FreeCalypso > hg > gsm-codec-lib
annotate doc/RTP-analysis @ 196:1cdbaeec7bcc
libgsmefr/levinson.c: perf opt
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Wed, 04 Jan 2023 03:58:26 +0000 |
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1 The present package includes a number of utilities for analyzing RTP streams |
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2 that have been captured with tcpdump or equivalent tools in pcap format. In |
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3 order to use any of these utilities, you need to have a pcap file (obviously), |
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4 and you need to identify the RTP stream to be analyzed or extracted by either |
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5 source or destination IP:port. All tools begin by applying a filter, |
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6 considering only those packets that are UDP in IPv4 (no IPv6 support currently), |
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7 and only those that match the specified source or destination IP:port. Every |
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8 matched packet is checked for a valid RTP header, and then the actual RTP stream |
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9 analysis or extraction takes place, depending on the specific tool: |
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11 rtp-cont-check This program checks the selected RTP stream for continuity. It |
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12 verifies that every matched packet has the same SSRC, that the |
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13 sequence number always increments by 1 from each individual |
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14 packet to the next, and that the RTP header timestamp always |
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15 increments by 160 units. (The assumption is that the |
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16 application at hand is in the traditional telephony domain, |
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17 with a sampling rate of 8000 samples/s and 20 ms packetization |
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18 for RTP.) This tool also looks at the capture time deltas |
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19 between successive packets and reports the observed minimum and |
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20 maximum; by seeing min and max delta-T, a developer can easily |
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21 notice timing aberrations that aren't caught by RTP header |
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22 sequence number and timestamp checks. |
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23 |
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24 rtp-g711-extr This program focuses on a single selected RTP stream like the |
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25 others, enforces its continuity just like rtp-cont-check, and |
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26 then further enforces that every RTP packet be a 160-byte |
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27 payload, presumed to be either PCMU or PCMA. (The payload type |
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28 number is NOT considered, only the payload length.) The |
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29 selected G.711 RTP stream is then extracted and written into a |
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30 raw binary file. |
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31 |
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32 rtp-gsmfr-extr This program focuses on a single selected RTP stream like the |
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33 others, enforces its continuity just like rtp-cont-check, but |
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34 then further enforces that every RTP packet be one of these 3 |
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35 possibilities: a GSM FR1 codec frame, a GSM EFR codec frame or |
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36 a Themyscira BFI marker as defined in the RTP-BFI-extension |
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37 document. (The payload type number is NOT considered, only the |
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38 payload length and the characteristic signature of each of the |
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39 3 allowed possibilities.) The selected RTP stream is then |
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40 extracted and written into a gsmx file (see Binary-file-format), |
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41 which can then be analyzed with gsmrec-dump or decoded to |
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42 playable WAV with gsmfr-decode or gsmefr-decode. |
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43 |
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44 rtp-jitter-view This program analyzes a single selected RTP stream just like |
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45 rtp-cont-check, but instead of reporting minimum and maximum |
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46 time deltas for the entire stream, it prints the individual |
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47 capture time delta between every successive pair of packets. |