We use an embedded device to send packets from a serial port over a serial-to-Ethernet converter to a server. One manufacturer we use, Moxa, will always send the packets in the same manner which they are constructed. Meaning, if we construct a packet size of 255, it will always send the packet in a 255 length. The other manufacturer, Tibbo, if we send the packet size 255, it will break the packet up if it is greater than 128. This is the answer I received from the Tibbo engineers at the time:
"From what I understand and what the
engineers said, even if the other
devices provide you with the right
packet size now does not guarantee
that when implemented in other
networks the same will happen. This
is the reason why we feel that packet
size based data transfer through TCP
is not reliable as it was not the way
TCP was designed to be used."
I understand that this may not be how TCP was designed to be used, but if I create a packet of 255 bytes and TCP allows it, then how is this outside of how TCP works? I understand that at some point the packet may get broken up but if the server is expecting a certain packet size and Moxa's offering does not have the same problem as the Tibbo device.
So, is it possible to guarantee a reasonable TCP packet size?
No. TCP is not a packet protocol, it is a stream protocol. It guarantees that the bytes you send will all arrive, and in the right order, but nothing else. In particular, TCP does not give you any kind of message or packet boundaries. If you want such things, they need to be implemented at a higher level by your protocol.
Related
I have 1 server and several (maybe up to 20) clients. All clients are sending UDP datagram at random time. Each datagram is quite short (about 10B), but I must make sure all the data from each client is received correctly.
If I let all clients send datagram to the same port, and client B sends it datagram at the exact time when the server is receiving data from client A, it seems the server will miss the data from client A.
So what's the correct method to do this job? Do I need to create a listener for each of the 20 clients?
When you bind a UDP socket to a port, the networking stack will allocate a buffer for a finite number of incoming UDP packets for you, so that (assuming you call recv() in a relatively timely manner), no incoming packets should get lost.
If you want see your buffer size in terminal, you can take a look at:
/proc/sys/net/core/rmem_default for recv
and
/proc/sys/net/core/wmem_default for send
I think the default buffer size on Linux is 131071B.
On Linux, you can change the UDP buffer size (e.g. to 26214400) by (as root):
sysctl -w net.core.rmem_max=26214400
You can also make it permanent by adding this line to /etc/sysctl.conf:
net.core.rmem_max=26214400
Since each packet is only 10B, shouldnt be a problem.
If you are still worried about packet loss you could implement a protocol where your client waits for a ACK from the server or it will resend. Many protocols use such a feature, but this is only possible if timing allows it. For example in streaming data it is not useful because there is no time to resend.
or consider using tcp ( if it is an option)
I am super new to the networking world, so I have a QTcpserver that's currently working with the newConnection signal, but I was wondering if I could use QUdpSocket with a QTcpServer ? It's this possible at all ?
TCP is a connection oriented stream over an IP network. It guarantees
that all sent packets will reach the destination in the correct order.
This imply the use of acknowledgement packets sent back to the sender,
and automatic retransmission, causing additional delays and a general
less efficient transmission than UDP.
UDP is a connection-less protocol. Communication is datagram oriented.
The integrity is guaranteed only on the single datagram. Datagrams
reach destination and can arrive out of order or don't arrive at all.
It is more efficient than TCP because it uses non ACK. It's generally
used for real time communication, where a little percentage of packet
loss rate is preferable to the overhead of a TCP connection.
StackOverFlow
So the simple Answer is No , You can't ,because tcp and udp are 2 different protocols .
As we know the UDP places limit on the block size when the datagrampacket is sent from the client to the server. Still there are protocols like RTP which use UDP to transmit real time video/audio to the destination. I presume that the streaming of data will be variable in nature.
Could someone explain how does UDP manage to transmit variable block size over the internet? I have not found anything much in this regard.
Thanks,
How UDP could be changed to support arbitrary block sizes?
It does support arbitrary block sizes, up to 65507 bytes, more in IPv6.
Could someone explain how does UDP manage to transmit variable block size over the internet?
It transmits variable block sizes within the limit, which is imposed not by UDP but by the path MTU at the IP layer and the socket send buffer at the application layer.
I have not found anything much in this regard.
Hard to believe.
i am writing my programming for ar drone using UDP, but the drone sometimes hangs, i think maybe the UDP is not big enough, is there any way to improve the situation if still using UDP?
UDP datagram sizes are limited by:
The outgoing socket send buffer. Defaults for this vary wildly, from 8k in Windows to 53k or more in other systems.
The Path MTU or Maximum Transmission Unit of the entire network path between you and the receiver. This is typically no more than 1260 bytes.
The safest size for a UDP datagram is 534 bytes as this will never be fragmented by the transport. If you're planning on sending datagrams larger than this you need to consider a different transport, usually TCP.
Let's say I am trying to send data using udp socket. If the data is big then I think the data is going to be divided into several packets and sent to the destination.
At the destination, if there is more than one incoming packets then how to I combined those separated packets into the original packet? Do I need to have a data structure that save all the incoming udp based on the sender ? Thanks in advance..
If you are simply sending the data in one datagram, using a single send() call, then the fragmentation and reassembly will be done for you, by the transport layer. All you need to do is supply a large enough buffer to recv(), and if all the fragments have arrived, then they will be reassembled and presented to you as a single datagram.
Basically, this is the service that UDP provides you (where a "datagram" is a single block of data sent by a single send() call):
The datagram may not arrive at all;
The datagram may arrive out-of-order with respect to other datagrams;
The datagram may arrive more than once;
If the datagram does arrive, it will be complete and correct1.
However, if you are performing the division of the data into several UDP datagrams yourself, at the application layer, then you will of course be responsible for reassembling it too.
1. Correct with the probability implied by the UDP checksum, anyway.
You should use TCP for this. TCP is for structured data that needs to arrive in a certain order without being dropped.
On the other hand, UDP is used when the packet becomes irrelevant after ~500 ms. This is used in games, telephony, and so on.
If your problem requires UDP, then you need to handle any lost, duplicate, or out-of-order packets yourself, or at least write code that is resilient to that possibility.
http://en.wikipedia.org/wiki/User_Datagram_Protocol
If you can't afford lost packets, then TCP is probably a better option than UDP, since it provides that guarantee out of the box.