Supported audio codecs:
- G.711 (ulaw and alaw)
- AMR (NB)
- silk (sampling rates 8, 12, 16, 24 kHz): used at Skype
- slin (sampling rates 8, 12, 16, 24, 32, 44, 48, 96, 192 kHz)
- speex (sampling rates 8, 32 kHz)
- opus: used at WebRTC
All codecs support translation on fly.
For G.729 encoding is performed only to G.729a format.
Decoding is supported both for G.729a and G.729b.
In case if both legs have been created with G.729, frames are passed transparently, without translation G.729a <-> G.729b.
- p2p (packet to packet).
Available for SIP channel;
Available for all codecs.
Both the application-level proxy and the OS kernel-level proxy mode are available (Media Proxy);
- Codec translation.
It turns on automatically if during the session negotiation a scheme was selected according to which transcoding is required.
- buffering (can be enabled optionally).
You can define the duration of the voice frame in milliseconds and the system will buffer the stream according to the settings.
- reading DTMF from voice stream (processing tones at the software DSP) - for G.711 ulaw/alaw codecs
- writing to file from RTP stream.
Playing from a file to an RTP stream.
Supported file formats:
- wav. This activates the codec translation mode.
- mp3. This activates the codec translation mode.
- g729, g723, etc. (formats for all codecs are supported)
If the file format does not match the currently selected format for the channel, the transcoding mode is activated.
- generate audio to RTP stream. Generation of ringback tones.