If one is streaming MP3 or video using HTTP, does it internally use UDP for transport?
From RFC 2616:
HTTP communication usually takes place
over TCP/IP connections. The
default port is TCP 80, but other
ports can be used. This does not
preclude HTTP from being implemented
on top of any other protocol on the
Internet, or on other networks. HTTP
only presumes a reliable transport;
any protocol that provides such
guarantees can be used; the mapping
of the HTTP/1.1 request and response
structures onto the transport data
units of the protocol in question is
outside the scope of this
specification.
So although it doesn't explicitly say so, UDP is not used because it is not a "reliable transport".
EDIT - more recently, the QUIC protocol (which is more strictly a pseudo-transport or a session layer protocol) does use UDP for carrying HTTP/2.0 traffic and much of Google's traffic already uses this protocol. It's currently progressing towards standardisation as HTTP/3.
Typically, no.
Streaming is seldom used over HTTP itself, and HTTP is seldom run over UDP. See, however, RTP.
For something as your example (in the comment), you're not showing a protocol for the resource. If that protocol were to be HTTP, then I wouldn't call the access "streaming"; even if it in some sense of the word is since it's sending a (possibly large) resource serially over a network. Typically, the resource will be saved to local disk before being played back, so the network transfer is not what's usually meant by "streaming".
As commenters have pointed out, though, it's certainly possible to really stream over HTTP, and that's done by some.
Maybe just a bit of trivia, but UPnP will use HTTP formatted messages over UDP for device discovery.
Yes, HTTP, as an application protocol, can be transferred over UDP transport protocol.
Here are some of the services that use UDP and an underlying protocol for transferring HTTP data and streaming it to the end-user:
XMPP's Jingle Raw UDP Transport Method
A number for services that use UDT --- UDP-based Data Transfer Protocol, which is the a superset of UDP protocol.
The Transport Layer Security (TLS) protocol encapsulating HTTP as well as the above mentioned XMPP and other application protocols does have an implementation that uses UDP in its transport layer; this implementation is called Datagram Transport Layer Security (DTLS).
Push notifications in GNUTella are HTTP requests sent over UDP transport.
This article contains further details on streaming over UDP and its reliable superset, the RUDP: Reliable UDP (RUDP): The Next Big Streaming Protocol?
Of course, it doesn't necessarily have to be transmitted over TCP. I implemented HTTP on top of UDP, for use in the Satellite TV Broadcasting industry.
If you are streaming an mp3 or video that may not necessarily be over HTTP, in fact I'd be suprised if it was. It would probably be another protocol over TCP but I see no reason why you cannot stream over UDP.
If you do you have to take into account that there is no certainty that your data will arrive at the other end, but I can take it that you know about UDP.
To answer you question, No, HTTP does NOT use UDP.
For what you talk about though, mp3/video streaming COULD happen over UDP and in my opinion should never happen over HTTP.
Maybe some change on this topic with QUIC
QUIC (Quick UDP Internet Connections, pronounced quick) is an experimental transport layer network protocol developed by Google and implemented in 2013. QUIC supports a set of multiplexed connections between two endpoints over User Datagram Protocol (UDP), and was designed to provide security protection equivalent to TLS/SSL, along with reduced connection and transport latency, and bandwidth estimation in each direction to avoid congestion. QUIC's main goal is to optimize connection-oriented web applications currently using TCP.
I think some of the answers are missing an important point. The choice between UDP and TCP should not be based on the type of data (e.g., audio or video) or whether the application starts to play it before the transfer is completed ("streaming"), but whether it is real time. Real time data is (by definition) delay-sensitive, so it is often best sent over RTP/UDP (Real Time Protocol over UDP).
Delay is not an issue with stored data from a file, even if it's audio and/or video, so it is probably best sent over TCP so any packet losses can be corrected. The sender can read ahead and keep the network pipe full and the receiver can also use lots of playout buffering so it won't be interrupted by the occasional TCP retransmission or momentary network slowdown. The limiting case is where the entire recording is transferred before playback begins. This eliminates any risk of a playback stall, but is often impractical.
The problem with TCP for real-time data isn't retransmissions so much as excessive buffering as TCP tries to use the pipe as efficiently as possible without regard to latency. UDP preserves application packet boundaries and has no internal storage, so it does not introduce any latency.
(This is an old question, but it deserves an updated answer.)
In all likelihood, HTTP/3 will be using the QUIC protocol, which is described as
multiplexed transport over UDP
So, from a certain point of view, you could say that HTTP/3 will be using UDP.
The answer: Yes
Reason: See the OSI model.
Explaination:
HTTP is an application layer protocol, which could be encapsulated with a protocol that uses UDP, providing arguably faster reliable communication than TCP. The server daemon and client would obviously need to support this new protocol. Quake 2 protocol proves that UDP can be used over TCP to provide a basis for a structured communication system insuring flow control (e.g. chunk ids).
http over udp is used by some torrent tracker implementations (and supporteb by all main clients)
In theory yes it is possible to use UDP for http but that might be problematic. Say for instance in your example a mp3 or a video is being streamed there will be problem of ordering and some bits might go missing as UDP is not connection oriented there is no retransmit mechanism.
HTTP/3 (aka QUIC) uses UDP instead of TCP.
https://http3-explained.haxx.se/en/the-protocol/feature-udp
UDP is the best protocol for streaming, because it doesn't make demands for missing packages like TCP. And if it doesn't make demands, the flow is far more faster and without any buffering.
Even the stream delay is lesser than TCP. That is because TCP (as a far more secure protocol) makes demands for missing packages, overwriting the existing ones.
So TCP is a protocol too advanced to be used for streaming.