LanScape VOIP Media Engine
SIP_ACTIVE_CALL_INFO
The SIP_ACTIVE_CALL_INFO structure defines the information an application can request when an out going or in coming call connect notification is sent to the application. The application usually retrieves this information during the following events: SipIncomingCallConnected, SipOutgoingCallConnected, SipCallHoldOn, SipCallHoldOff, SipFarEndHoldOn, and SipFarEndHoldOff.
typedef struct
{
char *pSipUrlFarEnd;
char *pCallId;
// raw SIP messages used to set up the call.
char *pInviteSipMesage;
char *pResponse2XXSipMesage;
// audio.
int LocalAudioRtpPort;
int FarEndAudioRtpPort;
MEDIA_FORMAT_AUDIO
AudioFormat;
int AudioFormatRtp;
// video.
int LocalVideoRtpPort;
int FarEndVideoRtpPort;
MEDIA_FORMAT_VIDEO VideoFormat;
int VideoFormatRtp;
// call hold.
BOOL LocalHoldActive;
BOOL FarEndHoldActive;
}SIP_ACTIVE_CALL_INFO;
Members:
pSipUrlFarEnd
This character array contains the SIP URL of the far end of the call. Please refer to the installed SIP RFCs for a description of the format of SIP URLs.
pCallId
The call ID for the call.
pInviteSipMesage
The raw INVITE SIP message for the call.
p2XXSipMesage
The raw 2xx response SIP message for the
call.
LocalAudioRtpPort
This is the local UDP audio port that is being used by the telephony engine’s RTP transceiver to send and receive media stream data.
FarEndAudioRtpPort
This is the UDP audio port that is being used by the far end to receive RTP data.
AudioFormat
The audio format member specifies the outcome of the SIP format/rate negotiation with the far end. This is the media format used by the active phone call.
AudioFormatRtp
The raw RTP audio format
as specified in transmitted and received RTP packets.
LocalVideoRtpPort
Not yet used.
FarEndVideoRtpPort
Not yet used.
VideoFormat
Not yet used.
VideoFormatRtp
Not yet used.
LocalHoldActive
If non zero, this member indicates that we have put the far end on hold.
FarEndHoldActive
If non zero, this member indicates that the far end has put us on hold.