I am facing an issue where DTLS handshake packets are greater than 1500 bytes and I do fragment to make sure it is less than 1500 bytes. The fragmentation is as per RFC dictated. Now, when I check captured pcap, fragment and reassembled when complete packet received. However DTLS SSL_connect function when I feed fragmented packets. The question here, should I reconstruct complete packet and give it to SSL_connect function?
As I read the current(?) draft:
3.1.2. Message Size TLS and DTLS handshake messages can be quite large (in theory up to 2^24-1 bytes, in practice many kilobytes). By
contrast, UDP datagrams are often limited to less than 1500 bytes if
IP fragmentation is not desired. In order to compensate for this
limitation, each DTLS handshake message may be fragmented over several
DTLS records, each of which is intended to fit in a single IP
datagram. Each DTLS handshake message contains both a fragment offset
and a fragment length. Thus, a recipient in possession of all bytes of
a handshake message can reassemble the original unfragmented message.
Related
So I am seeding on BitTorrent and there appears to be two-fold communication showing up on Wireshark. From peers around the world I receive UDP packets with 20 bytes of data. In response my BitTorrent sends UDP packets with around 1438 bytes of data.
This uTorrent protocol suggested here does not seem to have anything as small as the 28 byte (20 bytes data 8 bytes header) UDP packets, likewise this link isn't helpful.
What is the formal communication mechanism or protocol at play here? Is it possible to analyze those 1438-byte packets or in more detail in order to get a snippet of the file being sent? Or the structure of the 20 bytes of data being sent from my peers?
This uTorrent protocol suggested here does not seem to have anything as small as the 28 byte (20 bytes data 8 bytes header) UDP packets
The µTP header is 20 bytes. So those most likely are ACK messages. Wireshark should support decoding those packets, at least if you captured a connection from the beginning.
when transmit data from esp32 to an esp32,does the BLE's libary contain errorchecking? AKA resending packets? uses Ack-Naks? error correction also?
Thanks
There is error checking.
You can check if a write was successful.
A Ble packet consist of:
Preamble: RF synchronization sequence.
Access address: 32 bits, advertising or data access addresses (it is
used to identify the communication).
Header: its content depends on the packet type (advertising or data
packet).
Length: Length of the data payload(When encryption is used, the
message integrity code (MIC) at the end of the packet is 4 bytes, so
this leads to 251 bytes as actual maximum available payload size.
Checksum: CRC.
There is no error correction or resending of packets, unless you implement it yourself.
So,if a write was not successful you can resend a packet.
Is there any benifit to combining several UDP packets into one as opposed to sending them all one right after the other? I know that if the large packet gets courrupted then i loose all of them, but is there possibly some upside to sending them all in one? such as a lower chance of the large one being lost?
That would be at the discretion of the sending application.
Note that your large packet is limited by the MTU of the underlying network. e.g. the theoretical size of a UDP packet is 64k, but an ethernet frame is only ~1500 bytes. So I suspect this is not a practical feature.
Generally networking channels will be limited on the rate of packets that can be sent per second. Thus if you want to send millions of messages per second you generally want to combine into a smaller number of packets to run without major packet loss.
As an over generalisation, Windows doesn't like > 10,000 packets per second for UDP, but you can saturate a gigabit network with large MTU packets.
Is there any benifit to combining several UDP packets into one as opposed to sending them all one right after the other?
One can save on UDP header which is 8 bytes per datagram hence reducing the amount of data sent over the wire. Just make sure you don't send more then MTU sans IP and UDP header sizes to avoid fragmenting on IP layer.
Also, the standard POSIX socket API requires one send/sendto/sendmsg() system call to send or receive one datagram, so by sending fewer datagrams one does fewer system call reducing the overall latency (an order of a few microseconds per call). Linux kernels starting from 3.0 provide sendmsg() and recvmmsg() functions to send and receive multiple datagrams in one system call.
I know that if the large packet gets courrupted then i loose all of the
True. However, if the protocol can't cope with UDP datagram loss at all it may not matter that much - as soon as one datagram is lost it's broken anyway.
It is important for situations where packet size is small (less than 100 byte). The IP/UDP header is at least 28 bytes.
Imagine you have streaming connection to a server, each packet contains 50 bytes and your software sends packets with rate 1000 packet per second.
The actual payload is 1000 * 50 bytes = 50000 bytes. Headers overhead 1000 * 28 = 28000 bytes Total bytes : 50000 + 28000 = 87000 ==> 87 KBps
Imagine you can combine each 3 UDP packets into one packet:
Headers overhead 1000 / 3 * 28 = 9333 Total bytes : 50000 + 9333 ===> 60 KBps
This -in some applications- saves good portion of the bandwidth.
I want to know, how much overhead WebRTC produces when sending data over datachannels.
I know that Websockets have 2 - 14 Bytes overhead for each frame. Does WebRTC use more Overhead? I cannot find some useful information on the web. Its clear for me, that Datachannels can not be used for now. How much overhead is used with Mediastreams?
Thanks
At the application layer, you can think of DataChannel as sending and
receiving over SCTP. In the PPID (Payload Protocol Identifier) field of the
SCTP header, Datachannel sets value 0x51 for indicating that it's sending UTF-8
data and 0x52 for binary data.
Yes, you are right. RTCDataChannel uses SCTP over DTLS and UDP. DTLS is used for
security. However, SCTP has problems traversing most NAT/Firewall setups.
Hence, to overcome that, SCTP is tunneled through UDP. So the overall overhead
to send data would be overhead of:
SCTP + DTLS + UDP + IP
and that is:
28 bytes + 20-40 bytes + 8 bytes + 20 - 40 bytes
So, the overhead would be rougly about 120 bytes. The maximum size of the SCTP
packet that a WebRTC client can send is 1280 bytes. So at max, you can send
roughly 1160 bytes of data per SCTP packet.
WebRTC uses RTP to send its media. RTP runs over UDP.
Besides the usual IP and UDP headers, there are two additional headers:
The RTP header itself starts from 12 bytes and can grow from there, depending on what gets used.
The payload header - the header that is used for each data packet of the specific codec being used. This one depends on the codec itself.
RTP is designed to have as little overhead as possible over its payload due to the basic reasoning that you want to achieve better media quality, which means dedicating as many bits as possible to the media itself.
Here's a screenshot of 2 peer.js instances (babylon.js front end) sending exactly 3 bytes every 16ms (~60 per second).
The profiler shows 30,000 bits / second:
30,000 bits / 8 bits per byte / 60 per second = 62.5 bytes, so after the 3 bytes I'm sending it's ~59.5 bytes according to the profiler.
I'm not sure if something is not counted on the incoming, because it is only profiling half that, 15k bits / second
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.