{"id":538,"date":"2024-08-30T00:17:25","date_gmt":"2024-08-29T21:17:25","guid":{"rendered":"https:\/\/itgen.itbumper.com\/?page_id=538"},"modified":"2024-09-03T02:24:08","modified_gmt":"2024-09-02T23:24:08","slug":"iperf3","status":"publish","type":"page","link":"https:\/\/itgen.itbumper.com\/?page_id=538","title":{"rendered":"0009_Iperf3"},"content":{"rendered":"\n\n

iperf3 is a tool for network performance measurement. It is widely used to test the bandwidth between two endpoints in a network and can measure TCP, UDP, and SCTP throughput. iperf3 is useful for diagnosing network issues, verifying the quality of a network, and ensuring that a network can handle the required traffic.<\/p>\n

Install iperf3<\/strong>:
On Linux, you can install it using the package manager:<\/p>\n\n

\nsudo apt-get install iperf3\n<\/pre><\/div>\n\n
<\/p>\n
Basic TCP Test (Server and Client)<\/strong>:<\/p>\n
    \n
  • Server<\/strong>: Start iperf3 on one machine in server mode.<\/li>\n<\/ul>\n\n
    \niperf3 -s\n\n<\/pre><\/div>\n\n
      \n
    • Client<\/strong>: Run the test from another machine.<\/li>\n<\/ul>\n\n
      \niperf3 -c <server_ip>\n\n<\/pre><\/div>\n\n
        \n
      • \n
        This command tests the TCP bandwidth from the client to the server.<\/p>\n<\/li>\n<\/ul>\n
        UDP Test<\/strong>:<\/p>\n
          \n
        • Server<\/strong>:<\/li>\n<\/ul>\n\n
          \niperf3 -s\n\n<\/pre><\/div>\n\n
          Client<\/strong>:<\/p>\n\n
          \niperf3 -c <server_ip> -u\n\n<\/pre><\/div>\n\n
          The -u<\/code> flag switches to UDP mode.<\/p>\n
          <\/p>\n
          Specifying Bandwidth<\/strong>:<\/p>\n
            \n
          • To specify the bandwidth for a UDP test, use the -b<\/code> option:<\/li>\n<\/ul>\n\n
            \niperf3 -c <server_ip> -u -b 10M\n\n<\/pre><\/div>\n\n
              \n
            • \n
              This example sends data at 10 Mbps.<\/p>\n<\/li>\n<\/ul>\n
              Test Duration<\/strong>:<\/span><\/p>\n
                \n
              • Use the -t<\/code> option to set the test duration (in seconds):<\/li>\n<\/ul>\n\n
                \niperf3 -c <server_ip> -t 30\n\n<\/pre><\/div>\n\n
                  \n
                • \n
                  This command runs the test for 30 seconds.<\/p>\n<\/li>\n<\/ul>\n
                  Parallel Streams<\/strong>:<\/p>\n
                    \n
                  • \n
                      \n
                    • To test multiple parallel streams, use the -P<\/code> option:<\/li>\n<\/ul>\n<\/li>\n<\/ul>\n\n
                      \niperf3 -c <server_ip> -P 5\n\n<\/pre><\/div>\n\n
                      This sends data over 5 parallel streams.<\/p>\n
                      <\/p>\n
                      Test with Time-To-Live (TTL):<\/strong><\/p>\n\n
                      \niperf3 -c <server_ip> --ttl 64\n<\/pre><\/div>\n\n
                      <\/p>\n
                      Test with Quality of Service (ToS) or Differentiated Services Code Point (DSCP):<\/strong><\/p>\n\n
                      \niperf3 -c <server_ip> -S 0x20\n\n\n<\/pre><\/div>\n\n
                      ToS\/DSCP:<\/strong> These values are used to prioritize network traffic. For example, 0x20<\/code> sets a priority for bulk data transfer.<\/p>\n
                      Viewing Results:<\/strong><\/p>\n
                        \n
                      • The results will show the bandwidth achieved, loss rates (for UDP), and other metrics to assess the network’s performance.<\/li>\n<\/ul>\n\n
                        \nServer or Client:\n  -p, --port      #         server port to listen on\/connect to\n  -f, --format    [kmgKMG]  format to report: Kbits, Mbits, KBytes, MBytes\n  -i, --interval  #         seconds between periodic bandwidth reports\n  -F, --file name           xmit\/recv the specified file\n  -B, --bind      <host>    bind to a specific interface\n  -V, --verbose             more detailed output\n  -J, --json                output in JSON format\n  --logfile f               send output to a log file\n  --forceflush              force flushing output at every interval\n  -d, --debug               emit debugging output\n  -v, --version             show version information and quit\n  -h, --help                show this message and quit\nServer specific:\n  -s, --server              run in server mode\n  -D, --daemon              run the server as a daemon\n  -I, --pidfile file        write PID file\n  -1, --one-off             handle one client connection then exit\nClient specific:\n  -c, --client    <host>    run in client mode, connecting to <host>\n  -u, --udp                 use UDP rather than TCP\n  -b, --bandwidth #[KMG][\/#] target bandwidth in bits\/sec (0 for unlimited)\n                            (default 1 Mbit\/sec for UDP, unlimited for TCP)\n                            (optional slash and packet count for burst mode)\n  -t, --time      #         time in seconds to transmit for (default 10 secs)\n  -n, --bytes     #[KMG]    number of bytes to transmit (instead of -t)\n  -k, --blockcount #[KMG]   number of blocks (packets) to transmit (instead of -t or -n)\n  -l, --len       #[KMG]    length of buffer to read or write\n                            (default 128 KB for TCP, dynamic or 1 for UDP)\n  --cport         <port>    bind to a specific client port (TCP and UDP, default: ephemeral port)\n  -P, --parallel  #         number of parallel client streams to run\n  -R, --reverse             run in reverse mode (server sends, client receives)\n  -w, --window    #[KMG]    set window size \/ socket buffer size\n  -M, --set-mss   #         set TCP\/SCTP maximum segment size (MTU - 40 bytes)\n  -N, --no-delay            set TCP\/SCTP no delay, disabling Nagle's Algorithm\n  -4, --version4            only use IPv4\n  -6, --version6            only use IPv6\n  -S, --tos N               set the IP 'type of service'\n  -Z, --zerocopy            use a 'zero copy' method of sending data\n  -O, --omit N              omit the first n seconds\n  -T, --title str           prefix every output line with this string\n  --get-server-output       get results from server\n  --udp-counters-64bit      use 64-bit counters in UDP test packets\n\n<\/pre><\/div>\n\n\n
                        ToS\/DSCP Value<\/strong><\/td>Hexadecimal (DSCP)<\/strong><\/td>Binary<\/strong><\/td>Priority\/Type of Traffic<\/strong><\/td>Description<\/strong><\/td><\/tr>
                        0<\/td>0x00<\/td>000000<\/td>Default<\/td>Standart traffic. No priority.<\/td><\/tr>
                        8<\/td>0x20<\/td>001000<\/td>CS1 <\/td>Standard traffic. No priority.<\/td><\/tr>
                        16<\/td>0x40<\/td>010000<\/td>CS2<\/td>OAM (Operations, Administration, and Management).<\/td><\/tr>
                        24<\/td>0x60<\/td>011000<\/td>CS3<\/td>Signaling, call setup.<\/td><\/tr>
                        32<\/td>0x80<\/td>100000<\/td>CS4<\/td>Streaming media, real-time interactive.<\/td><\/tr>
                        40<\/td>0xA0<\/td>101000<\/td>CS5<\/td>Critical applications, video conferencing.<\/td><\/tr>
                        48<\/td>0xC0<\/td>110000<\/td>CS6<\/td>Network control, routing protocols.<\/td><\/tr>
                        56<\/td>0xE0<\/td>111000<\/td>CS7<\/td>Network control, high-priority traffic.<\/td><\/tr>
                        10<\/td>0x28<\/td>001010<\/td>AF11 <\/td>Low drop precedence, medium priority traffic.<\/td><\/tr>
                        18<\/td>0x48<\/td>010010<\/td>AF21 <\/td>Low drop precedence, high priority traffic.<\/td><\/tr>
                        26<\/td>0x68<\/td>011010<\/td>AF31 <\/td>Low drop precedence, real-time interactive.<\/td><\/tr>
                        34<\/td>0x88<\/td>100010<\/td>AF41<\/td>Low drop precedence, high-priority video and voice.<\/td><\/tr>
                        46<\/td>0xB8<\/td>101110<\/td>EF<\/td>High-priority, low-latency traffic, often used for voice over IP (VoIP).<\/td><\/tr>
                        48<\/td>0xC0<\/td>110000<\/td>CS6<\/td>Network control, routing protocols.<\/td><\/tr>
                        56<\/td>0xE0<\/td>111000<\/td>CS7<\/td>Network control, high-priority traffic.<\/td><\/tr><\/tbody>
                        <\/td><\/td><\/td><\/td><\/td><\/tr><\/tfoot><\/table><\/figure>\n\n\n
                        Key Terms:<\/h3>\n
                          \n
                        • CS (Class Selector):<\/strong> Used for backward compatibility with IP precedence.<\/li>\n
                        • AF (Assured Forwarding):<\/strong> Assures delivery under prescribed conditions.<\/li>\n
                        • EF (Expedited Forwarding):<\/strong> Ensures low loss, latency, and jitter traffic.<\/li>\n<\/ul>\n
                          These DSCP values help prioritize network traffic, ensuring that more critical data (e.g., VoIP or video conferencing) gets the necessary bandwidth and lower latency.<\/p>\n
                          <\/p>\n
                          Example Analysis for VoIP Using Iperf3<\/strong><\/p>\n
                          Objective:<\/strong> To determine if the existing network can support VoIP traffic with minimal latency, packet loss, and jitter.<\/p>\n
                          Process:<\/strong><\/p>\n
                            \n
                          1. \n
                            Run Iperf3 Server:<\/strong><\/p>\n
                              \n
                            • Start the server on one machine.<\/li>\n<\/ul>\n<\/li>\n<\/ol>\n\n
                              \niperf3 -s\n<\/pre><\/div>\n\n
                              2. Run Iperf3 Client with VoIP Parameters:<\/strong><\/p>\n
                                \n
                              • On another machine, run the client specifying the server\u2019s IP and parameters to simulate VoIP traffic:<\/li>\n<\/ul>\n\n
                                \niperf3 -c <server_ip> -u -b 100K -t 30 -S 0xB8\n<\/pre><\/div>\n\n
                                3. Parameters:<\/strong><\/p>\n
                                  \n
                                1. \n
                                    \n
                                  • \n
                                      \n
                                    • -u<\/code>: Uses UDP (since VoIP uses UDP).<\/li>\n
                                    • -b 100K<\/code>: Limits bandwidth to 100 Kbps, typical for VoIP traffic.<\/li>\n
                                    • -t 30<\/code>: Test duration is 30 seconds.<\/li>\n
                                    • -S 0xB8<\/code>: Uses DSCP value 0xB8<\/code> (EF – Expedited Forwarding), meaning high priority and low latency.<\/li>\n<\/ul>\n<\/li>\n<\/ul>\n<\/li>\n<\/ol>\n
                                      4. Analyze Results:<\/strong><\/p>\n
                                        \n
                                      • \n
                                          \n
                                        • Latency:<\/strong> Latency is critical. For VoIP, latency should be under 150 ms.<\/li>\n
                                        • Packet Loss:<\/strong> Packet loss should be minimal (0-1%) for good VoIP quality.<\/li>\n
                                        • Jitter:<\/strong> Jitter should be minimal, preferably under 30 ms for VoIP.<\/li>\n<\/ul>\n<\/li>\n<\/ul>\n
                                          Possible Outcomes:<\/strong><\/p>\n
                                            \n
                                          1. \n
                                            Optimal Conditions:<\/strong><\/p>\n
                                              \n
                                            • Results:<\/strong> Latency < 100 ms, Packet Loss < 1%, Jitter < 20 ms.<\/li>\n
                                            • Conclusion:<\/strong> The network is well-suited for VoIP with high-quality communication.<\/li>\n<\/ul>\n<\/li>\n
                                            • \n
                                              Acceptable Conditions:<\/strong><\/p>\n
                                                \n
                                              • Results:<\/strong> Latency 100-150 ms, Packet Loss 1-2%, Jitter 20-30 ms.<\/li>\n
                                              • Conclusion:<\/strong> The network is acceptable for VoIP, but occasional quality issues may occur.<\/li>\n<\/ul>\n<\/li>\n
                                              • \n
                                                Unacceptable Conditions:<\/strong><\/p>\n