Skip to main content

Dissecting traces with DTMF tones

I'm sure I belong to the large group of people who love to analyse network traces with tools like Wireshark. Being able to see the details of a packet or datagram down to the level of the bits is not only extremely useful, but also fascinating.


Time ago I wrote a dissector for Wireshark, using the Lua interface, and that was fun (I see it's still available here). The official recommendation is to use Lua only for prototyping and testing, but when performances are not key and there isn't the intent to add the dissector to the official distribution, it's fast and effective.

In order to parse network traces with audio and extract it into the payload first, and then decode it into a WAV file, C is a viable solution. I wrote about a program that does that here and since it attracted some attention and feedback I wrote an updated version later.

More recently I wanted to identify programmatically the presence (and value) of DTMF tones - as RTP Events, RFC 2833 - in network traces. This time rather than using C, I wanted to integrate it with python, and scapy seemed a good choice.

scapy is quite complete, but interestingly it doesn't have a parser for the RTP Event extension. So I thought of mapping the raw content in the RTP payload to a structure, with the help of C types.

This it the core of the program:

def process_pcap(file_name, sut_ip):

  for (pkt_data, pkt_metadata,) in RawPcapReader(file_name):

    ether_pkt = Ether(pkt_data)

    # A little housekeeping to filter IPv4 UDP packets goes here
    ...

    if ether_pkt.haslayer(UDP):
      udp_pkt = ether_pkt[UDP]

    # Get the raw UDP packet into an RTP structure
    rtp_pkt = RTP(udp_pkt["Raw"].load)

    ptype = rtp_pkt.payload_type

    # Assume payload type 96 or 101 are used for RTP events
    if (ptype == 96 or ptype == 101):
      rtpevent_content = rtp_pkt.payload

    # map the payload into an RTPEvent object
    rtpevent_struct = RTPEvent.from_buffer_copy(rtpevent_content.load)

The RTPEvent class looks like this:

class RTPEvent(ctypes.BigEndianStructure):
  _fields_ = [
    ('event_id', ctypes.c_uint8),
    ('end_of_event', ctypes.c_uint8, 1),
    ('reserved', ctypes.c_uint8, 1),
    ('volume', ctypes.c_uint8, 6),
    ('duration', ctypes.c_uint16)
  ]

so once mapped, the rtpevent_struct object will have its DTMF-specific details, in particular with the digit contained in rtpevent_struct.event_id, and the indication whether it's the marker of end of the event in the end_of_event bit.

All the other information (source/destination IP address and port, timestamp, SSRC) is obviously available in the UDP and RTP portion, so it's easy to adapt to your needs and filter out the DTMF tones for the streams you're interesting in.


Popular posts from this blog

Troubleshooting TURN

  WebRTC applications use the ICE negotiation to discovery the best way to communicate with a remote party. I t dynamically finds a pair of candidates (IP address, port and transport, also known as “transport address”) suitable for exchanging media and data. The most important aspect of this is “dynamically”: a local and a remote transport address are found based on the network conditions at the time of establishing a session. For example, a WebRTC client that normally uses a server reflexive transport address to communicate with an SFU. when running inside the home office, may use a relay transport address over TCP when running inside an office network which limits remote UDP targets. The same configuration (defined as “iceServers” when creating an RTCPeerConnection will work in both cases, producing different outcomes.

VoIP calls encoded with SILK: from RTP to WAV

SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates and frame sizes), from narrowband (8 KHz) to wideband (24 KHz). Since Wireshark doesn't allow you to decode an RTP stream carrying SILK frames, I was curious to find a programmatic way to do it. In fact, this has also allowed to me to earn a "tumbleweed" badge in stackoverflow . You may argue that a Wireshark plugin would be the right solution, but that's probably for another day. Initially I thought it was sufficient to read the specification for RTP payload when using SILK ; the truth is that I had to reverse engineer a solution by looking at SILK SDK's test vectors. There, I discovered that a file containing SILK audio doesn't have the file header indicated in the IETF draft ("!#SILK"), but a slightly different one ("!#SILK_V3"). More importantly, each encoded frame is not preced...

Extracting Opus from a pcap file into an audible wav

From time to time I need to verify that the audio inside a trace is as expected. Not much in terms of quality, but more often content and duration. A few years ago I wrote a small program to transform a pcap into a wav file - the codec in use was SILK. These days I'm dealing with Opus , and I have to say things are greatly simplified, in particular if you consider opus-tools , a set of utilities to handle opus files and traces. One of those tools, opusrtp , can do live captures and write the interpreted payload into a .opus file. Still, what I needed was to achieve the same result but from a pcap already existing, i.e. "offline". So I come up with a small - quite shamlessly copy&pasted - patch to opusrtc, which is now in this fork . Once you have a pcap with an RTP stream with opus (say in input.pcap ) you can retrieve the .opus equivalent (in rtpdump.opus ) with: ./opusrtp --extract input.pcap Then you can generate an audible wav file with: ./opusd...