Effective Troubleshooting of Network Congestion and Packet Loss: Techniques and Solutions

 

Introduction

Network congestion is a critical challenge in today’s digital age, where the demand for data transmission continues to grow exponentially. As more users and devices connect to networks, the likelihood of congestion increases, causing slowdowns and decreased efficiency in communication systems. Addressing network congestion is therefore essential to maintain optimal performance and user satisfaction.

In this issue, we’re discussing a common network problem—packet loss. For example, when we ping a website, if the ping is successful and the website returns comprehensive information, it indicates that communication with the website server is smooth. If the ping fails or the website returns incomplete information, it is likely due to packet loss. Such situations are probably familiar to everyone. This article provides some common troubleshooting methods for network packet loss, aiming to enhance your understanding of packet loss. Don’t panic when facing packet loss; let’s explore and increase our knowledge together…

What is packet loss?

Packet loss occurs when one or more packets of data traveling across a computer network fail to reach their destination. This can result from a variety of issues such as network congestion, hardware failure, software bugs, or faulty network devices. Packet loss is significant because it can lead to reduced throughput, increased latency, or even complete disruption of network communications, which affects performance, particularly in real-time applications like video conferencing and online gaming.

Data is transmitted over the Internet in units called packets, measured in bytes. During network transmission, due to factors such as network equipment and network quality, the received data can be less than the sent data, resulting in packet loss.

Packet reception and transmission principles

Send Packet:

• 1. The application’s data packet adds a TCP header at the TCP layer to form a transmissible packet.
• 2. An IP header is added at the IP layer to create an IP packet.
• 3. The network card driver appends a 14-byte MAC header to the IP packet, forming a frame (temporarily without CRC). The frame contains the MAC addresses of both the sender and receiver.
• 4. The driver copies the frame (temporarily without CRC) to the network card’s buffer, where it is processed by the network card.
• 5. The network card adds synchronization information and CRC checks to the frame, encapsulating it as a packet ready for transmission. The packet is then sent onto the network cable, enabling all network cards connected to the cable to detect the packet. This completes the transmission of an IP packet.

Receive data packet:

• 1. The network card receives a packet from the network cable, first checking the packet’s CRC checksum to ensure integrity, then removes the packet header to obtain the frame. (The network card will check if the destination MAC address in the MAC packet matches its own MAC address; if not, it will discard the packet.)
• 2. The network card copies the frame into a pre-allocated ring buffer.
• 3. The network card driver notifies the kernel to process the data, which is decoded and processed through the layers of the TCP/IP protocol stack.
• 4. The application retrieves the data from the socket buffer.

Core Concepts

Understanding the principles of packet transmission and reception allows us to identify that the main reasons for packet loss are typically related to network card devices, network card drivers, and the kernel protocol stack, categorized into three major areas. Below, we will introduce the topic by following the principle of “layered analysis from the bottom up (analyzing potential packet loss scenarios at each layer), then examining critical information to ultimately derive the analytical results.”

Table of Contents — Overview of Network Packet Loss Scenarios

Hardware network card packet loss

> Packet loss caused by network adapter driver

> Ethernet Link Layer Packet Loss

> Network IP Layer Packet Loss

> Transport Layer UDP/TCP Packet Loss

> Packet loss at the application layer socket

Provide detailed explanations for the above six scenarios as follows~

Packet loss on hardware network card

Ring Buffer Overflow

As illustrated, when data frames arrive on the physical medium, they are first read by the NIC (Network Interface Card) and written into the internal buffer of the device, known as the Ring Buffer. The Softirq is then triggered by the interrupt handler to consume from it. The size of the Ring Buffer varies depending on the network card device. When the rate at which network packets arrive (production) is faster than the rate at which the kernel processes them (consumption), the Ring Buffer will quickly fill up, and newly arriving packets will be discarded.

View:

Using `ethtool` or `/proc/net/dev`, you can view the statistics of packets dropped due to the Ring Buffer being full. In these statistics, it is indicated with `fifo`:

$ ethtool -S eth0|grep rx_fifo
rx_fifo_errors: 0
$ cat /proc/net/dev
Inter-|Receive | Transmitface |bytes packets errs drop fifo frame compressed 
multicast|bytes packets errs drop fifo colls carrier compressed
eth0: 17253386680731 42839525880 0 0 0 0 0 244182022 14879545018057 41657801805 0 0 0 0 0 0

# Check the Maximum and Current Settings of the eth0 Network Card Ring Buffer

$ ethtool -g eth0

Solution: Adjusting the Hardware Buffer Size for Receiving and Sending on Network Card eth0

$ ethtool -G eth0 rx 4096 tx 4096

Network card port negotiation packet loss

1. Check network card packet loss statistics: `ethtool -S eth1/eth0`

2. Check the network interface card configuration status: ethtool eth1/eth0

Primarily check if the negotiation rate and mode of the network card and upstream network equipment meet expectations.

Solutions:

1 Re-negotiate: `ethtool -r eth1/eth0;`

2. If the upstream does not support auto-negotiation, you can forcibly set the port speed:

ethtool -s eth1 speed 1000 duplex full autoneg off

Network Card Flow Control Packet Loss

1. View Traffic Control Statistics:

ethtool -S eth1 | grep control

`rx_flow_control_xon` is the count of pause frames sent to the switch port to enable flow control when the RX Buffer of the network card is full or other internal resources of the network card are limited. Correspondingly, `tx_flow_control_xoff` is the count of pause frames sent to disable flow control after resources become available.

2. Check network flow control configuration: ethtool -a eth1

Solution: Disable NIC Flow Control

ethtool -A ethx autoneg off // Auto-negotiation disabled  
ethtool -A ethx tx off // Transmission module disabled  
ethtool -A ethx rx off // Reception module disabled

 

Packet MAC Address Packet Loss

Typically, a computer’s network interface card (NIC) operates in non-promiscuous mode. In this mode, the NIC only accepts data packets from the network port that are addressed to its own destination address. If the packet’s destination MAC address does not match the interface’s MAC address, the packet is usually discarded. This situation might likely occur when the source has a static ARP table entry set, or the dynamically learned ARP table entry isn’t updated promptly, yet the destination MAC address has changed (e.g., the network card has been replaced) without an update notification reaching the source (such as when the update packet is lost, or there’s an issue with the intermediate switch).

View:

1. Capturing packets on the destination side: tcpdump can activate promiscuous mode, allowing it to capture the corresponding packets, after which you can check the MAC address;

2. From the source side, examine the ARP table or capture packets (on the next-hop device) to verify whether the MAC address being sent matches the MAC address of the next-hop destination.

Solution:

1. Refresh the ARP table and then send packets to trigger ARP relearning (this may affect other packets and increase latency, so proceed with caution);

2. You can manually set the correct static ARP entries on the source end;

Other network cards experiencing abnormal packet loss

This type of anomaly is less common, but if it doesn’t fall into any of the aforementioned situations and there are still packet loss counts in the network card statistics, you can try to troubleshoot the following:

Network card firmware version:

Verify if there is a bug in the network card’s PHY chip firmware, and check if the installed version meets expectations. View `ethtool -i eth1`:

Contact the manufacturer to inquire if it’s a known issue and whether there is a new version, etc.;

Loose Ethernet Cable Connection:

If there is an increasing count of CRC errors in the network card statistics, it is likely due to a loose connection in the network cable. You may notify the network administrator to investigate:

ethtool -S eth0

Solution: Generally, try unplugging and reconnecting the network cable, or replace it with a new one. Check if the socket meets port specifications, etc.;

Packet Length Packet Loss

The network adapter has a correct packet length range for reception, typically the normal Ethernet packet length range is between 64-1518 bytes. Under normal circumstances, the sender will pad or fragment to fit. Occasionally, some abnormal situations may occur, leading to packet loss during transmission.

View:

ethtool -S eth1|grep length_errors

Solution:

1 Adjust the MTU configuration of the interface, whether to enable support for Ethernet jumbo frames;

2 The sender enables PATH MTU for appropriate fragmentation;

A brief summary of network card packet loss:

Network card driver packet loss

Inspect: ifconfig eth1/eth0 and similar interfaces

1. RX errors: Indicates the total number of received packet errors, including too-long-frames errors, Ring Buffer overflow errors, CRC check errors, frame synchronization errors, FIFO overruns, and missed packages, among others.

2. RX dropped: This indicates that a data packet has entered the Ring Buffer but was discarded during the memory copy process due to system reasons like insufficient memory.

3. RX overruns: This indicates FIFO overruns, which occur when the I/O transmitted by the Ring Buffer (also known as the Driver Queue) exceeds the I/O that the kernel can handle. The Ring Buffer refers to the buffer before an IRQ request is initiated. Clearly, an increase in overruns implies that packets are being discarded by the physical layer of the network card before they even reach the Ring Buffer. One reason the Ring Buffer fills up is the inability of the CPU to promptly handle interrupts. The problematic machine mentioned earlier was experiencing packet loss due to an uneven distribution of interrupts (all concentrated on core0) and a lack of affinity configuration.

4. RX frame: Indicates misaligned frames.

5. Regarding TX, the main reasons for the increase in the aforementioned counter include aborted transmission, errors due to carrier, FIFO error, heartbeat errors, and window error. Meanwhile, collisions indicate transmission interruptions caused by CSMA/CD.

Driver overflow packet loss

`netdev_max_backlog` is a kernel parameter that defines the buffer queue for packets received from the NIC before they are processed by the protocol stack (such as IP, TCP). Each CPU core has its own backlog queue, similar to a Ring Buffer. When the rate of incoming packets exceeds the processing capacity of the kernel protocol stack, the CPU’s backlog queue will continue to grow. Once it reaches the configured `netdev_max_backlog` value, packets will be dropped.

View:

By examining /proc/net/softnet_stat, you can determine whether a netdev backlog queue overflow has occurred:

Wherein: Each line represents the state statistics for each CPU core, starting from CPU0 onwards; each column represents various statistics for a single CPU core: The first column denotes the total number of packets received by the interrupt handler; the second column indicates the total number of packets dropped due to netdev_max_backlog queue overflow. From the above output, it can be seen that, in this server statistics, there are no packet drops caused by netdev_max_backlog.

Solution:

The default value of netdev_max_backlog is 1000. On high-speed links, the second statistic mentioned above might not be zero. This can be resolved by modifying the kernel parameter net.core.netdev_max_backlog:

$ sysctl -w net.core.netdev_max_backlog=2000

Single-core load is high, leading to packet loss.

A high soft interrupt load on a single-core CPU causes applications to have limited opportunities to send or receive packets, or results in slow packet processing. Even after adjusting the size of the netdev_max_backlog queue, packet drops still occur after a period of time because processing speed cannot keep up with the network card’s receiving speed.

Check: `mpstat -P ALL 1`

Single-core soft interrupt occupies 100%, resulting in applications having no chance to send/receive or experience slow packet handling, leading to packet loss.

SolutionIt appears there isn’t any text content for translation provided in your message. If you have WordPress post content you would like to be translated, please paste the text here, and I’ll assist you with the translation while preserving any HTML tags or formatting.

1. Adjust Network Card RSS Queue Configuration:

View: ethtool -x ethx;

Adjustment: ethtool -X ethx xxxx;

2. Check if the network card interrupt configuration is balanced by using `cat /proc/interrupts`.

Adjustment:

1) IRQ balance adjustment:
# Check current status
service irqbalance status
# Terminate service
service irqbalance stop

2) Bind interrupt to CPU core echo mask > /proc/irq/xxx/smp_affinity

3. Adjust the multi-queue configuration of the network card and the RPS (Receive Packet Steering) settings based on the number of CPU and network card queues.

-CPU count exceeds the number of NIC queues:

Check the network card queue with `ethtool -x ethx`;

Enabling RPS in the protocol stack and configuring RPS;

echo $mask (CPU configuration) > /sys/class/net/$eth/queues/rx-$i/rps_cpus
echo 4096 (NIC buff) > /sys/class/net/$eth/queues/rx-$i/rps_flow_cnt
2) If the CPU is smaller than the number of NIC queues, you can bind the interrupt. You can try to disable RPS to see the effect:
echo 0 > /sys/class/net/<dev>/queues/rx-<n>/rps_cpus

4. By adjusting the NUMA CPU alignment and the positioning of network cards, the kernel processing speed can be improved. This allows more CPU resources to be allocated to applications for packet reception, thus reducing the likelihood of packet loss.

Check NUMA location of the network card:

ethtool -i eth1|grep bus-info
lspci -s bus-info -vv|grep node

The above interruption and RPS settings need to have the mask reconfigured according to the NUMA CPU allocation.

5. You can try enabling interrupt coalescing (check if your network card supports it).

View:

 ethtool -c ethx
Coalesce parameters for eth1:
Adaptive RX: on  TX: on
stats-block-usecs: 0
sample-interval: 0
pkt-rate-low: 0
pkt-rate-high: 0

rx-usecs: 25
rx-frames: 0
rx-usecs-irq: 0
rx-frames-irq: 256

tx-usecs: 25
tx-frames: 0
tx-usecs-irq: 0
tx-frames-irq: 256

rx-usecs-low: 0
rx-frame-low: 0
tx-usecs-low: 0
tx-frame-low: 0

rx-usecs-high: 0
rx-frame-high: 0
tx-usecs-high: 0
tx-frame-high: 0

Adjustments:

ethtool -C ethx adaptive-rx on

A Brief Summary of Handling Packet Loss in Network Card Drivers:

 

Kernel protocol stack packet loss

Ethernet Link Layer Packet Loss

neighbor system ARP packet loss

The configuration for `arp_ignore` is causing packet loss.

The purpose of the arp_ignore parameter is to control whether the system sends an ARP response when it receives an external ARP request. Common values for the arp_ignore parameter are primarily 0, 1, 2, and values from 3 to 8 are less frequently used.

Check: `sysctl -a | grep arp_ignore`

Solution: Set corresponding values according to the actual scenario;

0: Responds to any ARP requests received on the machine’s IP address across all network interfaces (including those on the loopback interface), regardless of whether the destination IP is on the receiving interface.

1: Only respond to ARP requests where the destination IP address matches a local address on the receiving network interface card.

2: Only respond to ARP requests if the destination IP address matches the local address on the receiving network card, and the source IP of the ARP request must be on the same subnet as the receiving network card.

3: If the IP address requested by the ARP request packet corresponds to a local address with a scope of host, do not respond with an ARP reply packet. If the scope is global or link, then respond with an ARP reply packet.

Configuration of `arp_filter` and Packet Loss

In systems with multiple interfaces (such as in the case of Tencent Cloud’s elastic network interface), each of these interfaces can respond to ARP requests. This might cause the remote end to learn different MAC addresses, leading to potential packet loss if the MAC address used for sending subsequent packets does not match the MAC address of the interface receiving the packet. The arp_filter is primarily used to adapt to this scenario.

View:

sysctl -a | grep arp_filter

Solution:

Set the corresponding values based on the actual scenario, generally, this filtering rule is turned off by default; it can be enabled under special circumstances.  
0: Default value, indicating that when responding to ARP requests, the interface condition is not checked.  
1: Indicates that when responding to ARP requests, the interface is checked to see if it matches the receiving request interface; if they do not match, no response is sent.

ARP table overflow resulting in packet loss

For example, in a situation like this, where there are suddenly a large number of ARP entries exceeding the default configuration of the protocol stack, some ARP creations fail during packet transmission, resulting in transmission failure and thus packet loss.

View:

• View ARP status: `cat /proc/net/stat/arp_cache`, table_fulls statistics:

• Check dmesg messages (kernel logs):

dmesg|grep neighbour
neighbour: arp_cache: neighbor table overflow!

• Check the current ARP table size: `ip n|wc -l`

Check the system quotas:

sysctl -a | grep net.ipv4.neigh.default.gc_thresh  
gc_thresh1: This is the minimum number of entries that must exist in the ARP cache for the garbage collector not to run. The default value is 128.

gc_thresh2: This is the soft upper limit on the maximum number of entries that can be stored in the ARP cache. The garbage collector will allow the number of entries to exceed this value for 5 seconds before starting collection. The default value is 512.

gc_thresh3: This is the hard upper limit on the maximum number of entries that can be stored in the ARP cache. Once the number of entries in the cache exceeds this threshold, the garbage collector will run immediately. The default value is 1024.

Typically, when there is sufficient memory, it can be assumed that the `gc_thresh3` value represents the total size of the ARP table.

SolutionAdjust the ARP table size according to the actual maximum ARP value situation (such as the maximum number of other sub-devices accessed).

$ sudo sysctl -w net.ipv4.neigh.default.gc_thresh1=1024
$ sudo sysctl -w net.ipv4.neigh.default.gc_thresh2=2048
$ sudo sysctl -w net.ipv4.neigh.default.gc_thresh3=4096
$ sudo sysctl  -p

ARP request cache queue overflow packet loss

View:

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Solution: Adjust Cache Queue Size According to Customer Needs `unres_qlen_bytes`:

 

The IP layer packet loss in networks

Interface IP address configuration packet loss

1. If the local service is unreachable, check whether the lo interface is configured with the address 127.0.0.1;

2. Local reception failure, check the local routing table: `ip r show table local | grep submachine IP address`; this type of packet loss typically occurs in multi-IP scenarios. A failure in the underlying configuration of multiple IPs on the submachine results in the corresponding IP not receiving packets, leading to packet loss.

Solution:

1. Configure the correct interface IP address; for example, ip a add 1.1.1.1 dev eth0

2. If you discover that an interface has an address but is still experiencing packet loss, it might be because the local routing table lacks the corresponding entry. In an emergency, you can manually add it:

For example, `ip r add local your_ip_address dev eth0 table local`;

 

Packet loss in routing

Route configuration packet loss

See:

1. Check the configuration to see if the routing is correctly set up (whether it’s reachable), and whether policy routing is configured (this configuration might appear in Elastic Network Interface scenarios) ip rule:

Then locate the corresponding routing table. View the routing table:

Or directly use `ip r get x.x.x.x` to let the system help you determine if there is a reachable route and whether the interface meets expectations;

2. View system statistics:

netstat -s|grep "dropped because of missing route"

Solution: Reconfigure the correct routing;

Reverse path filtering packet loss

Reverse path filtering is a mechanism in Linux that utilizes reverse path lookup to check whether the source IP of a received packet is routable (Loose mode) or is the best route (Strict mode). If the validation fails, the packet is discarded. The purpose of this design is to prevent IP address spoofing attacks.

View:

`rp_filter` offers three modes for configuration:

0 – No Verification

1 – RFC3704 defines strict mode: For each received packet, perform a reverse path lookup. If the packet’s entry point does not match the reverse path’s exit point, it will be dropped.

2 – Loose Mode as Defined by RFC3704: For each received packet, perform a reverse path lookup. If the packet is unreachable from any interface, it is not forwarded.

View the current rp_filter policy configuration:

$cat /proc/sys/net/ipv4/conf/eth0/rp_filter

If this is set to 1, you need to check the network environment and routing policy of the host to determine whether they may cause inbound packets from the client to fail reverse-path verification.

From a theoretical standpoint, this mechanism operates at the network layer. Therefore, if the client can successfully ping the server, this factor can be ruled out.

Solution:

Set the rp_filter to 0 or 2 according to the actual network environment:

To translate the provided text:

- The command `$ sysctl -w net.ipv4.conf.all.rp_filter=2` is used to set the reverse path filtering mode for all network interfaces.
- Alternatively, `$ sysctl -w net.ipv4.conf.eth0.rp_filter=2` sets this mode specifically for the `eth0` network interface.

There is no plain text to translate; these are command instructions for configuring reverse path filtering on a system.

 

Packet loss in firewall

Client’s rule configuration caused packet loss.

View:

  iptables -nvL |grep DROP ;

Solution: Modify firewall rules;

Connection tracking leads to packet loss.

Connection tracking table overflow packet loss

The kernel uses the `ip_conntrack` module to track the state of network packets handled by `iptables`, saving each record in a table (this table is in memory and can be viewed via `/proc/net/ip_conntrack` to check the current total recorded). If network conditions are busy, such as high connections or high concurrency, it can gradually occupy the available space in this table. Generally, this table is large and not easily filled, as it can self-purge. Entries in the table remain there, occupying space until the source IP sends an RST packet. However, during scenarios like an attack, network misconfigurations, faulty routing/routers, or problematic network cards, the RST packet from the source IP might not be received, leading to accumulation in the table. This accumulation grows until it fills the table completely. Regardless of what causes the table to become full, once it is filled, packet loss occurs, and external connections to the server fail. The kernel then logs the error message: `kernel: ip_conntrack: table full, dropping packet;`.

View Current Connection Tracking Count :

cat /proc/sys/net/netfilter/nf_conntrack_max

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Increase the maximum number of tracked connections  
`net.netfilter.nf_conntrack_max = 3276800`  
Reduce the maximum lifespan of tracked connections  
`net.netfilter.nf_conntrack_tcp_timeout_established = 1200`  
`net.netfilter.nf_conntrack_udp_timeout_stream = 180`  
`net.netfilter.nf_conntrack_icmp_timeout = 30`

The creation of CT conflicts leads to packet loss.

View: Current connection tracking statistics: `cat /proc/net/stat/nf_conntrack`, where you can check various connection tracking anomalies and packet loss statistics.

Solution: Kernel hot patch fix or update the kernel version (incorporate patch modifications);

UDP/TCP Packet Loss in the Transport Layer

TCP connection tracking security check packet loss

Packet loss reason: Since the connection remains open, but there have been packet transmission anomalies between the server and client (such as a segment not updating the connection tracking module window count), the legitimate window range has not been updated, leading to packet loss during subsequent security checks. The protocol stack uses `nf_conntrack_tcp_be_liberal` to control this option:

1: Disabled, only RST packets that are outside of the TCP window are flagged as invalid;

0: Enable; All packets not within the TCP window are marked as invalid;

View:

Inspect Configuration:

sysctl -a|grep nf_conntrack_tcp_be_liberal 
net.netfilter.nf_conntrack_tcp_be_liberal = 1

Check the log:

Under normal circumstances, the netfiler module does not load logs by default and needs to be loaded manually.

modprobe ipt_LOG11
sysctl -w net.netfilter.nf_log.2=ipt_LOG

Then check the syslog after sending the packet;

Solution: Based on the actual packet capture analysis, determine whether this mechanism is causing packet loss. You can try disabling it to see if that resolves the issue.

Fragmented reassembly packet loss

Situation Summary:Timeout

View:

netstat -s|grep timeout
601 fragments dropped after timeout

Solution: Adjust the Timeout Duration

net.ipv4.ipfrag_time = 30
sysctl -w net.ipv4.ipfrag_time=60

frag_high_thresh, when the fragmented memory exceeds a certain threshold, it can cause system security checks to drop packets.

View:

netstat -s|grep reassembles
8094 packet reassembles failed

Solution: Resize

net.ipv4.ipfrag_high_thresh 
net.ipv4.ipfrag_low_thresh

Fragmentation security distance check to prevent packet loss

View:

netstat -s|grep reassembles
8094 packet reassembles failed

Solution: Set `ipfrag_max_dist` to 0 to disable this security check.

The `pfrag_max_dist` feature, in certain scenarios, is actually not applicable:

1. There is a large amount of network packet interaction.

2. The concurrency level at the sending end is very high, and coupled with the SMP architecture, this can easily lead to such out-of-order situations.

Shard hash bucket conflict chain too long exceeds the system default value of 128.

View:

dmesg|grep “Dropping fragment”
inet_frag_find: Fragment hash bucket 128 list length grew over limit. Dropping fragment.

Solution: Hotfix adjustment to hash size;

System memory insufficient, failed to create a new shard queue.

Viewing Methods:

netstat -s|grep reassembles
8094 packet reassembles failed

dropwatch packet drop location:

Solution:

a. Increase system network memory:

net.core.rmem_default 
net.core.rmem_max 
net.core.wmem_default

b. System Memory Reclamation:

In an emergency, you can use /proc/sys/vm/drop_caches to free up virtual memory.

To free pagecache:
# echo 1 > /proc/sys/vm/drop_caches
To free dentries and inodes:
# echo 2 > /proc/sys/vm/drop_caches
To free pagecache, dentries and inodes:
echo 3 > /proc/sys/vm/drop_caches

 

Packet Loss in MTU

View:

1. Check the MTU configuration for the interface using `ifconfig eth1/eth0`, the default is 1500;

2. Conduct MTU probing and then configure the MTU value corresponding to the interface;

Solution:

1. Set the correct MTU value based on actual conditions;

2. Configure an appropriate TCP MSS and enable TCP MTU Probe:

cat /proc/sys/net/ipv4/tcp_mtu_probing:
tcp_mtu_probing - INTEGER Controls TCP Packetization-Layer Path MTU Discovery.
Takes three values:
0 - Disabled 
1 - Disabled by default, enabled when an ICMP black hole detected
2 - Always enabled, use initial MSS of tcp_base_mss.

Packet loss at the TCP layer

An excessive number of TIME_WAIT states can lead to packet loss.

A large number of TIMEWAIT states appear and need to be resolved in a scenario on a high-concurrency short-connection TCP server. When the server completes processing a request and immediately initiates a proper active closure of the connection… In this scenario, many sockets will be in the TIMEWAIT state. If the client’s concurrency level remains very high, some clients may then experience connection failures.

View:

Check system log:

dmsg TCP: time-wait bucket table overflow;

Review System Configuration:

sysctl -a|grep tcp_max_tw_buckets
net.ipv4.tcp_max_tw_buckets = 16384

Solution:

1. `tw_reuse` and `tw_recycle` are only effective when client and server timestamps are enabled (default is on).

2. tw_reuse only affects the client. Once enabled, the client will reclaim within 1 second;

3. `tw_recycle` functions on both the client and server sides. Once enabled, it recycles within 3.5*RTO. The RTO ranges from 200ms to 120s, depending on network conditions. In internal network scenarios, it performs slightly faster than `tw_reuse`, while in public networks, especially mobile networks, it is generally slower than `tw_reuse`. The advantage lies in its ability to recycle the TIME_WAIT sockets on the server side.

On the server side, if the network path traverses a NAT node, do not enable `net.ipv4.tcp_tw_recycle`, as it will lead to timestamp conflicts and cause additional packet loss issues.

4. Adjust the size of `tcp_max_tw_buckets` if enough memory is available:

It appears you have provided a line of sysctl command syntax. The command adjusts kernel parameters at runtime. Here’s a breakdown:

- `sysctl`: A tool for examining and changing kernel parameters at runtime.
- `-w`: Indicates a write operation, meaning it will modify a parameter.
- `net.ipv4.tcp_max_tw_buckets=163840`: Sets the maximum number of TCP time-wait sockets to 163840.

If you need further explanation regarding this command or its impact on system performance and security, feel free to ask!

Timestamp anomaly packet loss

When multiple clients are in the same NAT environment and access the server simultaneously, the timing for different clients may not match. In such cases, the server might receive requests from the same NAT with timestamp discrepancies, leading to subsequent packets being dropped. This usually manifests as the client sending a SYN, but the server not responding with an ACK. On the server, you can use the following command to verify whether packets are continuously being dropped.

Inspect:

netstat -s | grep rejects

Solution:

If the network path will traverse a NAT node, do not enable `net.ipv4.tcp_tw_recycle`.

TCP queue issues causing packet loss

Principle:

The TCP State Machine (Three-Way Handshake)

Protocol Handling:

One is the semi-connection queue (syn queue):

In the three-way handshake protocol, the server maintains a half-open connection queue, which allocates an entry for each client’s SYN packet (when the server receives the SYN packet, it creates a request_sock structure that is stored in the half-open connection queue). This entry indicates that the server has received the SYN packet and sent an acknowledgment back to the client, waiting for the client’s acknowledgment packet (initiating the second handshake by sending a SYN+ACK packet to confirm). Connections identified by these entries are in the Syn_RECV state on the server. Once the server receives the client’s acknowledgment packet, it deletes the entry, and the server moves to the ESTABLISHED state. This queue is known as the SYN queue, and its length is max(64, /proc/sys/net/ipv4/tcp_max_syn_backlog). The machine’s tcp_max_syn_backlog value is configured under /proc/sys/net/ipv4/tcp_max_syn_backlog.

One is the fully connected queue (accept queue):

During the third handshake, when the server receives the ACK packet, it enters a new queue called the accept queue. The length of this queue is min(backlog, somaxconn). By default, the value of somaxconn is 128, indicating that there can be a maximum of 129 ESTABLISHED connections waiting for accept(). The value of backlog should be specified by the second parameter in the function int listen(int sockfd, int backlog), and in listen, the backlog can be defined by our application.

View:

Connection establishment failed, SYN packet loss:

netstat -s |grep -i listen
SYNs to LISTEN sockets dropped

It may also be affected by connection saturation and packet loss.

Solution: Increase the size of `tcp_max_syn_backlog`

Connection experiencing high packet loss

-xxx times the listen queue of a socket overflowed

View:

• Check the accept queue size: net.core.somaxconn
• ss -lnt command to query the socket queue: LISTEN state: Recv-Q indicates the current number of pending requests that have completed the three-way handshake and are waiting for the server’s accept call, i.e., when a client connects to a listening server through connect(), these connections remain in this queue until accepted by the server; Send-Q, on the other hand, represents the maximum listen backlog value, which is the result of min(backlog, somaxconn) as mentioned above.
• Check if it’s the application setting a limit, `int listen(int sockfd, int backlog);`.

Solution:

• Optimizing the Linux kernel parameters can help alleviate pressure. Set `tcp_abort_on_overflow=1`.
• Modify the net.core.somaxconn size;
• Application configuration issue, notify client of program modification;

SYN flood attack packet loss

Currently, on Linux, the default setting is to retransmit SYN-ACK packets five times. The retry interval starts at 1 second, and each subsequent retry interval doubles from the previous one. The retry intervals for the five attempts are 1s, 2s, 4s, 8s, and 16s, totaling 31 seconds. After the fifth attempt, there is an additional 32-second wait to confirm the fifth attempt has also timed out. Thus, it takes a total of 1s + 2s + 4s + 8s + 16s + 32s = 63 seconds for TCP to terminate the connection. Since SYN timeouts require 63 seconds, this provides an opportunity for attackers to exploit the server. Attackers may launch a SYN flood attack by sending a large number of SYN packets to the server in a short time, with the intent to exhaust the server’s SYN queue. To address the issue of excessive SYN requests…

View: View syslog: kernel: [3649830.269068] TCP: Possible SYN flooding on port xxx. Sending cookies. Check SNMP counters.

Solution:

• Increase `tcp_max_syn_backlog`.
• Reduce tcp_synack_retries
• Enable TCP syncookies
• Enable `tcp_abort_on_overflow`, set `tcp_abort_on_overflow` to 1. A setting of 1 indicates that if the fully established connection queue is full during the third step, the server will send a reset packet to the client, indicating to terminate the handshake process and abandon this connection (since this connection has not yet been fully established on the server side).

PAWS mechanism packet loss

Principle: PAWS (Protect Against Wrapped Sequence numbers). In environments with high bandwidth, TCP sequence numbers may be reused (recycled/wrapped) within a short period. This can potentially result in two legitimate data packets and their acknowledgment packets within the same TCP stream having identical sequence numbers in a short time frame.

View:

$netstat -s |grep -e "passive connections rejected because of time 
stamp" -e "packets rejects in established connections because of 
timestamp” 
387158 passive connections rejected because of time stamp
825313 packets rejects in established connections because of timestamp

Check if `tcp_tw_recycle` and `tcp_timestamp` are enabled using sysctl:

$ sysctl net.ipv4.tcp_tw_recycle
net.ipv4.tcp_tw_recycle = 1 
$ sysctl net.ipv4.tcp_timestamps
net.ipv4.tcp_timestamps = 1

1. The `tcp_tw_recycle` parameter. It is used for quickly reclaiming TIME_WAIT connections, but it can cause issues in NAT environments;

2. When multiple clients connect to the internet using NAT and interact with the server, the server perceives them as having the same IP. This means, from the server’s perspective, these clients are essentially identical. Unfortunately, due to possible timestamp discrepancies among these clients, the server might observe timestamp disorder. This can directly lead to packets with smaller timestamps being discarded. If such issues occur, the typical manifestation is that even though the client clearly sends a SYN, the server does not respond with an ACK.

Solution:

In a NAT environment, disable the TCP timestamp option or do not enable the tcp_tw_recycle parameter.

Packet loss issue with TLP

TLP primarily aims to address the issue of the efficiency of tail packet loss retransmission. TLP can effectively avoid prolonged RTO timeouts, thereby improving TCP performance. For detailed information, please refer to the article:

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However, in low latency scenarios (short connection with small packet volume), the combination of TLP (Tail Loss Probe) and delayed ACK may result in ineffective retransmissions, leading the client to detect a large number of false retransmission packets, thereby increasing response latency.

View:

View Protocol Stack Statistics:

netstat -s |grep TCPLossProbes

Check system configuration:

 sysctl -a | grep tcp_early_retrans

Solution:

1. Disable delayed ACK and enable quick ACK;

2. In Linux, implementing the nodelay semantic does not equate to fast acknowledgment; it merely disables the Nagle algorithm.

3. Enable the quick ACK option. Inside the socket, there’s a TCP_QUICKACK option that needs to be set again after each recv.

Insufficient memory causing packet loss

View:

View log:

dmesg|grep “out of memory”

View System Configuration:

cat /proc/sys/net/ipv4/tcp_mem
cat /proc/sys/net/ipv4/tcp_rmem
cat /proc/sys/net/ipv4/tcp_wmem

Solution:

Based on concurrent TCP traffic, adjust system parameters, generally by attempting to increase them by a factor of two or another multiple to see if the issue is mitigated.

sysclt -w net.ipv4.tcp_mem=
sysclt -w net.ipv4.tcp_wmem=
sysclt -w net.ipv4.tcp_rmem=
sysctl -p

TCP timeout packet loss

View:

Conduct a packet capture analysis of network RTT:

Consider using other tools like hping to test the current end-to-end network quality.

# hping -S 9.199.10.104 -A
HPING 9.199.10.104 (bond1 9.199.10.104): SA set, 40 headers + 0 data bytes
len=46 ip=9.199.10.104 ttl=53 DF id=47617 sport=0 flags=R seq=0 win=0 rtt=38.3 ms
len=46 ip=9.199.10.104 ttl=53 DF id=47658 sport=0 flags=R seq=1 win=0 rtt=38.3 ms
len=46 ip=9.199.10.104 ttl=53 DF id=47739 sport=0 flags=R seq=2 win=0 rtt=30.4 ms
len=46 ip=9.199.10.104 ttl=53 DF id=47842 sport=0 flags=R seq=3 win=0 rtt=30.4 ms
len=46 ip=9.199.10.104 ttl=53 DF id=48485 sport=0 flags=R seq=4 win=0 rtt=38.7 ms
len=46 ip=9.199.10.104 ttl=53 DF id=49274 sport=0 flags=R seq=5 win=0 rtt=34.1 ms
len=46 ip=9.199.10.104 ttl=53 DF id=49491 sport=0 flags=R seq=6 win=0 rtt=30.3 ms

Solution:

• Disable Nagle’s Algorithm to Reduce Small Packet Latency;
• Disabling Delayed ACK:

  sysctl -w net.ipv4.tcp_no_delay_ack=1

TCP Out-of-Order Packet Loss

At this point, TCP cannot determine whether packet loss or reordering has occurred, since both scenarios result in the receiving end processing packets in a disordered manner, leading to data gaps. TCP will tentatively diagnose this situation as packet reordering, as reordering is a matter of timing (potentially late packet arrival), whereas packet loss implies the need for retransmission. When TCP detects packet reordering, it will promptly send an ACK, where the TSER part contains a TSEV value that records the moment the receiver last received an ordered segment. This increases the RTT sample value for the packets, which in turn prolongs the RTO time. This is undoubtedly beneficial for TCP, as it allows ample time to discern whether a packet is reordered or lost, preventing unnecessary data retransmissions. However, severe reordering can lead the sender to assume packet loss; once the duplicate ACKs exceed TCP’s threshold, it will trigger the timeout retransmission mechanism to promptly resolve the issue. For more details, please refer to the blog:

https://blog.csdn.net/dog250/article/details/78692585

View: Analyze packet capture for potential out-of-order packets:

Solution: In scenarios with multipath transmission or poor network quality, you can enhance the system’s tolerance for out-of-order TCP transmissions by modifying the following values:

 

Congestion Control Packet Loss

During the evolution of the Internet, the TCP algorithm has also undergone certain changes, evolving through various stages.

Reno, NewReno, Cubic, and Vegas—these improved algorithms can generally be categorized into loss-based and delay-based congestion control algorithms. Loss-based congestion control algorithms, represented by Reno and NewReno, have primary issues such as buffer bloat and long fat pipes. Loss-based protocols utilize a passive congestion control mechanism that assesses network congestion based on packet loss events. Even if network load is high, the protocol will not proactively reduce its transmission speed as long as there’s no congestion-induced packet loss. Initially, routers had relatively small buffers at the forwarding exit, which could cause global synchronization and reduce bandwidth utilization when TCP was used. As router manufacturers decreased hardware costs, they continuously increased buffer sizes, allowing loss-based feedback protocols to occupy router buffers persistently without packet loss. This increased bandwidth utilization but also led to greater network jitter after congestion-induced packet losses.

The problem of long fat pipes, characterized by high bandwidth and RTT, presents a high likelihood of random packet loss. TCP’s default buffer setting is relatively small, and random packet loss can frequently decrease the congestion window (cwnd), resulting in low bandwidth utilization.

BBR (Bottleneck Bandwidth and Round-trip propagation time) is a bandwidth and delay feedback-based congestion control algorithm. It has already evolved to the second version and exemplifies a typical closed-loop feedback system where the amount of data sent and the rate of sending these data packets are continuously adjusted based on each feedback cycle. Before BBR, congestion control algorithms were event-driven, necessitating packet loss or delay events as triggers. With BBR’s introduction, congestion control became an autonomous automatic control algorithm based on feedback. The rate control is determined by the algorithm rather than network events. The core of the BBR algorithm is determining two parameters: maximum bandwidth (Max BW) and minimum round-trip time (Min RTT). The product of the maximum bandwidth and the minimum round-trip time yields the BDP (Bandwidth Delay Product), which represents the maximum capacity for data storage in the network link. The BDP drives the probing state machine to determine the rate quantum and cwnd, which, when set in the sending engine, addresses the issues of transmission speed and data volume.

The Linux 4.9 kernel first introduced the BBR (Bottleneck Bandwidth and Round-trip propagation time) congestion control algorithm. The initial version of BBR has stronger packet loss resilience compared to other algorithms. However, this version encounters issues under certain scenarios (drawbacks), particularly in the realm of real-time audio and video, where BBR does not perform as well in deep queue scenarios compared to Cubic.

The issue observed is: In deep queue scenarios, during the ProbeRTT phase of BBR, only 4 packets are sent. The considerable reduction in transmission rate can lead to increased latency and stuttering issues.

View:

ss -sti // On the source side ss -sti|grep 10.125.42.49:47699 -A 3 (10.125.42.49:47699 is the destination address and port number)

Solution:

• ProbeRTT is not suitable for real-time audio and video applications, so it can be either removed directly or, like BBRV2, the probe RTT can be shortened to once every 2.5 seconds, using 0.5xBDP for sending.
• If there are no special requirements, switch to the stable cubic algorithm;

Packet loss at the UDP layer

Packet transmission failed, packet loss.

View: netstat Statistics

If there are continuous receive buffer errors/send buffer errors counts;

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1. CPU load (multi-core affinity configuration), network load (soft interrupt optimization, adjusting driver queues netdev_max_backlog), memory configuration (protocol stack memory);
2. Increase buffer cache size during peak times:

net.ipv4.udp_mem = xxx
net.ipv4.udp_rmem_min = xxx
net.ipv4.udp_wmem_min = xxx

3. Adjust Application Design:

• UDP itself is a connectionless and unreliable protocol, suitable for scenarios where occasional packet loss does not affect the program state, such as video, audio, gaming, and monitoring. Applications with high reliability requirements for packets should not use UDP; it is recommended to use TCP instead. Of course, you can also implement retries and deduplication at the application layer to ensure reliability.
• To address packet loss on a server, first check the system load through monitoring tools. If the load is too high, try to reduce it and then see if the packet loss issue resolves.
• If the system load is too high, there is no effective solution for UDP packet loss. If an application anomaly is causing high CPU, memory, or IO usage, promptly identify and fix the problematic application. If resources are insufficient, monitoring should be able to detect this in a timely manner and facilitate rapid scaling.
• For systems that send or receive large volumes of UDP packets, adjusting the system and application’s socket buffer size can reduce the likelihood of packet loss.
• When handling UDP packets, the application should use an asynchronous approach, ensuring that there isn’t too much processing logic between receiving packets.

 

Packet loss at the application layer socket

Socket receive buffer packet loss

View:

1. Conduct packet capture analysis to check for packet loss;

2. View Statistics:

netstat -s|grep "packet receive errors"

Solution:

Adjusting Socket Buffer Size:

Socket configuration (all protocol sockets):
# Default Socket Receive Buffer
net.core.rmem_default = 31457280
# Maximum Socket Receive Buffer
net.core.rmem_max = 67108864

Principles of adjusting specific dimensions:

The buffer size doesn’t have any optimal set value, as the ideal size varies depending on the specific circumstances.

Principle of Buffer Size Estimation: In data communication, the bandwidth-delay product refers to the product of a data link’s capacity (in bits per second) and the round-trip communication delay (in seconds). The result is a total amount of data, measured in bits (or bytes), equivalent to the maximum amount of data that can be on the network line at any given time—data that has been sent but not yet acknowledged.

It looks like you’re referencing a formula related to networking, specifically the Bandwidth-Delay Product (BDP). This formula is used to calculate the amount of data that can be in transit in the network. The formula is:

\[ \text{BDP} = \text{Bandwidth} \times \text{RTT} \]

Where:
– Bandwidth is the data transfer rate, typically measured in bits per second (bps).
– RTT (Round-Trip Time) is the time it takes for a signal to go from the sender to the receiver and back, usually measured in seconds.

If you have any specific questions or need further explanation, feel free to ask!

It is possible to estimate BDP, namely the buffer size, by calculating the bandwidth of the current node and averaging the latency. You can refer to the estimation in common scenarios below:

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Application settings TCP connection count packet loss

View:

Please refer to the above TCP connection queue analysis;

Solution:

Set an appropriate connection queue size. During the third handshake, when the server receives the ACK, it enters a new queue called “accept.” The length of this queue is defined by `min(backlog, somaxconn)`. By default, the `somaxconn` value is 128, indicating that a maximum of 129 ESTABLISHED connections can wait for `accept()`. The `backlog` value should be specified by the second parameter in `int listen(int sockfd, int backlog)`, which can be defined by our application within the `listen`.

Rapid application sending results in packet loss.

View Statistics:

 netstat -s|grep "send buffer errors

Solution:

• ICMP/UDP lacks flow control mechanisms, requiring applications to be designed with reasonable sending methods and speeds, taking into account the size of the underlying buffer, CPU load, and the quality of network bandwidth.
• Setting an Appropriate SOCK Buffer Size:

   setsockopt(s,SOL_SOCKET,SO_SNDBUF,  i(const char*)&nSendBuf,sizeof(int));

• Adjust the system socket buffer size:

   # Default Socket Send Buffer
   net.core.wmem_default = 31457280
   # Maximum Socket Send Buffer
   net.core.wmem_max = 33554432

Appendix: A Brief Summary of Packet Loss in the Kernel Protocol Stack:

Introduction to Relevant Tools

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Principle: Monitor the `kfree_skb` function (which is called when network packets are discarded) or the corresponding event, and then print the relevant call stack. If you want detailed insights on which function in the Linux system drops packets, you can use the `dropwatch` tool. This tool listens to system packet drop information and prints out the function where the packet drop occurs:

2. The tcpdump Utility

Principle: tcpdump is a powerful packet capture tool for Unix that allows users to intercept and display TCP/IP and other packets sent or received via network connections to that computer.

Packet capture command reference:

https://www.tcpdump.org/manpages/tcpdump.1.html

Packet Analysis:

1. Use Wireshark Tool for Analysis Reference: Practical Packet Analysis using Wireshark.pdf

2. The data can be converted into a CSV format, which can be analyzed using Excel or shell scripts for specific scenario message analysis;

3. You can use the `tshark` command line tool for analysis on Linux:

https://www.wireshark.org/docs/man-pages/tshark.html

Summary

This article merely analyzes the majority of potential packet loss nodes, providing individual node packet loss troubleshooting and related solutions. Packet loss issues involve various components of the network link, especially in the cloud network era, where network topology is complex and dynamic. It involves carrier networks, IDC (Internet Data Center) networks, dedicated lines as underlay networks, and boundary gateways, VPC (Virtual Private Cloud) networks, CLB (Cloud Load Balancing), among other cloud overlay networks. Troubleshooting packet loss issues is very complex and challenging, but by mastering the basic principles of network communication, one can decompose the network topology, conduct a thorough investigation of communication nodes, and identify the packet loss location. In subsequent discussions, we will delve deeper into packet loss troubleshooting methods in the era of cloud computing, network architecture analysis, etc., to enable quick troubleshooting and pinpointing of any packet loss issues, assisting customers in rapidly restoring business operations. Until next time.

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