Re: [PATCH v6 net-next 0/2] tcp: Redundant Data Bundling (RDB)

Bendik Rønning Opstad
2016-03-13T23:18:43Z

On 03/10/2016 01:20 AM, Yuchung Cheng wrote:
> I read the paper. I think the underlying idea is neat. but the
> implementation is little heavy-weight that requires changes on fast
> path (tcp_write_xmit) and space in skb control blocks.

Yuchung, thank you for taking the time to review the patch submission
and read the paper.

I must admit I was not particularly happy about the extra if-test on the
fast path, and I fully understand the wish to keep the fast path as
simple and clean as possible.
However, is the performance hit that significant considering the branch
prediction hint for the non-RDB path?

The extra variable needed in the SKB CB does not require increasing the
CB buffer size due to the "tcp: refactor struct tcp_skb_cb" patch:
http://patchwork.ozlabs.org/patch/510674 and uses only some of the space
made available in the outgoing SKBs' CB. Therefore I hoped the extra
variable would be acceptable.

> ultimately this
> patch is meant for a small set of specific applications.

Yes, the RDB mechanism is aimed at a limited set of applications,
specifically time-dependent applications that produce non-greedy,
application limited (thin) flows. However, our hope is that RDB may
greatly improve TCP's position as a viable alternative for applications
transmitting latency sensitive data.

> In my mental model (please correct me if I am wrong), losses on these
> thin streams would mostly resort to RTOs instead of fast recovery, due
> to the bursty nature of Internet losses.

This depends on the transmission pattern of the applications, which
varies to a great deal, also between the different types of
time-dependent applications that produce thin streams. For short flows,
(bursty) loss at the end will result in an RTO (if TLP does not probe),
but the thin streams are often long lived, and the applications
producing them continue to write small data segments to the socket at
intervals of tens to hundreds of milliseconds.

What controls if an RTO and not fast retransmit will resend the packet,
is the number of PIFs, which directly correlates to how often the
application writes data to the socket in relation to the RTT. As long as
the number of packets successfully completing a round trip before the
RTO is >= the dupACK threshold, they will not depend on RTOs (not
considering TLP). Early retransmit and the TCP_THIN_DUPACK socket option
will also affect the likelihood of RTOs vs fast retransmits.

> The HOLB comes from RTO only
> retransmit the first (tiny) unacked packet while a small of new data is
> readily available. But since Linux congestion control is packet-based,
> and loss cwnd is 1, the new data needs to wait until the 1st packet is
> acked which is for another RTT.

If I understand you correctly, you are referring to HOLB on the sender
side, which is the extra delay on new data that is held back when the
connection is CWND-limited. In the paper, we refer to this extra delay
as increased sojourn times for the outgoing data segments.

We do not include this additional sojourn time for the segments on the
sender side in the ACK Latency plots (Fig. 4 in the paper). This is
simply because the pcap traces contain the timestamps when the packets
are sent, and not when the segments are added to the output queue.

When we refer to the HOLB effect in the paper as well as the thesis, we
refer to the extra delays (sojourn times) on the receiver side where
segments are held back (not made available to user space) due to gaps in
the sequence range when packets are lost (we had no reordering).

So, when considering the increased delays due to HOLB on the receiver
side, HOLB is not at all limited to RTOs. Actually, it's mostly not due
to RTOs in the tests we've run, however, this also depends very much on
the transmission pattern of the application as well as loss levels.
In general, HOLB on the receiver side will affect any flow that
transmits a packet with new data after a packet is lost (sender may not
know yet), where the lost packet has not already been retransmitted.

Consider a sender application that performs write calls every 30 ms on a
150 ms RTT link. It will need a CWND that allows 5-6 PIFs to be able to
transmit all new data segments with no extra sojourn times on the sender
side.
When one packet is lost, the next 5 packets that are sent will be held
back on the receiver side due to the missing segment (HOLB). In the best
case scenario, the first dupACK triggers a fast retransmit around the
same time as the fifth packet (after the lost packet) is sent. In that
case, the first segment sent after the lost segment is held back on the
receiver for 150 ms (the time it takes for the dupACK to reach the
sender, and the fast retrans to arrive at the receiver). The second is
held back 120 ms, the third 90 ms, the fourth 60 ms, an the fifth 30 ms.

All of this extra delay is added before the sender even knows there was
a loss. How it decides to react to the loss signal (dupACKs) will
further decide how much extra delays will be added in addition to the
delays already inflicted on the segments by the HOLB.

> Instead what if we only perform RDB on the (first and recurring) RTO
> retransmission?

That will change RDB from being a proactive mechanism, to being
reactive, i.e. change how the sender responds to the loss signal. The
problem is that by this point (when the sender has received the loss
signal), the HOLB on the receiver side has already caused significant
increases to the application layer latency.

The reason the RDB streams (in red) in fig. 4 in the paper get such low
latencies is because there are almost no retransmissions. With 10%
uniform loss, the latency for 90% of the packets is not affected at all.
The latency for most of the lost segments is only increased by 30 ms,
which is when the next RDB packet arrives at the receiver with the lost
segment bundled in the payload.
For the regular TCP streams (blue), the latency for 40% of the segments
is affected, where almost 30% of the segments have additional delays of
150 ms or more.
It is important to note that the increases to the latencies for the
regular TCP streams compared to the RDB streams are solely due to HOLB
on the receiver side.

The longer the RTT, the greater the gains are by using RDB, considering
the best case scenario of minimum one RTT required for a retransmission.
As such, RDB will reduce the latencies the most for those that also need
it the most.

However, even with an RTT of 20 ms, an application writing a data
segment every 10 ms will still get significant latency reductions simply
because a retransmission will require a minimum of 20 ms, compared to
the 10 ms it takes for the next RDB packet to arrive at the receiver.

Bendik