Codec conversion

Our RTP stack has a built-in capability to handle various codecs and to do automatic transcoding when necessary. While you usually should avoid any codec conversion, there are some scenarios when such kind of behavior is very useful. For example you can have conferences where clients are using different codecs, or if one of your carriers supports only one predefined codec which is missing from the client device.
We support transcoding from/to the most codecs used in telecommunication:

  • OPUS
  • Speex
  • G.729
  • G.723
  • G.722
  • G.726
  • iLBC
  • GSM
  • PCMU (G.711 A-Law)
  • PCMA (G.711 μ-Law)
Supported sampling rates:
  • Narrowband -8kHz
  • Wideband -16 kHz
  • Super Wideband -24 kHz
  • Fullband -48 kHz

VoIP Conferencing

The mizu VoIP server has built-in conference mixing capabilities:

  • Conference calls
  • Unlimited users (limited by hardware resources only)
  • Three-Way Calling (SIP standard)
  • DTMF triggered conference calls
  • Conference rooms