Softphone.answer() returns an inbound call session. Softphone.call() returns
an outbound call session. Both share the controls and events below; outbound-only
members are marked explicitly.
Methods
Method
Result
Description
hangup()
Promise<void>
Hang up an active call.
sendDTMF(char)
void
Send one of 0-9, *, #, or A-D.
sendDTMFs(chars, delay?)
Promise<void>
Send validated characters with a 500 ms default delay after each one.
Immediately start sending raw audio in the selected codec's input format.
sendPacket(rtpPacket)
void
Forward a received RTP packet to this session.
transfer(destination)
Promise<void>
Transfer an active call and resolve after a successful SIP notification.
hold()
Promise<void>
Stop receiving remote audio through a SIP re-invite.
unhold()
Promise<void>
Resume receiving remote audio.
cancel()
Promise<void>
Cancel before the peer answers. Outbound only.
Properties
Property
Type
Description
callId
string
SIP Call-ID.
disposed
boolean
Whether the session has been disposed.
sessionId
string
RingCentral telephony session ID. Outbound only.
partyId
string
RingCentral telephony party ID. Outbound only.
Events
Event
Payload
Description
answered
none
The peer answered. Outbound only.
busy
none
The destination returned SIP 486; the session is then disposed. Outbound only.
disposed
none
The session closed and its media resources were released.
audioPacket
RtpPacket
Received audio. The payload has been decoded for the selected codec.
dtmf
string
A decoded DTMF character.
dtmfPacket
RtpPacket
A telephone-event RTP packet.
rtpPacket
RtpPacket
Any decrypted incoming RTP packet, emitted before audio decoding.
rtpPacket and sendPacket() are the supported observation and forwarding
hooks. Raw socket writes, SRTP state, codec workers, peer fields, sequence
counters, and helper methods are implementation details.