VocalDocscontracts 1.65.0

SipMediaEncryption

Whether a LiveKit SIP trunk uses SRTP for the media of a call (1.34.0). none: plain RTP, with no offer of SRTP. allow: SRTP when the other side offers it and plain RTP otherwise, a step on the way to require and not a place to stay, because a call then connects without encryption and without a sign. require: LiveKit SIP refuses a call without SRTP with a SIP 400 before any room exists.

An enumeration of 3 values.

Values

  • none
  • allow
  • require