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 .
Related
I want to develop a program in c using pjsip for peer to peer file transfer. As pjsip uses ice and in ICE UDP is used, so do I need to handle the packet delivery assurance.
And as I would be sending the file by breaking it into several parts and them re assemble all the parts at the receiver's end, so do I have to maintain the sequence of the packets or can i assume that packets are delivered in the correct sequence??
With UDP you can neither assume that packets are delivered in order nor that they are delivered exactly once nor that they are delivered at all! So you need to come up with a protocol that does a lot of things which normally TCP would take care of. It has to reassemble the original data stream and handle the things I listed above.
Additionally, with UDP you can have the problem that you cause congestion. TCP can avoid that with its congestion avoidance algorithms, with UDP you can easily send packets too fast causing them to drop at the overloaded router.
All these are non trivial problems to solve so I suggest you read up on the topic. I'd start with a good book about TCP.
It seems that a UDP packet can be sent without a payload.
The only thing I can think of that doesn't need a payload is for NAT hole punching.
What else could this be used for?
This relates to my previous question Under Linux, can recv ever return 0 on UDP?
I suppose more to the point is that if it's been specified as part of some standard, then it's been thought to be useful somewhere right?
Anything! The UDP packet isn't empty -- it comes with the sender's identity. Therefore, such a packet could be used as a primitive kind of signal: maybe a hello, a goodbye, or a keep-alive.
With interfaces like sendmsg, an empty packet might be used in order to send auxiliary data, like a cmsg structure (which can be used for things like transferring file descriptors between two processes on Linux).
EDIT: One more use: NAT traversal algorithms such as STUN or UDP hole punching.
To answer the question of "why would a protocol do this": the old Daytime protocol just uses the arrival of a UDP packet to send back a reply packet. Similarly, it replies with time value as soon as a TCP connection happens regardless of any actual data that the TCP connection contains.
a UDP packet without payload may be sent to detect if a UDP port is closed. if closed, an ICMP-unreach is replied.
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.
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.
I'm attempting to send multichannel audio over WiFi from one server to multiple client computers using UDP broadcast on a private network.
I'm using software called Pure Data, with a UDP broadcast tool called netsend~ and netreceive~. The code is here:
http://www.remu.fr/sound-delta/netsend~/
To cut a long story short, I'm able to achieve sending 9 channels to one client computer in a point-to-point network, but when I try to do broadcast to 2 clients (haven't yet tried more), I get no sound. I can compress the audio and send 4 channels compressed (about 10% size of uncompressed) over UDP broadcast to 2 clients successfully. Or I can send 1 channel over UDP broadcast to 2 clients, with some glitches.
The WiFi router is a Linksys WRT300N. All computers are running Windows XP. The IP addresses are 192.168.1.x, with subnet mask 255.255.255.0 and the subnet broadcast address: 192.168.1.255.
I'm curious - what happens to UDP broadcast packets in the router?
If I have a subnet mask of 255.255.255.0, then does the router make 254 packets for every packet sent ot the broadcast address?
My WiFi bandwidth is at least 100Mbps, but I can't seem to send audio of more than around 10Mbps over UDP broadcast to multiple clients.
What's stopping me from sending audio up to the bandwidth limit of the WiFi?
any suggestions for socket code modifications, network setups, router setups, subnet modifications... all very much appreciated!
thanks
Nick
Your problem is caused by the access point's rate control algorithm. With unicast the access point tracks what data rate every particular receiver can reliably receive at, and sends about that rate. With multicast the access point does not know which receivers are interested in the data, so simple access points send the data at the slowest possible rate (1Mb/s). Better implemented access points may send the data at the rate that the slowest connected client is using, and the best access points use IGMP snooping to see who's receiving each IP multicast stream, and they will choose the slowest rate out of the receivers for that stream.
The simplest solution is to not use multicast when you have a small number of WiFi receivers.
Are all parties connected via WiFi or is the sender using a
wired connection to the Access Point? Broadcast data will
be transmitted as unicast data from a station to an access
point and the access point will then retransmit the data
as broadcast/multicast traffic so it will use twice the
on-air bandwidth compared to when the sender sits on the
wired side of the AP.
When sending a unicast frame the AP will wait for an ACK
from the receiving station and it will retransmit the
frame until the ACK arrives (or it times out). Broadcast/multicast
frames are not ACKed and therefore not retransmitted.
If you have a busy/noisy radio environment this will
cause the likelyhood of dropped packets to increase,
potentially a lot, for multicast traffic compared to unicast
traffic. In an audio application this could certainly be audible.
Also, IIRC, broadcast/multicast traffic does not use the
RTS/CTS procedure for reserving the media which exarbates
the dropped packets problem.
It could actually be the case that multiple unicast streams
work better than a single multicast stream under less-than-ideal
radio conditions given that the aggregated bandwidth is
high enough.
If you can I would suggest that you use wireshark to sniff
the WiFi traffic and take a look at the destination address
in the 802.11 header. Then you can verify if the packets
are actually broadcast or not over the air.
Your design is failing due to a common misconception with WiFi speeds. With 802.11n the number 300mb/s is the link speed, not the actual bandwidth available for user data or even the IP layer. The effective bandwidth is closer to 40mb/s best case, have a look at the FAQ on SmallNetBuilder.com that discusses this in further detail.
http://www.smallnetbuilder.com/wireless/wireless-basics/31083-smallnetbuilders-wireless-faq-the-essentials
I'm curious - what happens to UDP broadcast packets in the router? If I have a subnet mask of 255.255.255.0, then does the router make 254 packets for every packet sent ot the broadcast address?
No the "router" doesn't make 254 individual packets. Furthermore, I suspect the protocol leverages "multicast" addresses rather than using a "broadcast" address.
Since broadcast/multicast traffic can easily be misused, there are many networking equipment that limit/block by default such traffic. Of course, some essential protocols (e.g. ARP, DHCP) rely on broadcast/multicast addresses to function and won't be blocked by default.
Hence, it might be a good thing to check for multicast/broadcast control knobs on your router.