rtpmixsound 介绍


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rtpmixsound 源代码版本库


rtpmixsound - 实时混合预先录制的音频

:~# rtpmixsound -h

rtpmixsound - Version 3.0
              January 03, 2007
 Mandatory -
    pathname of file whose audio is to be mixed into the
        targeted live audio stream. If the file extension is
        .wav, then the file must be a standard Microsoft
        RIFF formatted WAVE file meeting these constraints:
          1) header 'chunks' must be in one of two sequences:
               RIFF, fmt, fact, data
               RIFF, fmt, data
          2) Compression Code = 1 (PCM/Uncompressed)
          3) Number of Channels = 1 (mono)
          4) Sample Rate (Hz) = 8000
          5) Significant Bits/Sample =
                  signed,   linear 16-bit
                  unsigned, linear  8-bit
        If the file name does not specify a .wav extension,
        then the file is presumed to be a tcpdump formatted
        file with a sequence of, exclusively, G.711 u-law
        RTP/UDP/IP/ETHERNET messages
        Note: Yep, the format is referred to as 'tcpdump'
              even though this file must contain udp messages
 Optional -
    -a source RTP IPv4 addr
    -A source RTP port
    -b destination RTP IPv4 addr
    -B destination RTP port
    -f spoof factor - amount by which to:
         a) increment the RTP hdr sequence number obtained
            from the ith legitimate packet to produce the
            RTP hdr sequence number for the ith spoofed packet
         b) multiply the RTP payload length and add that
            product to the RTP hdr timestamp obtained from
            the ith legitimate packet to produce the RTP hdr
            timestamp for the ith spoofed packet
         c) increment the IP hdr ID number obtained from the
            ith legitimate packet to produce the IP hdr ID
            number for the ith spoofed packet
       [ range: +/- 1000, default: 2 ]
    -i interface (e.g. eth0)
    -j jitter factor - the reception of a legitimate RTP
         packet in the target audio stream enables the output
         of the next spoofed packet. This factor determines
         when that spoofed packet is actually transmitted.
         The factor relates how close to the next legitimate
         packet you'd actually like the enabled spoofed packet
         to be transmitted. For example, -j 10 means 10% of
         the codec's transmission interval. If the transmission
         interval = 20,000 usec (i.e. G.711), then delay the
         output of the spoofed RTP packet until the time-of-day
         is within 2,000 usec (i.e. 10%) of the time the next
         legitimate RTP packet is expected. In other words,
         delay 100% minus the jitter factor, or 18,000 usec
         in this example. The smaller the jitter factor, the
         greater the risk you run of not outputting the
         spoofed packet before the next legitimate RTP packet
         is received. Therefore, a factor >= 10 is advised.
       [ range: 0 - 80, default: 80 = output spoof ASAP ]
    -p seconds to pause between setup and injection
    -h help - print this usage
    -v verbose output mode

Note: If you are running the tool from a host with multiple
      ethernet interfaces which are up, be forewarned that
      the order those interfaces appear in your route table
      and the networks accessible from those interfaces might
      compel Linux to output spoofed audio packets to an
      interface different than the one stipulated by you on
      command line. This should not affect the tool unless
      those spoofed packets arrive back at the host through
      the interface you have specified on the command line
      (e.g. the interfaces have connectivity through a hub).

rtpmixsound 用法示例


:~# rtpmixsound /usr/share/rtpmixsound/stapler.wav -v
Targeting interface eth0
libfindrtp_find_rtp(): using pcap filter "ip".
State: ip_a ==  | port_a == 0 | ip_b ==  | port_b == 0