QUIC Working GroupJ. Iyengar, Editor
Internet-DraftI. Swett, Editor
Intended status: Standards TrackGoogle
Expires: November 22, 2017May 21, 2017

partnersuche online österreich QUIC Loss Detection and Congestion Control

draft-ietf-quic-recovery-03

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top free dating sites in germany QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades of transport and security experience, and implements mechanisms that make it attractive as a modern general-purpose transport. The QUIC protocol is described in [QUIC-TRANSPORT].how to chat free online in india

free mobile dating sites in germany QUIC implements the spirit of known TCP loss recovery mechanisms, described in RFCs, various Internet-drafts, and also those prevalent in the Linux TCP implementation. This document describes QUIC congestion control and loss recovery, and where applicable, attributes the TCP equivalent in RFCs, Internet-drafts, academic papers, and/or TCP implementations.chat to therapist online free

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singles in deiner stadt All transmissions in QUIC are sent with a packet-level header, which includes a packet sequence number (referred to below as a packet number). These packet numbers never repeat in the lifetime of a connection, and are monotonically increasing, which makes duplicate detection trivial. This fundamental design decision obviates the need for disambiguating between transmissions and retransmissions and eliminates significant complexity from QUIC’s interpretation of TCP loss detection mechanisms.chatroom usa free

free online chat dating service Every packet may contain several frames. We outline the frames that are important to the loss detection and congestion control machinery below.free live chat with astrologer online

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chat online free dating indonesia TCP conflates transmission sequence number at the sender with delivery sequence number at the receiver, which results in retransmissions of the same data carrying the same sequence number, and consequently to problems caused by “retransmission ambiguity”. QUIC separates the two: QUIC uses a packet sequence number (referred to as the “packet number”) for transmissions, and any data that is to be delivered to the receiving application(s) is sent in one or more streams, with stream offsets encoded within STREAM frames inside of packets that determine delivery order.meet chat friends online free

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top ten dating sites in germany QUIC ACKs contain information that is equivalent to TCP SACK, but QUIC does not allow any acked packet to be reneged, greatly simplifying implementations on both sides and reducing memory pressure on the sender.frauen suchen männer facebook

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top ten dating site in germany QUIC ACKs explicitly encode the delay incurred at the receiver between when a packet is received and when the corresponding ACK is sent. This allows the receiver of the ACK to adjust for receiver delays, specifically the delayed ack timer, when estimating the path RTT. This mechanism also allows a receiver to measure and report the delay from when a packet was received by the OS kernel, which is useful in receivers which may incur delays such as context-switch latency before a userspace QUIC receiver processes a received packet.free to chat online dating sites


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per chat nach date fragen QUIC uses a combination of ack information and alarms to detect lost packets. An unacknowledged QUIC packet is marked as lost in one of the following ways:chat with lawyer online free india

  • A packet is marked as lost if at least one packet that was sent a threshold number of packets (kReorderingThreshold) after it has been acknowledged. This indicates that the unacknowledged packet is either lost or reordered beyond the specified threshold. This mechanism combines both TCP’s FastRetransmit and FACK mechanisms.
  • If a packet is near the tail, where fewer than kReorderingThreshold packets are sent after it, the sender cannot expect to detect loss based on the previous mechanism. In this case, a sender uses both ack information and an alarm to detect loss. Specifically, when the last sent packet is acknowledged, the sender waits a short period of time to allow for reordering and then marks any unacknowledged packets as lost. This mechanism is based on the Linux implementation of TCP Early Retransmit.
  • If a packet is sent at the tail, there are no packets sent after it, and the sender cannot use ack information to detect its loss. The sender therefore relies on an alarm to detect such tail losses. This mechanism is based on TCP’s Tail Loss Probe.
  • If all else fails, a Retransmission Timeout (RTO) alarm is always set when any retransmittable packet is outstanding. When this alarm fires, all unacknowledged packets are marked as lost.
  • Instead of a packet threshold to tolerate reordering, a QUIC sender may use a time threshold. This allows for senders to be tolerant of short periods of significant reordering. In this mechanism, a QUIC sender marks a packet as lost when a packet larger than it is acknowledged and a threshold amount of time has passed since the packet was sent.
  • Handshake packets, which contain STREAM frames for stream 0, are critical to QUIC transport and crypto negotiation, so a separate alarm period is used for them.

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free online psychic chat no credit card Constants used in loss recovery are based on a combination of RFCs, papers, and common practice. Some may need to be changed or negotiated in order to better suit a variety of environments.chat room psychic free

kMaxTLPs (default 2):
Maximum number of tail loss probes before an RTO fires.
kReorderingThreshold (default 3):
Maximum reordering in packet number space before FACK style loss detection considers a packet lost.
kTimeReorderingFraction (default 1/8):
Maximum reordering in time space before time based loss detection considers a packet lost. In fraction of an RTT.
kMinTLPTimeout (default 10ms):
Minimum time in the future a tail loss probe alarm may be set for.
kMinRTOTimeout (default 200ms):
Minimum time in the future an RTO alarm may be set for.
kDelayedAckTimeout (default 25ms):
The length of the peer’s delayed ack timer.
kDefaultInitialRtt (default 100ms):
The default RTT used before an RTT sample is taken.

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free chat room indiana Variables required to implement the congestion control mechanisms are described in this section.yahoo chat room pakistani online

loss_detection_alarm:
Multi-modal alarm used for loss detection.
handshake_count:
The number of times the handshake packets have been retransmitted without receiving an ack.
tlp_count:
The number of times a tail loss probe has been sent without receiving an ack.
rto_count:
The number of times an rto has been sent without receiving an ack.
largest_sent_before_rto:
The last packet number sent prior to the first retransmission timeout.
smoothed_rtt:
The smoothed RTT of the connection, computed as described in [RFC6298]
rttvar:
The RTT variance, computed as described in [RFC6298]
reordering_threshold:
The largest delta between the largest acked retransmittable packet and a packet containing retransmittable frames before it’s declared lost.
time_reordering_fraction:
The reordering window as a fraction of max(smoothed_rtt, latest_rtt).
loss_time:
The time at which the next packet will be considered lost based on early transmit or exceeding the reordering window in time.
sent_packets:
An association of packet numbers to information about them, including a number field indicating the packet number, a time field indicating the time a packet was sent, and a bytes field indicating the packet’s size. sent_packets is ordered by packet number, and packets remain in sent_packets until acknowledged or lost.

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free chat room indian At the beginning of the connection, initialize the loss detection variables as follows:google chat for windows phone 8.1

   loss_detection_alarm.reset()
   handshake_count = 0
   tlp_count = 0
   rto_count = 0
   if (UsingTimeLossDetection())
     reordering_threshold = infinite
     time_reordering_fraction = kTimeReorderingFraction
   else:
     reordering_threshold = kReorderingThreshold
     time_reordering_fraction = infinite
   loss_time = 0
   smoothed_rtt = 0
   rttvar = 0
   largest_sent_before_rto = 0

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pool games with chat After any packet is sent, be it a new transmission or a rebundled transmission, the following OnPacketSent function is called. The parameters to OnPacketSent are as follows:ch frauen suchen männer

  • packet_number: The packet number of the sent packet.
  • is_retransmittable: A boolean that indicates whether the packet contains at least one frame requiring reliable deliver. The retransmittability of various QUIC frames is described in [QUIC-TRANSPORT]. If false, it is still acceptable for an ack to be received for this packet. However, a caller MUST NOT set is_retransmittable to true if an ack is not expected.
  • sent_bytes: The number of bytes sent in the packet.

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 OnPacketSent(packet_number, is_retransmittable, sent_bytes):
   sent_packets[packet_number].packet_number = packet_number
   sent_packets[packet_number].time = now
   if is_retransmittable:
     sent_packets[packet_number].bytes = sent_bytes
     SetLossDetectionAlarm()

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   OnAckReceived(ack):
     // If the largest acked is newly acked, update the RTT.
     if (sent_packets[ack.largest_acked]):
       rtt_sample = now - sent_packets[ack.largest_acked].time
       if (rtt_sample > ack.ack_delay):
         rtt_sample -= ack.delay
       UpdateRtt(rtt_sample)
     // The sender may skip packets for detecting optimistic ACKs
     if (packets acked that the sender skipped):
       abortConnection()
     // Find all newly acked packets.
     for acked_packet in DetermineNewlyAckedPackets():
       OnPacketAcked(acked_packet.packet_number)

     DetectLostPackets(ack.largest_acked_packet)
     SetLossDetectionAlarm()


   UpdateRtt(rtt_sample):
     // Based on {{RFC6298}}.
     if (smoothed_rtt == 0):
       smoothed_rtt = rtt_sample
       rttvar = rtt_sample / 2
     else:
       rttvar = 3/4 * rttvar + 1/4 * (smoothed_rtt - rtt_sample)
       smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * rtt_sample

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   OnPacketAcked(acked_packet_number):
     // If a packet sent prior to RTO was acked, then the RTO
     // was spurious.  Otherwise, inform congestion control.
     if (rto_count > 0 &&
         acked_packet_number > largest_sent_before_rto)
       OnRetransmissionTimeoutVerified()
     handshake_count = 0
     tlp_count = 0
     rto_count = 0
     sent_packets.remove(acked_packet_number)

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pakistani chat rooms without registration Pseudocode for SetLossDetectionAlarm follows:free dating site in hamburg germany

 SetLossDetectionAlarm():
    if (retransmittable packets are not outstanding):
      loss_detection_alarm.cancel()
      return

    if (handshake packets are outstanding):
      // Handshake retransmission alarm.
      if (smoothed_rtt == 0):
        alarm_duration = 2 * kDefaultInitialRtt
      else:
        alarm_duration = 2 * smoothed_rtt
      alarm_duration = max(alarm_duration, kMinTLPTimeout)
      alarm_duration = alarm_duration * (2 ^ handshake_count)
    else if (loss_time != 0):
      // Early retransmit timer or time loss detection.
      alarm_duration = loss_time - now
    else if (tlp_count < kMaxTLPs):
      // Tail Loss Probe
      if (retransmittable_packets_outstanding = 1):
        alarm_duration = 1.5 * smoothed_rtt + kDelayedAckTimeout
      else:
        alarm_duration = kMinTLPTimeout
      alarm_duration = max(alarm_duration, 2 * smoothed_rtt)
    else:
      // RTO alarm
      alarm_duration = smoothed_rtt + 4 * rttvar
      alarm_duration = max(alarm_duration, kMinRTOTimeout)
      alarm_duration = alarm_duration * (2 ^ rto_count)

    loss_detection_alarm.set(now + alarm_duration)

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   OnLossDetectionAlarm():
     if (handshake packets are outstanding):
       // Handshake retransmission alarm.
       RetransmitAllHandshakePackets()
       handshake_count++
     else if (loss_time != 0):
       // Early retransmit or Time Loss Detection
       DetectLostPackets(largest_acked_packet)
     else if (tlp_count < kMaxTLPs):
       // Tail Loss Probe.
       SendOnePacket()
       tlp_count++
     else:
       // RTO.
       if (rto_count == 0)
         largest_sent_before_rto = largest_sent_packet
       SendTwoPackets()
       rto_count++

     SetLossDetectionAlarm()

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pakistani chat rooms free online no registration Packets in QUIC are only considered lost once a larger packet number is acknowledged. DetectLostPackets is called every time an ack is received. If the loss detection alarm fires and the loss_time is set, the previous largest acked packet is supplied.fb chat application samsung mobile

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   DetectLostPackets(largest_acked):
     loss_time = 0
     lost_packets = {}
     delay_until_lost = infinite
     if (time_reordering_fraction != infinite):
       delay_until_lost =
         (1 + time_reordering_fraction) * max(latest_rtt, smoothed_rtt)
     else if (largest_acked.packet_number == largest_sent_packet):
       // Early retransmit alarm.
       delay_until_lost = 9/8 * max(latest_rtt, smoothed_rtt)
     foreach (unacked < largest_acked.packet_number):
       time_since_sent = now() - unacked.time_sent
       packet_delta = largest_acked.packet_number - unacked.packet_number
       if (time_since_sent > delay_until_lost):
         lost_packets.insert(unacked)
       else if (packet_delta > reordering_threshold)
         lost_packets.insert(unacked)
       else if (loss_time == 0 && delay_until_lost != infinite):
         loss_time = now() + delay_until_lost - time_since_sent

     // Inform the congestion controller of lost packets and
     // lets it decide whether to retransmit immediately.
     if (!lost_packets.empty())
       OnPacketsLost(lost_packets)
       foreach (packet in lost_packets)
         sent_packets.remove(packet.packet_number)

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kDefaultMss (default 1460 bytes):
The default max packet size used for calculating default and minimum congestion windows.
kInitialWindow (default 10 * kDefaultMss):
Default limit on the amount of outstanding data in bytes.
kMinimumWindow (default 2 * kDefaultMss):
Default minimum congestion window.
kLossReductionFactor (default 0.5):
Reduction in congestion window when a new loss event is detected.

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bytes_in_flight:
The sum of the size in bytes of all sent packets that contain at least one retransmittable frame, and have not been acked or declared lost.
congestion_window:
Maximum number of bytes in flight that may be sent.
end_of_recovery:
The packet number after which QUIC will no longer be in recovery.
ssthresh
Slow start threshold in bytes. When the congestion window is below ssthresh, it grows by the number of bytes acknowledged for each ack.

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   congestion_window = kInitialWindow
   bytes_in_flight = 0
   end_of_recovery = 0
   ssthresh = infinite

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   OnPacketAcked(acked_packet):
     if (acked_packet.packet_number < end_of_recovery):
       return
     if (congestion_window < ssthresh):
       congestion_window += acket_packets.bytes
     else:
       congestion_window +=
           acked_packets.bytes / congestion_window

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   OnPacketsLost(lost_packets):
     largest_lost_packet = lost_packets.last()
     // Start a new recovery epoch if the lost packet is larger
     // than the end of the previous recovery epoch.
     if (end_of_recovery < largest_lost_packet.packet_number):
       end_of_recovery = largest_sent_packet
       congestion_window *= kLossReductionFactor
       congestion_window = max(congestion_window, kMinimumWindow)
       ssthresh = congestion_window

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   OnRetransmissionTimeoutVerified()
     congestion_window = kMinimumWindow

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   TimeToSend(packet_size):
     if (bytes_in_flight + packet_size > congestion_window)
       return infinite
     return time_of_last_sent_packet +
         (packet_size * smoothed_rtt) / congestion_window

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free dating site in australia 2013 References

free dating sites in 2013 Normative References

[QUIC-TRANSPORT]
Iyengar, J., Ed. and M. Thomson, Ed., “QUIC: A UDP-Based Multiplexed and Secure Transport”, Internet-Draft draft-ietf-quic-transport-latest (work in progress).
[RFC2119]
Bradner, S., “free online hindi chat rooms”, BCP 14, RFC 2119, online indian chat rooms, March 1997, <free live astrology chat online>.

how to chat online in india Informative References

[LOSS-PROBE]
Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis, “play free online rpg sim dating games”, Internet-Draft draft-dukkipati-tcpm-tcp-loss-probe-01 (work in progress), February 2013.
[RFC5682]
Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata, “play yaoi dating sim game online free”, RFC 5682, chat to doctor online free uk, September 2009, <live chat to a doctor online for free uk>.
[RFC5827]
Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and P. Hurtig, “chat doctor online gratis”, RFC 5827, free online dating sites in germany, May 2010, <100 percent free german dating sites>.
[RFC6298]
Paxson, V., Allman, M., Chu, J., and M. Sargent, “google chat for windows phone 8”, RFC 6298, free online german dating sites, June 2011, <free online dating in germany>.
[RFC6582]
Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, “online dating websites in germany”, RFC 6582, free online singles chat rooms, April 2012, <free online singles chat room no registration>.

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  • Integrate F-RTO (#544, #409)
  • Add congestion control (#545, #395)
  • Require connection abort if a skipped packet was acknowledged (#415)
  • Simplify RTO calculations (#142, #417)

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  • Overview added to loss detection
  • Changes initial default RTT to 100ms
  • Added time-based loss detection and fixes early retransmit
  • Clarified loss recovery for handshake packets
  • Fixed references and made TCP references informative

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  • Improved description of constants and ACK behavior

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  • Adopted as base for draft-ietf-quic-recovery
  • Updated authors/editors list
  • Added table of contents

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