Three and a half years ago (which really sounds like a lot of time!) I was working with a VoIP infrastructure using SILK. As it often happens to server-side developers/integrators, you have to prove whether the audio provided by a client or to a client is correctly encoded :-)
Wireshark is able to decode, and play, G.711 streams, but not SILK (or Opus - more on this later). So I thought of having my own tool handy, to generate a WAV file for a PCAP with RTP carrying SILK frames.
The first part requires extracting the SILK payload and writing it down into a bistream file. Then you have to decode the audio using the SILK SDK decoder, to get a raw audio file. From there to a WAV file it is very easy.
As I tried to describe in this previous post, I had to reverse engineer the test files contained in the SDK, to see what a SILK file looked like.
Since the SILK payload is not constant, all that was needed was to insert 2 Bytes with the length of the following SILK frame. At the beginning of the file you have to add a header containing "#!SILK_V3", and voilà.
This is accomplished by silk_rtp_to_bistream.c (from https://github.com/giavac/silk_rtp_to_bitstream), a small program based on libpcap that extracts the SILK payload from a PCAP and writes it properly into a bistream file.
Build the binary with:
gcc silk_rtp_to_bitstream.c -lpcap -o silk_rtp_to_bitstream
(you'll need libpcap-dev installed)
Create the bistream with:
./silk_rtp_to_bitstream input.pcap silk.bit
Now you can decode, using the SILK SDK, from bitstream into raw audio with:
$SILK_SDK/decoder silk.bit silk.raw
Raw audio to WAV can be done with sox:
sox -V -t raw -b 16 -e signed-integer -r 24000 silk.raw silk.wav
This works fine with single channel SILK at 8000 Hz.
More to come: an update on how to accomplish the same but for Opus.
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 detai...
Verto is a newly designed signalling protocol for WebRTC clients interacting with FreeSWITCH . It has an intuitive, JSON-based RPC whic...
SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates ...
On a previous post I shared my experiments with node.js as a WebSocket server. This is quite useful for people working on WebRTC prototyp...