Can UDP packets be partially sent like TCP ones?

6
votes

I've created some type of client/server application that has its own data ACK system. It was originally written in TCP because of some limitations, but the base was written thinking about UDP.

The packets that I sent to the server had their own encapsulation (packet id and packet size headers. I know that UDP has also a checksum so I didn't add a header for that), but how TCP works, I know that the server may not receive the entire packet, so I gathered and buffered the data received until a full valid packet was received.

Now I have the chance to change my client/server program to UDP, and I know that one difference with TCP is that data is not received in the same order as sent (which is why I added a packet id header).

The thing that I want to know is: If I send multiple packets, will they be received with no guaranteed order but with guaranteed encapsulation? I mean, if I send a packet sized 1000 bytes of data and another packet sized 400 bytes of data later, will the server receive 2 packets, one of 1000 bytes and another one of 400 bytes, or is there a chance to receive 200 of that 1000 bytes, then 400 bytes of that 1000 bytes and later the rest of the bytes like TCP does?

2
socketstcpudpdifference
You really shouldn't use the word "packet" to refer to so many different things. Protocol data units that might be sent in any number of packets should be called "messages". – David Schwartz
Well, I'm calling them packets in the application layer, and, as I read everywhere (including wiki), that's how people understand them when talking in this layer. I see them talking about messages when they also talk about the transport layer and have to difference between messages and packets. – Jorge Fuentes González
The computer science definition of a packet is, "the unit of data that is routed between an origin and a destination on the Internet or any other packet-switched network." It's true that the word is sometimes used to mean something else, but when you're talking about systems involving actual packets, it causes confusion to use the same word to mean two totally different things. – David Schwartz

2 Answers

16
votes

UDP is a datagram service. Datagrams may be split for transport, but they will be reassembled before being passed up to the application layer.

0
votes

With small packet sizes you should have no concern that packets will be broken into multiple packets. That generally is only an issue when the packet gets over an Ethernet network.

You ask" will the server receive 2 packets, one of 1000 bytes and another one of 400 bytes, or there's a chance to receive 200 of that 1000 bytes, then 400 bytes of that 1000 bytes and later the rest of the bytes like TCP can do?

With a packet size of under 1492 bytes there is not going to be any partial packets.

UPDATE:
Apparently I see a need to clarify why I say UDP packet lengths 1492 bytes or less will not affect transport robustness.

The maximum UDP length as implicitly specified in RFC 768 is 65535 including the 8 byte Header. Max Payload Frame Length is 65527 bytes.

While this should not be disputed number the UDP data length is often reported incorrectly. This is exemplified in a previous post:

What is the largest Safe UDP Packet Size on the Internet

A data packet is not constrained by the MTU of the underlying network ToS or communications protocol's Frame length (e.g. IP and Ethernet respectively). Discrepancies between MTU and Protocol Lengths are remedied by Fragmentation and Reassembly

At the Transport Layer each network Type of Service (ToS) has a specific Maximum Transmission Unit (MTU). UDP is encapsulated within IP Packets and IP Packets are encapsulated by the transporting Network's ToS. IP packets are often transmitted through networks of various ToS which include Ethernet, PPP, HDLC, and ADCCP.

When the MTU for a receiving Network ToS is less than the sending ToS then the receiving network must Fragment the received packet. When the Network sends a packet to a network with a higher MTU, the receiving Network must reassemble any Fragmented packets.

Ethernet is the defacto mainstream protocol with the lowest MTU. Non-Mainsteam Arcnet the MTU is 507 bytes. The practical lowest MTU is Ethernet's 1500 bytes, minus the overhead makes maximum payload length 1492 bytes.

If the UDP packet has more than 1492 bytes the data packet will likely be Fragmented and Reassembled. The Fragment and Reassembly adds complexity to the already complex process coupling UDP and IP, and therefore should be avoided.

Because UDP is a non-guaranteed datagram delivery protocol it boosts the transport performance. Robustness is left to the originating and terminating Application. RFC 1166 sets the standards for the communication protocol link layer, IP layer, and transport layer, the UDP Application is responsible for packetization, reassembly, and flow control.

The maximum UDP packet size can also be lowered by a Communication Host's Application Layer. The packet length is a balance between performance and robustness.

The Communications Host's Application Layer may set a maximum UDP packet size. The typical UDP max data length at the Application layer will use the maximum allowed by the IP protocol or the Host Data Link Layer, typically Ethernet.

It is the Application's programer that chooses to use the Host Application Layer or the Host Data Link layer. The Host Application Layer will detect UDP packet errors and discard the packet if necessary. When the application communicates directly with the Host Data Link, the application then has the responsibility of detecting packet errors.

Using maximum UDP data packet length of Ethernet's max payload length of 1492 bytes will eliminate the issues of Fragmentation and Delivery Order of multiple Frames.

That is why I said packet length is not a Fragmentation issue with packet lengths of 1000 and 400 bytes.

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I do not know what you mean by "guaranteed encapsulation", it makes no sense to me.

With IP there is no guarantee of packet delivery of the order whether UDP or TCP.

As long as you control both sides of the conversation, you can work out your own protocol within the data packet to handle ordering and post packets. Reserve the first x bytes of the packet for a sequential order number and total number of packets. (e.g. 1 of 3, 2 of 2, 3 of 3). If the client side is missing a packet then the client must send a request for retransmission. You need to determine to what level you are going to go for data integrity. like maybe the re-transmission packet is lost.

That may be what you meant by "guaranteed encapsulation", Where there is other information within your datagram packet to ensure some integrity. You should add your own CRC for the total data being sent if broken into multiple datagrams. the checksum is not very robust and is only for the one packet.

UDP is much faster then TCP but TCP has flow control and guaranteed delivery.

UDP is good for streaming content like voice where a lost packet is not going to matter.

Network reliability has improved a lot since the days when these issues were a major concern.