I've known about SCTP for a decade or so, and although I never got to use it yet, I've always wanted to, because of some of its promising (purported) features:
multi-homing
multiplexing w/o head-of-line blocking
mixed order/unordered delivery on the same connection (aka association)
no TIME_WAIT
no SYN flooding
A Comparison between QUIC and SCTP however claims
SCTP intended to get rid of HOL-Blocking by substreams, but its
Transmission Sequence Number (TSN) couples together the transmission
of all data chunks. [...] As a result, in SCTP if a packet is lost,
all the packets with TSN after this lost packet cannot be received
until it is retransmitted.
That statement surprised me because:
removing head-of-line blocking is a stated goal of SCTP
SCTP does have a per-stream sequence number, see below quote from RFC 4960, which should allow processing per stream, regardless of the association-global TSN
SCTP has been in use in the telecommunications sector for perhaps close to 2 decades, so how could this have been missed?
Internally, SCTP assigns a Stream Sequence Number to each message
passed to it by the SCTP user. On the receiving side, SCTP ensures
that messages are delivered to the SCTP user in sequence within a
given stream. However, while one stream may be blocked waiting for
the next in-sequence user message, delivery from other streams may
proceed.
Also, there is a paper Head-of-line Blocking in TCP and SCTP: Analysis and Measurements that actually measures round-trip time of a multiplexed echo service in the face of package loss and concludes:
Our results reveal that [..] a small number of SCTP streams or SCTP unordered mode can avoid this head-of-line blocking. The alternative solution of multiple TCP connections performs worse in most cases.
The answer is not very scholarly, but at least according to the specification in RFC 4960, SCTP seems capable of circumventing head-of-line blocking. The relevant claim seems to be in Section 7.1.
Note: TCP guarantees in-sequence delivery of data to its upper-layer protocol within a single TCP session. This means that when TCP notices a gap in the received sequence number, it waits until the gap is filled before delivering the data that was received with sequence numbers higher than that of the missing data. On the other hand, SCTP can deliver data to its upper-layer protocol even if there is a gap in TSN if the Stream Sequence Numbers are in sequence for a particular stream (i.e., the missing DATA chunks are for a different stream) or if unordered delivery is indicated. Although this does not affect cwnd, it might affect rwnd calculation.
A dilemma is what does "are in sequence for a particular stream" entail? There is some stipulation about delaying delivery to the upper layer until packages are reordered (see Section 6.6, below), but reordering doesn't seem to be conditioned by filling the gaps at the level of the association. Also note the mention in Section 6.2 on the complex distinction between ACK and delivery to the ULP (Upper Layer Protocol).
Whether other stipulations of the RFC indirectly result in the occurence of HOL, and whether it is effective in real-life implementations and situations - these questions warrant further investigation.
Below are some of the excerpts which I've come across in the RFC and which may be relevant.
RFC 4960, Section 6.2 Acknowledgement on Reception of DATA Chunks
When the receiver's advertised window is 0, the receiver MUST drop any new incoming DATA chunk with a TSN larger than the largest TSN received so far. If the new incoming DATA chunk holds a TSN value less than the largest TSN received so far, then the receiver SHOULD drop the largest TSN held for reordering and accept the new incoming DATA chunk. In either case, if such a DATA chunk is dropped, the receiver MUST immediately send back a SACK with the current receive window showing only DATA chunks received and accepted so far. The dropped DATA chunk(s) MUST NOT be included in the SACK, as they were not accepted.
Under certain circumstances, the data receiver may need to drop DATA chunks that it has received but hasn't released from its receive buffers (i.e., delivered to the ULP). These DATA chunks may have been acked in Gap Ack Blocks. For example, the data receiver may be holding data in its receive buffers while reassembling a fragmented user message from its peer when it runs out of receive buffer space. It may drop these DATA chunks even though it has acknowledged them in Gap Ack Blocks. If a data receiver drops DATA chunks, it MUST NOT include them in Gap Ack Blocks in subsequent SACKs until they are received again via retransmission. In addition, the endpoint should take into account the dropped data when calculating its a_rwnd.
Circumstances which highlight how senders may receive acknowledgement for chunks which are ultimately not delivered to the ULP (Upper Layer Protocol).Note this applies to chunks with TSN higher than the Cumulative TSN (i.e. from Gap Ack Blocks). This together with unreliability of SACK order represent good reasons for the stipulation in Section 7.1 (see below).
RFC 4960, Section 6.6 Ordered and Unordered Delivery
Within a stream, an endpoint MUST deliver DATA chunks received with the U flag set to 0 to the upper layer according to the order of their Stream Sequence Number. If DATA chunks arrive out of order of their Stream Sequence Number, the endpoint MUST hold the received DATA chunks from delivery to the ULP until they are reordered.
This is the only stipulation on ordered delivery within a stream in this section; seemingly, reordering does not depend on filling the gaps in ACK-ed chunks.
RFC 4960, Section 7.1 SCTP Differences from TCP Congestion Control
Gap Ack Blocks in the SCTP SACK carry the same semantic meaning as the TCP SACK. TCP considers the information carried in the SACK as advisory information only. SCTP considers the information carried in the Gap Ack Blocks in the SACK chunk as advisory. In SCTP, any DATA chunk that has been acknowledged by SACK, including DATA that arrived at the receiving end out of order, is not considered fully delivered until the Cumulative TSN Ack Point passes the TSN of the DATA chunk (i.e., the DATA chunk has been acknowledged by the Cumulative TSN Ack field in the SACK).
This is stated from the perspective of the sending endpoint, and is accurate for the reason emphasized in section 6.6 above.
Note: TCP guarantees in-sequence delivery of data to its upper-layer protocol within a single TCP session. This means that when TCP notices a gap in the received sequence number, it waits until the gap is filled before delivering the data that was received with sequence numbers higher than that of the missing data. On the other hand, SCTP can deliver data to its upper-layer protocol even if there is a gap in TSN if the Stream Sequence Numbers are in sequence for a particular stream (i.e., the missing DATA chunks are for a different stream) or if unordered delivery is indicated. Although this does not affect cwnd, it might affect rwnd calculation.
This seems to be the core answer to what interests you.
In support of this argument, the format of the SCTP SACK chunk as exposed here and here.
Related
I have read already lot of specs and code about UART, but I cannot find any indication on how to find by software interface if the transmit FIFO is full. There is an interrupt when the FIFO is empty. Then I can write at least N characters, where N is the fifo size. But when I have written these N characters, a number of them have already been sent. So I can in fact write more than N characters, but there is no FIFO full interrupt. The specs says that when the fifo is full indeed the TXREADY pin on the chip is inverted. Is there a way to find this by software ? The Line Status Register bit only says that the fifo is not empty, which does not mean it is full...
Anyone can help ? I want to write characters until the fifo is full...
Looks to me also that they neglected this, but most people get by with the thing as it is. The usual way to use it is to get an interrupt, fill the FIFO (normally very fast compared to serial data rate) and then return.
There is a situation where it seems to me that what you are asking for could be nice...if transmitting in a polling mode...you want to send 10 bytes, your polling shows the FIFO is not empty, so you have not way to know if you can send them all or not...either you wait there until it is empty, which sort of defeats the purpose of the FIFO, or you continue polling other stuff until you get back to checking for FIFO empty, and maybe that slows your overall transmission rate. Guess it is not a very usual way to operate, so nobody worries about it.
The 16550D datasheet says the following:
The transmitter holding register interrupt (02) occurs when the XMIT
FIFO is empty; it is cleared as soon as the transmitter holding
register is written to (1 to 16 characters may be written to the XMIT
FIFO while servicing this interrupt) or the IIR is read.
This means that when the Line Status Register register (port base + 5) indicates Transmitter Empty condition (in bit 5), the transmit FIFO is completely empty and you may write up to 16 bytes to the transmitter holding register (port base + 0). It is important not to write more than 16 bytes between occurrences of the transmitter empty bit being set.
If you don't need to write 16 bytes at the point when you received the IRQ (or saw the transmitter register empty bit set, if polling), you can either keep track of how many bytes you wrote since the last transmitter empty state, or, just defer writing further bytes until the next transmitter empty state.
How can I handle buffer overflow in ndis driver. Can anybody tell some buffer overflow scenarios or some use cases of buffer overflow conditions.
For NDIS miniport drivers
If you receive a packet that is larger than the MTU, discard it. Do not indicate the packet up to NDIS (i.e., do not pass the packet to NdisMIndicateReceiveNetBufferLists). If possible, increment the ifInErrors statistical counter.
The above rule is not affected by the NDIS_PACKET_TYPE_PROMISCUOUS flag; do not indicate excessively-large packets even when in promiscuous mode. However, you should indicate excessively-small (aka "runt") packets when in promiscuous mode, if your hardware permits it.
If you are asked to transmit a packet that is larger than the MTU, do not attempt to transmit it. Assign NET_BUFFER_LIST::Status = NDIS_STATUS_INVALID_LENGTH and return the NBL back to NDIS with NdisMSendNetBufferListsComplete. (I wouldn't expect you to ever see such a packet; it would be a bug for NDIS to attempt to send you such a packet.)
For NDIS protocol drivers
If you receive a packet that is larger than the MTU, you are free to discard it.
Never attempt to send a packet that is larger than the MTU.
For NDIS filter drivers
If a filter receives a packet that is larger than the MTU (FilterReceiveNetBufferLists), the filter may immediately discard the packet (NdisFReturnNetBufferLists if the receive indication is not made with NDIS_RECEIVE_FLAGS_RESOURCES, or just returning immediately if the resources flag is set).
If a filter is asked to send a packet that is larger than the MTU (FilterSendNetBufferLists), the filter may assign NET_BUFFER_LIST::Status = NDIS_STATUS_INVALID_LENGTH and return the packet immediately (NdisFSendNetBufferListsComplete).
Filters are not obligated to validate the size of every packet that passes through them. However, your filter should validate the size of any packets where a malformed packet would otherwise cause your filter to trigger a buffer overflow. For example, if your filter copies all ARP replies into a pre-allocated buffer, first validate that the ARP reply isn't too large to fit into the buffer. (This is not strictly necessary, since the miniport "shouldn't" give you an excessively-large packet. However, you are on the network datapath, which means you're handling untrusted data being processed by a potentially-buggy miniport. A little extra defense-in-depth is a good idea.)
Filters must not originate packets that are larger than the MTU (on either the send or receive paths).
I am about to write a message protocol going over a TCP stream. The receiver needs to know where the message boundaries are.
I can either send 1) fixed length messages, 2) size fields so the receiver knows how big the message is, or 3) a unique message terminator (I guess this can't be used anywhere else in the message).
I won't use #1 for efficiency reasons.
I like #2 but is it possible for the stream to get out of sync?
I don't like idea #3 because it means receiver can't know the size of the message ahead of time and also requires that the terminator doesn't appear elsewhere in the message.
With #2, if it's possible to get out of sync, can I add a terminator or am I guaranteed to never get out of sync as long as the sender program is correct in what it sends? Is it necessary to do #2 AND #3?
Please let me know.
Thanks,
jbu
You are using TCP, the packet delivery is reliable. So the connection either drops, timeouts or you will read the whole message.
So option #2 is ok.
I agree with sigjuice.
If you have a size field, it's not necessary to add and end-of-message delimiter --
however, it's a good idea.
Having both makes things much more robust and easier to debug.
Consider using the standard netstring format, which includes both a size field and also a end-of-string character.
Because it has a size field, it's OK for the end-of-string character to be used inside the message.
If you are developing both the transmit and receive code from scratch, it wouldn't hurt to use both length headers and delimiters. This would provide robustness and error detection. Consider the case where you just use #2. If you write a length field of N to the TCP stream, but end up sending a message which is of a size different from N, the receiving end wouldn't know any better and end up confused.
If you use both #2 and #3, while not foolproof, the receiver can have a greater degree of confidence that it received the message correctly if it encounters the delimiter after consuming N bytes from the TCP stream. You can also safely use the delimiter inside your message.
Take a look at HTTP Chunked Transfer Coding for a real world example of using both #2 and #3.
Depending on the level at which you're working, #2 may actually not have an issues with going out of sync (TCP has sequence numbering in the packets, and does reassemble the stream in correct order for you if it arrives out of order).
Thus, #2 is probably your best bet. In addition, knowing the message size early on in the transmission will make it easier to allocate memory on the receiving end.
Interesting there is no clear answer here. #2 is generally safe over TCP, and is done "in the real world" quite often. This is because TCP guarantees that all data arrives both uncorrupted* and in the order that it was sent.
*Unless corrupted in such a way that the TCP checksum still passes.
Answering to old message since there is stuff to correnct:
Unlike many answers here claim, TCP does not guarantee data to arrive uncorrupted. Not even practically.
TCP protocol has a 2-byte crc-checksum that obviously has a 1:65536 chance of collision if more than one bit flips. This is such a small chance it will never be encountered in tests, but if you are developing something that either transmits large amounts of data and/or is used by very many end users, that dice gets thrown trillions of times (not kidding, youtube throws it about 30 times a second per user.)
Option 2: size field is the only practical option for the reasons you yourself listed. Fixed length messages would be wasteful, and delimiter marks necessitate running the entire payload through some sort of encoding-decoding stage to replace at least three different symbols: start-symbol, end-symbol, and the replacement-symbol that signals replacement has occurred.
In addition to this one will most likely want to use some sort of error checking with a serious checksum. Probably implemented in tandem with the encryption protocol as a message validity check.
As to the possibility of getting out of sync:
This is possible per message, but has a remedy.
A useful scheme is to start each message with a header. This header can be quite short (<30 bytes) and contain the message payload length, eventual correct checksum of the payload, and a checksum for that first portion of the header itself. Messages will also have a maximum length. Such a short header can also be delimited with known symbols.
Now the receiving end will always be in one of two states:
Waiting for new message header to arrive
Receiving more data to an ongoing message, whose length and checksum are known.
This way the receiver will in any situation get out of sync for at most the maximum length of one message. (Assuming there was a corrupted header with corruption in message length field)
With this scheme all messages arrive as discrete payloads, the receiver cannot get stuck forever even with maliciously corrupted data in between, the length of arriving payloads is know in advance, and a successfully transmitted payload has been verified by an additional longer checksum, and that checksum itself has been verified. The overhead for all this can be a mere 26 byte header containing three 64-bit fields, and two delimiting symbols.
(The header does not require replacement-encoding since it is expected only in a state whout ongoing message, and the entire 26 bytes can be processed at once)
There is a fourth alternative: a self-describing protocol such as XML.
I need to calculate total data transfer while transferring a fixed size data from client to server in TCP/IP. It includes connecting to the server, sending request,header, receiving response, receiving data etc.
More precisely, how to get total data transfer while using POST and GET method?
Is there any formula for that? Even a theoretical one will do fine (not considering packet loss or connection retries etc)
FYI I tried RFC2616 and RFC1180. But those are going over my head.
Any suggestion?
Thanks in advance.
You can't know the total transfer size in advance, even ignoring retransmits. There are several things that will stop you:
TCP options are negotiated between the hosts when the connection is established. Some options (e.g., timestamp) add additional data to the TCP header
"total data transfer size" is not clear. Ethernet, for example, adds quite a few more bits on top of whatever IP used. 802.11 (wireless) will add even more. So do HDLC or PPP going over a T1. Don't even think about frame relay. Some links may use compression (which will reduce the total size). The total size depends on where you measure it, even for a single packet.
Assuming you're just interested in the total octet size at layer 2, and you know the TCP options that will be negotiated in advance, you still can't know the path MTU. Which may change, even while the connection is in progress. Or if you're not doing path MTU discovery (which would be wierd), then the packet may get fragmented somewhere, and the remote end will see a different amount of data transfer than you.
I'm not sure why you need to know this, but I suggest that:
If you just want an estimate, watch a typical connection in Wireshark. Calculate the percent overhead (vs. the size of data you gave to TCP, and received from TCP). Use that number to estimate: it will be close enough, except in pathological situations.
If you need to know for sure how much data your end saw transmitted and received, use libpcap to capture the packet stream and check.
i'd say on average that request and response have about 8 lines of headers each and about 30 chars per line. Then allow for the size increase of converting any uploaded binary to Base64.
You didn't say if you also want to count TCP packet headers, in which case you could assume an MTU of about 1500 so add 16 bytes (tcp header) per 1500 data bytes
Finally, you could always setup a packet sniffer and count actual bytes for a sample of data.
oh yeah, and you may need to allow for deflate/gzip encoding as well.
As stated in RFC 3286:
"...endpoints must manage the conversion between bytes sent and received and TSNs sent and received, since TSN is per chunk rather than per byte".
How does this affect the congestion control algorithm?
There are two reasons:
1. Pragmatically, RFC 3286 refers RFC 2581 for most of the congestion control, and it works in bytes.
2. Practically, and this is a stronger reason, there needs to be a buffer assigned at each end and these would be hard to define in terms of TSNs (chunks) since these are variably size. This would either mean over-allocating space in the buffer e.g. 64K * TSNs, or using a dynamically allocated list. The former is wasteful of space, the latter relatively slow.
Does this answer your question, or was it more related to your last question?