C# 클래스 SIPSorcery.SoftPhone.AudioChannel

파일 보기 프로젝트 열기: sipsorcery/sipsorcery 1 사용 예제들

공개 프로퍼티들

프로퍼티 타입 설명
SupportedAudioTypes List

공개 메소드들

메소드 설명
AudioChannel ( ) : System
AudioSampleReceived ( byte sample, int offset ) : void

Event handler for receiving an RTP packet containing and audio payload from the remote end of the VoIP call.

Close ( ) : void

Called when the audo channel is no longer required, such as when the VoIP call using it has terminated, and all resources can be shutdown and closed.

GetMediaAnnouncement ( ) : SDPMediaAnnouncement

Gets the media announcement to include in the SDP payload for a call.

GetSDP ( bool usePublicIP ) : SDP

Gets an SDP packet that can be used by VoIP clients to negotiate an audio connection. The SDP will only offer PCMU since that's all I've gotten around to handling.

SetRemoteRTPEndPoint ( IPEndPoint remoteEndPoint ) : void

Allows an arbitrary block of bytes to be sent on the RTP channel. This is mainly used for the Gingle client which needs to send a STUN binding request to the Google Voice gateway.

Sets the remote end point for the RTP channel. This will be set from the SDP packet received from the remote end of the VoIP call.

StartRecording ( ) : void

비공개 메소드들

메소드 설명
AudioSampleAvailable ( object sender, NAudio.Wave.WaveInEventArgs e ) : void

Event handler for receiving an audio sample that is ready for encoding, packaging into RTP and sending to the remote end of the VoIP call.

RTPChannelSampleAvailable ( object sender, NAudio.Wave.WaveInEventArgs e ) : void

Event handler for receiving an audio sample that is ready for encoding, packaging into RTP and sending to the remote end of the VoIP call.

RTPChannelSampleReceived ( RTPFrame rtpFrame ) : void

Event handler for receiving an RTP frmae from the remote end of the VoIP call.

메소드 상세

AudioChannel() 공개 메소드

public AudioChannel ( ) : System
리턴 System

AudioSampleReceived() 공개 메소드

Event handler for receiving an RTP packet containing and audio payload from the remote end of the VoIP call.
public AudioSampleReceived ( byte sample, int offset ) : void
sample byte The audio sample.
offset int The offset in the sample that the audio starts.
리턴 void

Close() 공개 메소드

Called when the audo channel is no longer required, such as when the VoIP call using it has terminated, and all resources can be shutdown and closed.
public Close ( ) : void
리턴 void

GetMediaAnnouncement() 공개 메소드

Gets the media announcement to include in the SDP payload for a call.
public GetMediaAnnouncement ( ) : SDPMediaAnnouncement
리턴 SIPSorcery.Net.SDPMediaAnnouncement

GetSDP() 공개 메소드

Gets an SDP packet that can be used by VoIP clients to negotiate an audio connection. The SDP will only offer PCMU since that's all I've gotten around to handling.
public GetSDP ( bool usePublicIP ) : SDP
usePublicIP bool If true and the public IP address is available from the STUN client then /// the public IP address will be used in the SDP otherwise the hsot machine's default IPv4 address will /// be used.
리턴 SIPSorcery.Net.SDP

SetRemoteRTPEndPoint() 공개 메소드

Allows an arbitrary block of bytes to be sent on the RTP channel. This is mainly used for the Gingle client which needs to send a STUN binding request to the Google Voice gateway. Sets the remote end point for the RTP channel. This will be set from the SDP packet received from the remote end of the VoIP call.
public SetRemoteRTPEndPoint ( IPEndPoint remoteEndPoint ) : void
remoteEndPoint System.Net.IPEndPoint The remote end point to send RTP to.
리턴 void

StartRecording() 공개 메소드

public StartRecording ( ) : void
리턴 void

프로퍼티 상세

SupportedAudioTypes 공개적으로 프로퍼티

public List SupportedAudioTypes
리턴 List