sipXtapi.h File Reference

sipXtapi main API declarations More...

#include <memory.h>
#include <string.h>
#include <stddef.h>

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Classes
class	SIPX_SECURITY_ATTRIBUTES
	Container class for security attributes. More...
struct	SIPX_VIDEO_DISPLAY
	Structure used to pass window handle/filter interface for video calls. More...
struct	SIPX_CONTACT_ADDRESS
	The CONTACT_ADDRESS structure includes contact information (IP-address and port), address source type, and interface. More...
struct	SIPX_AUDIO_CODEC
	The SIPX_AUDIO_CODEC structure includes codec name and bandwidth info. More...
struct	SIPX_RTCP_STATS
	RTCP statistics computed according to RFC 3550. More...
struct	SIPX_VIDEO_CODEC
	The SIPX_VIDEO_CODEC structure includes codec name and bandwidth info. More...
struct	SIPX_CODEC_INFO
	In the MEDIA_LOCAL_START and MEDIA_REMOTE_START events the SIPX_CODEC_INFO structure is being passed up to the event handler and contains information about the negotiated audio and video codec. More...
struct	SIPX_CALL_OPTIONS
	This structure gets passed into sipxCallConnect, sipxCallAccept, and sipxConferenceAdd calls and sets options on a per call basis. More...

Macros
#define	SIPX_CALLING_CONVENTION
	The SIPX_CALLING_CONVENTION define controls the default calling convention. More...
#define	DEFAULT_UDP_PORT   5060
	Default UDP port. More...
#define	DEFAULT_TCP_PORT   5060
	Default TCP port. More...
#define	DEFAULT_TLS_PORT   5061
	Default TLS port. More...
#define	DEFAULT_RTP_START_PORT   9000
	Starting RTP port for RTP port range. More...
#define	DEFAULT_STUN_PORT   3478
	Default stun server port. More...
#define	DEFAULT_CONNECTIONS   32
	Default number of max sim. More...
#define	DEFAULT_IDENTITY   "sipx"
	sipx<IP>:UDP_PORT used as identify if lines are not defined. More...
#define	DEFAULT_BIND_ADDRESS   "0.0.0.0"
	Bind to the first physical interface discovered. More...
#define	GAIN_MIN   0
	Min acceptable gain value. More...
#define	GAIN_MAX   100
	Max acceptable gain value. More...
#define	GAIN_DEFAULT   70
	Nominal gain value. More...
#define	VOLUME_MIN   0
	Min acceptable volume value. More...
#define	VOLUME_MAX   100
	Max acceptable volume value. More...
#define	VOLUME_DEFAULT   70
	Nominal volume value. More...
#define	MAX_AUDIO_DEVICES   64
	Max number of input/output audio devices. More...
#define	MAX_VIDEO_DEVICES   8
	Max number of video capture devices. More...
#define	MAX_VIDEO_DEVICE_LENGTH   256
	Max length of video capture device string. More...
#define	CONF_MAX_CONNECTIONS   32
	Max number of conference participants. More...
#define	SIPX_MAX_IP_ADDRESSES   32
	Maximum number of IP addresses on the host. More...
#define	SIPX_MAX_CALLS   64
	Maximum number of simultaneous calls. More...
#define	SIPX_PORT_DISABLE   -1
	Special value that disables the transport type (e.g. More...
#define	SIPX_PORT_AUTO   -2
	Special value that instructs sipXtapi to automatically select an open port for signaling or audio when passed to sipXinitialize. More...
#define	SIPXTAPI_VERSION_STRING   "sipXtapi SDK %s.%s %s (built %s)"
	Version string format string. More...
#define	SIPXTAPI_VERSION   "3.3.0"
	sipXtapi API version – automatically filled in during release process More...
#define	SIPXTAPI_BUILDNUMBER   "0"
	Default build number – automatically filled in during release process. More...
#define	SIPXTAPI_BUILD_WORD   3,3,0,0
	Default build word – automatically filled in during release process. More...
#define	SIPXTAPI_FULL_VERSION   "3.3.0.0"
	Default full version number – automatically filled in during release process. More...
#define	SIPXTAPI_BUILDDATE   "0000-00-00"
	Default build date – automatically filled in during release process. More...
#define	SIPX_MAX_ADAPTER_NAME_LENGTH   256
	Max length of an adapter name. More...
#define	SIPXTAPI_API   extern "C"
	Assume extern "C" for non-win32 platforms. More...
#define	MAX_SRTP_KEY_LENGTH   31
	srtp key length More...
#define	MAX_SMIME_KEY_LENGTH   2048
	s/mime key length More...
#define	MAX_PKCS12_KEY_LENGTH   4096
	pkcs12 key length More...
#define	MAX_PASSWORD_LENGTH   32
	maximum password length PKI operations More...
#define	SIPXTAPI_CODEC_NAMELEN   32
	Maximum length for codec name. More...
#define	SIPXTAPI_CODEC_NAMELEN   32
	Maximum length for codec name. More...

Typedefs
typedef enum MEDIA_TYPE	MEDIA_TYPE
	Media type. More...
typedef enum SPEAKER_TYPE	SPEAKER_TYPE
	Speaker output types are used to differentiate between the logical ringer (used to alert user of in inbound call) and speaker (in call audio device). More...
typedef enum SIPX_AUDIO_BANDWIDTH_ID	SIPX_AUDIO_BANDWIDTH_ID
	Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements. More...
typedef enum SIPX_VIDEO_BANDWIDTH_ID	SIPX_VIDEO_BANDWIDTH_ID
	Video Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements. More...
typedef enum SIPX_VIDEO_QUALITY_ID	SIPX_VIDEO_QUALITY_ID
	Video Codec quality definitions. More...
typedef enum SIPX_VIDEO_FORMAT	SIPX_VIDEO_FORMAT
	Enumeration of possible video sizes. More...
typedef enum SIPX_AUDIO_DATA_FORMAT	SIPX_AUDIO_DATA_FORMAT
	Format definitions for memory resident audio data. More...
typedef void(*	sipxLogCallback) (const char *szPriority, const char *szSource, const char *szMsg)
	Signature for a log callback function that gets passed three strings, first string is the priority level, second string is the source id of the subsystem that generated the message, and the third string is the message itself. More...
typedef enum SIPX_RESULT	SIPX_RESULT
	SIPX_RESULT is an enumeration with all the possible result/return codes. More...
typedef enum SIPX_TONE_ID	SIPX_TONE_ID
	DTMF/other tone ids used with sipxCallStartTone/sipxCallStopTone. More...
typedef SIPX_TONE_ID	TONE_ID
	Left for backwards compatibility – please use SIPX_TONE_ID. More...
typedef enum SIPX_AUDIO_FILE_FORMAT	SIPX_AUDIO_FILE_FORMAT
	Audio file format types. More...
typedef enum SIPX_LOG_LEVEL	SIPX_LOG_LEVEL
	Various log levels available for the sipxConfigEnableLog method. More...
typedef void *	SIPX_WINDOW_HANDLE
	Type for storing a "window object handle" - in Windows, the application should cast their HWND to a SIPX_WINDOW_HANDLE. More...
typedef enum SIPX_VIDEO_DISPLAY_TYPE	SIPX_VIDEO_DISPLAY_TYPE
	Enum for specifying the type of display object to be used for displaying video. More...
typedef int	SIPX_CONTACT_ID
	Type for storing Contact Record identifiers. More...
typedef void *	SIPX_INST
	The SIPX_INST handle represents an instance of a user agent. More...
typedef unsigned int	SIPX_LINE
	The SIPX_LINE handle represents an inbound or outbound identity. More...
typedef unsigned int	SIPX_CALL
	The SIPX_CALL handle represents a call or connection between the user agent and another party. More...
typedef unsigned long	SIPX_CONF
	The SIPX_CONF handle represents a collection of CALLs that have bridge (mixed) audio. More...
typedef unsigned long	SIPX_INFO
	The SIPX_INFO handle represents a handle to an INFO message sent by a sipXtapi instance. More...
typedef unsigned long	SIPX_PUB
	The SIPX_PUB handle represent a publisher context. More...
typedef unsigned long	SIPX_SUB
	A SIPX_SUB handle represent a subscription to a remote publisher. More...
typedef unsigned long	SIPX_TRANSPORT
	The SIPX_TRANSPORT handle represents a user-implementation of a network transport mechanism to be used for SIP signaling. More...
typedef bool(SIPX_CALLING_CONVENTION *	SIPX_TRANSPORT_WRITE_PROC) (SIPX_TRANSPORT hTransport, const char *szDestinationIp, const int iDestPort, const char *szLocalIp, const int iLocalPort, const void *pData, const size_t nData, const void *pUserData)
	External Transport callback definition. More...
typedef void(*	fnMicAudioHook) (const int nSamples, short *pSamples)
	Typedef for audio source (microphone) hook procedure. More...
typedef void(*	fnSpkrAudioHook) (const int nSamples, short *pSamples)
	Typedef for audio target(speaker) hook procedure. More...
typedef enum SIPX_AEC_MODE	SIPX_AEC_MODE
	SIPX_AEC_MODE defines different AEC modes. More...
typedef enum SIPX_NOISE_REDUCTION_MODE	SIPX_NOISE_REDUCTION_MODE
	SIPX_NOISE_REDUCTION_MODE defines the various noise reduction options. More...

Enumerations
enum	MEDIA_TYPE { AUDIO_MEDIA, VIDEO_MEDIA }
	Media type. More...
enum	SPEAKER_TYPE { SPEAKER, RINGER }
	Speaker output types are used to differentiate between the logical ringer (used to alert user of in inbound call) and speaker (in call audio device). More...
enum	SIPX_AUDIO_BANDWIDTH_ID {   AUDIO_CODEC_BW_VARIABLE =0, AUDIO_CODEC_BW_LOW, AUDIO_CODEC_BW_NORMAL, AUDIO_CODEC_BW_HIGH,   AUDIO_CODEC_BW_CUSTOM, AUDIO_CODEC_BW_DEFAULT }
	Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements. More...
enum	SIPX_VIDEO_BANDWIDTH_ID {   VIDEO_CODEC_BW_VARIABLE =0, VIDEO_CODEC_BW_LOW, VIDEO_CODEC_BW_NORMAL, VIDEO_CODEC_BW_HIGH,   VIDEO_CODEC_BW_CUSTOM, VIDEO_CODEC_BW_DEFAULT }
	Video Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements. More...
enum	SIPX_VIDEO_QUALITY_ID { VIDEO_QUALITY_LOW =1, VIDEO_QUALITY_NORMAL =2, VIDEO_QUALITY_HIGH =3 }
	Video Codec quality definitions. More...
enum	SIPX_VIDEO_FORMAT { VIDEO_FORMAT_CIF =0, VIDEO_FORMAT_QCIF, VIDEO_FORMAT_SQCIF, VIDEO_FORMAT_QVGA }
	Enumeration of possible video sizes. More...
enum	SIPX_AUDIO_DATA_FORMAT { RAW_PCM_16 =0 }
	Format definitions for memory resident audio data. More...
enum	SIPX_RESULT {   SIPX_RESULT_SUCCESS = 0, SIPX_RESULT_FAILURE, SIPX_RESULT_NOT_IMPLEMENTED, SIPX_RESULT_OUT_OF_MEMORY,   SIPX_RESULT_INVALID_ARGS, SIPX_RESULT_BAD_ADDRESS, SIPX_RESULT_OUT_OF_RESOURCES, SIPX_RESULT_INSUFFICIENT_BUFFER,   SIPX_RESULT_EVAL_TIMEOUT, SIPX_RESULT_BUSY, SIPX_RESULT_INVALID_STATE, SIPX_RESULT_MISSING_RUNTIME_FILES,   SIPX_RESULT_TLS_DATABASE_FAILURE, SIPX_RESULT_TLS_BAD_PASSWORD, SIPX_RESULT_TLS_TCP_IMPORT_FAILURE, SIPX_RESULT_NSS_FAILURE,   SIPX_RESULT_NOT_SUPPORTED, SIPX_RESULT_NETWORK_FAILURE, SIPX_RESULT_LIB_NOT_FOUND, SIPX_RESULT_OS_PLATFORM_UNSUPPORTED }
	SIPX_RESULT is an enumeration with all the possible result/return codes. More...
enum	SIPX_TONE_ID {   ID_DTMF_INVALID = 0, ID_DTMF_0 = '0', ID_DTMF_1 = '1', ID_DTMF_2 = '2',   ID_DTMF_3 = '3', ID_DTMF_4 = '4', ID_DTMF_5 = '5', ID_DTMF_6 = '6',   ID_DTMF_7 = '7', ID_DTMF_8 = '8', ID_DTMF_9 = '9', ID_DTMF_STAR = '*',   ID_DTMF_POUND = '#', ID_DTMF_FLASH = '!', ID_TONE_DIALTONE = 512, ID_TONE_BUSY,   ID_TONE_RINGBACK, ID_TONE_RINGTONE, ID_TONE_CALLFAILED, ID_TONE_SILENCE,   ID_TONE_BACKSPACE, ID_TONE_CALLWAITING, ID_TONE_CALLHELD, ID_TONE_LOUD_FAST_BUSY }
	DTMF/other tone ids used with sipxCallStartTone/sipxCallStopTone. More...
enum	SIPX_AUDIO_FILE_FORMAT { SIPX_UNKNOWN_FORMAT = 0, SIPX_WAVE_PCM_16, SIPX_WAVE_GSM, SIPX_OGG_OPUS }
	Audio file format types. More...
enum	SIPX_LOG_LEVEL {   LOG_LEVEL_DEBUG, LOG_LEVEL_INFO, LOG_LEVEL_NOTICE, LOG_LEVEL_WARNING,   LOG_LEVEL_ERR, LOG_LEVEL_CRIT, LOG_LEVEL_ALERT, LOG_LEVEL_EMERG,   LOG_LEVEL_NONE }
	Various log levels available for the sipxConfigEnableLog method. More...
enum	SIPX_SRTP_LEVEL { SRTP_LEVEL_NONE =0, SRTP_LEVEL_ENCRYPTION, SRTP_LEVEL_AUTHENTICATION, SRTP_LEVEL_ENCRYPTION_AND_AUTHENTICATION }
	Enumeration of the possible levels of SRTP. More...
enum	SIPX_CONTACT_TYPE {   CONTACT_LOCAL, CONTACT_NAT_MAPPED, CONTACT_RELAY, CONTACT_CONFIG,   CONTACT_AUTO = -1, CONTACT_ALL = -2 }
	SIPX_CONTACT_TYPE is an enumeration of possible address types for use with SIP contacts and SDP connection information. More...
enum	SIPX_TRANSPORT_TYPE { TRANSPORT_UDP = 1, TRANSPORT_TCP = 0, TRANSPORT_TLS = 3, TRANSPORT_CUSTOM = 4 }
	SIPX_TRANSPORT_TYPE defines various protocols use for signaling transport. More...
enum	SIPX_VIDEO_DISPLAY_TYPE { SIPX_WINDOW_HANDLE_TYPE, DIRECT_SHOW_FILTER }
	Enum for specifying the type of display object to be used for displaying video. More...
enum	SIPX_RTP_TRANSPORT {   SIPX_RTP_TRANSPORT_UNKNOWN = 0x00000000, SIPX_RTP_TRANSPORT_UDP = 0x00000001, SIPX_RTP_TRANSPORT_TCP = 0x00000002, SIPX_RTP_TCP_ROLE_ACTIVE = 0x00000004,   SIPX_RTP_TCP_ROLE_PASSIVE = 0x00000008, SIPX_RTP_TCP_ROLE_ACTPASS = 0x00000010, SIPX_RTP_TCP_ROLE_CONNECTION = 0x00000020 }
	Possible roles that a Media connection can have. More...
enum	SIPX_KEEPALIVE_TYPE { SIPX_KEEPALIVE_CRLF, SIPX_KEEPALIVE_STUN, SIPX_KEEPALIVE_SIP_PING, SIPX_KEEPALIVE_SIP_OPTIONS }
	SIPX_KEEPALIVE_TYPEs define different methods of keeping NAT/firewall port open. More...
enum	SIPX_AEC_MODE { SIPX_AEC_DISABLED, SIPX_AEC_SUPPRESS, SIPX_AEC_CANCEL, SIPX_AEC_CANCEL_AUTO }
	SIPX_AEC_MODE defines different AEC modes. More...
enum	SIPX_NOISE_REDUCTION_MODE { SIPX_NOISE_REDUCTION_DISABLED, SIPX_NOISE_REDUCTION_LOW, SIPX_NOISE_REDUCTION_MEDIUM, SIPX_NOISE_REDUCTION_HIGH }
	SIPX_NOISE_REDUCTION_MODE defines the various noise reduction options. More...

Functions
Initialization
SIPXTAPI_API SIPX_RESULT	sipxInitialize (SIPX_INST *phInst, const int udpPort=DEFAULT_UDP_PORT, const int tcpPort=DEFAULT_TCP_PORT, const int tlsPort=DEFAULT_TLS_PORT, const int rtpPortStart=DEFAULT_RTP_START_PORT, const int maxConnections=DEFAULT_CONNECTIONS, const char *szIdentity=DEFAULT_IDENTITY, const char *szBindToAddr=DEFAULT_BIND_ADDRESS, bool bUseSequentialPorts=false, const char *szTLSCertificateNickname=NULL, const char *szTLSCertificatePassword=NULL, const char *szDbLocation=NULL, bool bEnableLocalAudio=true, const int internalSamplerate=8000, const int devicesSamplerate=48000, const int internalFrameSizeMs=10, const char *callInputDeviceName="", const char *callOutputDeviceName="")
	Initialize the sipX TAPI-like API layer. More...
SIPXTAPI_API SIPX_RESULT	sipxReInitialize (SIPX_INST *phInst, const int udpPort=DEFAULT_UDP_PORT, const int tcpPort=DEFAULT_TCP_PORT, const int tlsPort=DEFAULT_TLS_PORT, const int rtpPortStart=DEFAULT_RTP_START_PORT, const int maxConnections=DEFAULT_CONNECTIONS, const char *szIdentity=DEFAULT_IDENTITY, const char *szBindToAddr=DEFAULT_BIND_ADDRESS, bool bUseSequentialPorts=false, const char *szTLSCertificateNickname=NULL, const char *szTLSCertificatePassword=NULL, const char *szDbLocation=NULL)
	Re-initialize the sipX TAPI-like API layer. More...
SIPXTAPI_API SIPX_RESULT	sipxUnInitialize (SIPX_INST hInst, bool bForceShutdown=false)
	Uninitialize the sipX TAPI-like API layer. More...
Call Methods
SIPXTAPI_API SIPX_RESULT	sipxCallAccept (const SIPX_CALL hCall, SIPX_VIDEO_DISPLAY *const pDisplay=NULL, SIPX_SECURITY_ATTRIBUTES *const pSecurity=NULL, SIPX_CALL_OPTIONS *options=NULL, bool bSendEarlyMedia=false)
	Accepts an inbound call and proceed immediately to alerting. More...
SIPXTAPI_API SIPX_RESULT	sipxCallReject (const SIPX_CALL hCall, const int errorCode=400, const char *szErrorText="Bad Request")
	Reject an inbound call (prior to alerting the user). More...
SIPXTAPI_API SIPX_RESULT	sipxCallRedirect (const SIPX_CALL hCall, const char *szForwardURL)
	Redirect an inbound call (prior to alerting the user). More...
SIPXTAPI_API SIPX_RESULT	sipxCallAnswer (const SIPX_CALL hCall, bool bTakeFocus=true)
	Answer an alerting call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallCreate (const SIPX_INST hInst, const SIPX_LINE hLine, SIPX_CALL *phCall)
	Create a new call for the purpose of creating an outbound connection/call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallConnect (const SIPX_CALL hCall, const char *szAddress, SIPX_CONTACT_ID contactId=0, SIPX_VIDEO_DISPLAY *const pDisplay=NULL, SIPX_SECURITY_ATTRIBUTES *const pSecurity=NULL, bool bTakeFocus=true, SIPX_CALL_OPTIONS *options=NULL, const char *szCallId=NULL)
	Connects an idle call to the designated target address. More...
SIPXTAPI_API SIPX_RESULT	sipxCallHold (const SIPX_CALL hCall, bool bStopRemoteAudio=true)
	Place the specified call on hold. More...
SIPXTAPI_API SIPX_RESULT	sipxCallUnhold (const SIPX_CALL hCall)
	Take the specified call off hold. More...
SIPXTAPI_API SIPX_RESULT	sipxCallDestroy (SIPX_CALL &hCall)
	Drop/Destroy the specified call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetID (const SIPX_CALL hCall, char *szId, const size_t iMaxLength)
	Get the SIP call ID of the call represented by the specified call handle. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetLocalID (const SIPX_CALL hCall, char *szId, const size_t iMaxLength)
	Get the SIP identity of the local connection. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetLine (const SIPX_CALL hCall, SIPX_LINE &hLine)
	Get the line handle for the given call if it has one. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetRemoteID (const SIPX_CALL hCall, char *szId, const size_t iMaxLength)
	Get the SIP identity of the remote connection. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetContactID (const SIPX_CALL hCall, char *szId, const size_t iMaxLength)
	Get the SIP identity of the contact connection. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetConnectionId (const SIPX_CALL hCall, int &connectionId)
	Gets the media interface connection ID. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetConference (const SIPX_CALL hCall, SIPX_CONF &hConf)
	Get the conference handle for the specified call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetRequestURI (const SIPX_CALL hCall, char *szUri, const size_t iMaxLength)
	Get the SIP request URI. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetRemoteContact (const SIPX_CALL hCall, char *szContact, const size_t iMaxLength)
	Get the SIP remote contact. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetRemoteUserAgent (const SIPX_CALL hCall, char *szAgent, const size_t iMaxLength)
	Get the remote user agent of the call represented by the specified call handle. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetInviteHeader (const SIPX_CALL hCall, const char *headerName, char *headerValue, const size_t maxValueLength, bool *inviteFromRemote, const size_t headerInstanceIndex=0)
	Get the named header value from the last SIP INVITE request (sent or received) in the context of the given call handle. More...
SIPXTAPI_API SIPX_RESULT	sipxCallStartTone (const SIPX_CALL hCall, const SIPX_TONE_ID toneId, const bool bLocal, const bool bRemote)
	Play a tone (DTMF, dialtone, ring back, etc) to the local and/or remote party. More...
SIPXTAPI_API SIPX_RESULT	sipxCallStopTone (const SIPX_CALL hCall)
	Stop playing a tone (DTMF, dialtone, ring back, etc). More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioPlayFileStart (const SIPX_CALL hCall, const char *szFile, const bool bRepeat, const bool bLocal, const bool bRemote, const bool bMixWithMicrophone=false, const float fVolumeScaling=1.0)
	Play the designated file. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioPlayFileStop (const SIPX_CALL hCall)
	Stop playing a file started with sipxCallPlayFileStart If a sipxCallDestroy is attempted while an audio file is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordFileStart (const SIPX_CALL hCall, const char *szFile, const SIPX_AUDIO_FILE_FORMAT recordFormat=SIPX_WAVE_PCM_16, const bool appendToFile=false, const int numChannels=1)
	Record a call session (including other parties if this is a multi-party call / conference) to a file. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordPause (const SIPX_CALL hCall)
	Pause recording call to file or buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordResume (const SIPX_CALL hCall)
	Resume paused recording call to file or buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordFileStop (const SIPX_CALL hCall)
	Stop recording a call to file. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordBufferStart (const SIPX_CALL hCall, const char *pBuffer, const int bufferSize, const int bufferType=RAW_PCM_16, const int maxRecordTime=-1, const int maxSilence=-1)
	Record a call session (including other parties if this is a multi-party call / conference) to a buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxCallAudioRecordBufferStop (const SIPX_CALL hCall)
	Stop recording a call to buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxCallPlayBufferStart (const SIPX_CALL hCall, const char *szBuffer, const int bufSize, const int bufType, const bool bRepeat, const bool bLocal, const bool bRemote)
	Play the specified audio data. More...
SIPXTAPI_API SIPX_RESULT	sipxCallPlayBufferStop (const SIPX_CALL hCall)
	Stop playing the audio started with sipxCallPlayBufferStart If a sipxCallDestroy is attempted while an audio buffer is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. More...
SIPXTAPI_API SIPX_RESULT	sipxCallSetMediaPassThrough (const SIPX_CALL hCall, MEDIA_TYPE mediaType, int mediaTypeStreamIndex, const char *streamReceiveAddress, int rtpPort, int rtcpPort)
	Set the address and port to which the given media stream is to be recieved. More...
SIPXTAPI_API SIPX_RESULT	sipxCallSubscribe (const SIPX_CALL hCall, const char *szEventType, const char *szAcceptType, SIPX_SUB *phSub, bool bRemoteContactIsGruu=false)
	Subscribe for NOTIFY events which may be published by the other end-point of the call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallUnsubscribe (const SIPX_SUB hSub)
	Unsubscribe from previously subscribed NOTIFY events. More...
SIPXTAPI_API SIPX_RESULT	sipxCallSendInfo (SIPX_INFO *phInfo, const SIPX_CALL hCall, const char *szContentType, const char *szContent, const size_t nContentLength)
	Sends an INFO event to the specified call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallBlindTransfer (const SIPX_CALL hCall, const char *szAddress)
	Blind transfer the specified call to another party. More...
SIPXTAPI_API SIPX_RESULT	sipxCallTransfer (const SIPX_CALL hSourceCall, const SIPX_CALL hTargetCall)
	Transfer the source call to the target call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallUpdateVideoWindow (const SIPX_CALL hCall, const SIPX_WINDOW_HANDLE hWnd)
	Updates the Video window with a new frame buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxCallResizeWindow (const SIPX_CALL hCall, const SIPX_WINDOW_HANDLE hWnd)
	Resizes the video window. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetEnergyLevels (const SIPX_CALL hCall, int &iInputEnergyLevel, int &iOutputEnergyLevel, const size_t nMaxContributors, unsigned int CCSRCs[], int iEnergyLevels[], size_t &nActualContributors)
	Gets energy levels for a call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetAudioRtpSourceIds (const SIPX_CALL hCall, unsigned int &iSendSSRC, unsigned int &iReceiveSSRC)
	Gets the sending and receiving Audio RTP SSRC IDs. More...
SIPXTAPI_API SIPX_RESULT	sipxCallGetAudioRtcpStats (const SIPX_CALL hCall, SIPX_RTCP_STATS *pStats)
	Obtain RTCP stats for the specified call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallLimitCodecs (const SIPX_CALL hCall, const char *codecNames)
	Restrict the set of codecs that are allowed to be used for the given call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallLimitCodecPreferences (const SIPX_CALL hCall, const SIPX_AUDIO_BANDWIDTH_ID audioBandwidth, const SIPX_VIDEO_BANDWIDTH_ID videoBandwidth, const char *szVideoCodecName)
	Limits the codec preferences on a per-call basis. More...
SIPXTAPI_API SIPX_RESULT	sipxCallSetMicGain (const SIPX_CALL hCall, float gain)
	Sets the gain of the microphone for a given call. More...
SIPXTAPI_API SIPX_RESULT	sipxCallSetMixOutputGain (const SIPX_CALL hCall, int bridgeOutputIndex, float gain)
	Sets the gain for all inputs to the given bridge output, except for the input of the same index which is set to 0.0f gain. More...
Publishing Methods
SIPXTAPI_API SIPX_RESULT	sipxPublisherCreate (const SIPX_INST hInst, SIPX_PUB *phPub, const char *szResourceId, const char *szEventType, const char *szContentType, const char *pContent, const size_t nContentLength)
	Creates a publishing context, which performs the processing necessary to accept SUBSCRIBE requests, and to publish NOTIFY messages to subscribers. More...
SIPXTAPI_API SIPX_RESULT	sipxPublisherDestroy (const SIPX_PUB hPub, const char *szContentType, const char *pFinalContent, const size_t nContentLength)
	Tears down the publishing context. More...
SIPXTAPI_API SIPX_RESULT	sipxPublisherUpdate (const SIPX_PUB hPub, const char *szContentType, const char *pContent, const size_t nContentLength)
	Publishes an updated state to specific event via NOTIFY to its subscribers. More...
Conference Methods
SIPXTAPI_API SIPX_RESULT	sipxConferenceCreate (const SIPX_INST hInst, SIPX_CONF *phConference)
	Create a conference handle. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceJoin (const SIPX_CONF hConf, const SIPX_CALL hCall)
	Join (add) an existing held call into a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceSplit (const SIPX_CONF hConf, const SIPX_CALL hCall)
	Split (remove) a held call from a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceAdd (const SIPX_CONF hConf, const SIPX_LINE hLine, const char *szAddress, SIPX_CALL *phNewCall, SIPX_CONTACT_ID contactId=0, SIPX_VIDEO_DISPLAY *const pDisplay=NULL, SIPX_SECURITY_ATTRIBUTES *const pSecurity=NULL, bool bTakeFocus=true, SIPX_CALL_OPTIONS *options=NULL)
	Add a new party to an existing conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceRemove (const SIPX_CONF hConf, const SIPX_CALL hCall)
	Removes a participant from conference by hanging up on them. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceGetCalls (const SIPX_CONF hConf, SIPX_CALL calls[], const size_t iMax, size_t &nActual)
	Gets all of the calls participating in a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceHold (const SIPX_CONF hConf, bool bBridging=true)
	Places a conference on hold. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceUnhold (const SIPX_CONF hConf)
	Removes conference members from a held state. More...
SIPXTAPI_API SIPX_RESULT	sipxConferencePlayAudioFileStart (const SIPX_CONF hConf, const char *szFile, const bool bRepeat, const bool bLocal, const bool bRemote, const bool bMixWithMicrophone=false, const float fVolumeScaling=1.0)
	Play the designated audio file to all conference participants and/or the local speaker. More...
SIPXTAPI_API SIPX_RESULT	sipxConferencePlayAudioFileStop (const SIPX_CONF hConf)
SIPXTAPI_API SIPX_RESULT	sipxConferenceDestroy (SIPX_CONF hConf)
	Destroys a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceGetEnergyLevels (const SIPX_CONF hConf, int &iInputEnergyLevel, int &iOutputEnergyLevel)
	Gets energy levels for a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxConferenceLimitCodecPreferences (const SIPX_CONF hConf, const SIPX_AUDIO_BANDWIDTH_ID audioBandwidth, const SIPX_VIDEO_BANDWIDTH_ID videoBandwidth, const char *szVideoCodecName)
	Limits the codec preferences on a conference. More...
SIPXTAPI_API SIPX_RESULT	sipxMediaConnectionCreate (const SIPX_CONF hConf, int &connectionId)
	Create a new media connetion that is indpendent of any existing call legs in the conference. More...
SIPXTAPI_API SIPX_RESULT	sipxMediaConnectionRtpSetDestination (const SIPX_CONF hConf, int connectionId, MEDIA_TYPE mediaType, int mediaTypeStreamIndex, const char *streamSendAddress, int rtpPort, int rtcpPort)
	Set the destination to which RTP streams are to be sent. More...
SIPXTAPI_API SIPX_RESULT	sipxMediaConnectionRtpStartSend (const SIPX_CONF hConf, int connectionId, int numTokens, const char *codecTokens[], int payloadIds[])
	Start sending the RTP/RTCP stream. More...
SIPXTAPI_API SIPX_RESULT	sipxMediaConnectionRtpStopSend (const SIPX_CONF hConf, int connectionId)
	Stop sending RTP/RTCP. More...
Audio Methods
SIPXTAPI_API SIPX_RESULT	sipxAudioSetGain (const SIPX_INST hInst, const int iLevel)
	Set the local microphone gain. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetGain (const SIPX_INST hInst, int &iLevel)
	Get the current microphone gain. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioMute (const SIPX_INST hInst, const bool bMute)
	Mute or unmute the microphone. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioIsMuted (const SIPX_INST hInst, bool &bMuted)
	Gets the mute state of the microphone. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioEnableSpeaker (const SIPX_INST hInst, const SPEAKER_TYPE type)
	Enables one of the speaker outputs. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetEnabledSpeaker (const SIPX_INST hInst, SPEAKER_TYPE &type)
	Gets the enabled speaker selection. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetVolume (const SIPX_INST hInst, const SPEAKER_TYPE type, const int iLevel)
	Sets the audio level for the designated speaker type. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetVolume (const SIPX_INST hInst, const SPEAKER_TYPE type, int &iLevel)
	Gets the audio level for the designated speaker type. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetAECMode (const SIPX_INST hInst, const SIPX_AEC_MODE mode)
	Enables or disables Acoustic Echo Cancellation (AEC). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetAECMode (const SIPX_INST hInst, SIPX_AEC_MODE &mode)
	Get the mode of Acoustic Echo Cancellation (AEC). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetAGCMode (const SIPX_INST hInst, const bool bEnable)
	Enable/Disable Automatic Gain Control (AGC). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetAGCMode (const SIPX_INST hInst, bool &bEnabled)
	Get the enable/disable state of Automatic Gain Control (AGC). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetNoiseReductionMode (const SIPX_INST hInst, const SIPX_NOISE_REDUCTION_MODE mode)
	Set the noise reduction mode/policy for suppressing background noise. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetNoiseReductionMode (const SIPX_INST hInst, SIPX_NOISE_REDUCTION_MODE &mode)
	Get the mode/policy for Noise Reduction (NR). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetNumInputDevices (const SIPX_INST hInst, size_t &numDevices)
	Get the number of input devices available on this system. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetInputDevice (const SIPX_INST hInst, const int index, const char *&szDevice)
	Get the name/identifier for input device at position index. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetCurrentInputDevice (const SIPX_INST hInst, int &index, const char *&szDevice)
	Get the name/identifier for the current input device used by sipX. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetNumOutputDevices (const SIPX_INST hInst, size_t &numDevices)
	Get the number of output devices available on this system. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetOutputDevice (const SIPX_INST hInst, const int index, const char *&szDevice)
	Get the name/identifier for output device at position index. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioGetCurrentOutputDevice (const SIPX_INST hInst, int &index, const char *&szDevice)
	Get the name/identifier for the current output device used by sipX. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetCallInputDevice (const SIPX_INST hInst, const char *szDevice)
	Set the call input device (in-call microphone). More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetRingerOutputDevice (const SIPX_INST hInst, const char *szDevice)
	Set the call ringer/alerting device. More...
SIPXTAPI_API SIPX_RESULT	sipxAudioSetCallOutputDevice (const SIPX_INST hInst, const char *szDevice)
	Set the call output device (in-call speaker). More...
Line / Identity Methods
SIPXTAPI_API SIPX_RESULT	sipxLineAdd (const SIPX_INST hInst, const char *szLineURL, SIPX_LINE *phLine, SIPX_CONTACT_ID contactId=0)
	Adds a line appearance. More...
SIPXTAPI_API SIPX_RESULT	sipxLineAddAlias (const SIPX_LINE hLine, const char *szLineURL)
	Adds an alias for a line definition. More...
SIPXTAPI_API SIPX_RESULT	sipxLineRegister (const SIPX_LINE hLine, const bool bRegister)
	Registers a line with the proxy server. More...
SIPXTAPI_API SIPX_RESULT	sipxLineRemove (SIPX_LINE hLine)
	Remove the designated line appearance. More...
SIPXTAPI_API SIPX_RESULT	sipxLineAddCredential (const SIPX_LINE hLine, const char *szUserID, const char *szPasswd, const char *szRealm)
	Adds authentication credentials to the designated line appearance. More...
SIPXTAPI_API SIPX_RESULT	sipxLineGet (const SIPX_INST hInst, SIPX_LINE lines[], const size_t max, size_t &actual)
	Gets the active list of line identities. More...
SIPXTAPI_API SIPX_RESULT	sipxLineGetURI (const SIPX_LINE hLine, char *szBuffer, const size_t nBuffer, size_t &nActual)
	Get the Line URI for the designated line handle. More...
SIPXTAPI_API SIPX_RESULT	sipxLineGetContactId (const SIPX_LINE hLine, SIPX_CONTACT_ID &contactId)
	Get the contact ID for the designated line handle. More...
SIPXTAPI_API SIPX_RESULT	sipxLineFindByURI (const SIPX_INST hInst, const char *szURI, SIPX_LINE &hLine)
	Find a line definition given a URI. More...
Configuration Methods
SIPXTAPI_API SIPX_RESULT	sipxConfigSetLogLevel (SIPX_LOG_LEVEL logLevel)
	The sipxConfigEnableLog method enables logging for the sipXtapi API, media processing, call processing, SIP stack, and OS abstraction layer. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetLogFile (const char *szFilename)
	The sipxConfigSetlogFile method sets the filename of the log file and directs output to that file. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetLogCallback (sipxLogCallback pCallback)
	Set a callback function to collect logging information. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableGIPSTracing (SIPX_INST hInst, bool bEnable)
	Enables GIPS tracing in sipXtapi (if bundled with sipXtapi). More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetMicAudioHook (fnMicAudioHook hookProc)
	Designate a callback routine as a microphone replacement or supplement. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetSpkrAudioHook (fnSpkrAudioHook hookProc)
	Designate a callback routine for post-mixing audio data (e.g. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetEnergyLevelNotificationPeriod (const SIPX_INST hInst, int periodMs)
SIPXTAPI_API SIPX_RESULT	sipxConfigSetUserAgentName (const SIPX_INST hInst, const char *szName, const bool bIncludePlatformName=true)
	Sets the User-Agent name to be used with outgoing SIP messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetOutboundProxy (const SIPX_INST hInst, const char *szProxy)
	Defines the SIP proxy used for outbound requests. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetDnsSrvTimeouts (const int initialTimeoutInSecs, const int retries)
	Modifies the timeout values used for DNS SRV lookups. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetRegisterResponseWaitSeconds (const SIPX_INST hInst, const int seconds)
	Specifies the time to wait for a REGISTER response before sending a LINESTATE_REGISTER_FAILED (or LINESTATE_UNREGISTER_FAILED) message. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetDnsSrvFailoverTimeout (const SIPX_INST hInst, const int failoverTimeoutInSecs)
	Specifies the time to wait before trying the next DNS SRV record. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableRport (const SIPX_INST hInst, const bool bEnable)
	Enable or disable the use of "rport". More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableRportMapping (const SIPX_INST hInst, const bool bEnable)
	Enable or diable the use of IP and port provided in the response via (rport) for mapping of NAT/public IP address and port in the contact database. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetRegisterExpiration (const SIPX_INST hInst, const int nRegisterExpirationSecs)
	Specifies the expiration period for registration. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetSubscribeExpiration (const SIPX_INST hInst, const int nSubscribeExpirationSecs)
	Specifies the expiration period for subscription. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableStun (const SIPX_INST hInst, const char *szServer, int iServerPort, int iKeepAliveInSecs)
	Enables STUN (Simple Traversal of UDP through NAT) support for both UDP SIP signaling and UDP audio/video (RTP). More...
SIPXTAPI_API SIPX_RESULT	sipxConfigDisableStun (const SIPX_INST hInst)
	Disable the use of STUN. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableTurn (const SIPX_INST hInst, const char *szServer, const int iServerPort, const char *szUsername, const char *szPassword, const int iKeepAliveInSecs)
	Enable TURN for support for UDP audio/video (RTP). More...
SIPXTAPI_API SIPX_RESULT	sipxConfigDisableTurn (const SIPX_INST hInst)
	Disable the use of TURN. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableIce (const SIPX_INST hInst)
	Enables an ICE-like mechanism for determining connectivity of remote parties dynamically. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigDisableIce (const SIPX_INST hInst)
	Disable the use of ICE. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigKeepAliveAdd (const SIPX_INST hInst, SIPX_CONTACT_ID contactId, SIPX_KEEPALIVE_TYPE type, const char *remoteIp, int remotePort, int keepAliveSecs)
	Add a signaling keep alive to a remote ip address. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigKeepAliveRemove (const SIPX_INST hInst, SIPX_CONTACT_ID contactId, SIPX_KEEPALIVE_TYPE type, const char *remoteIp, int remotePort)
	Remove a signaling keepalive. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableOutOfBandDTMF (const SIPX_INST hInst, const bool bEnable)
	Enable/disable sending of RFC 2833 DTMF tones. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableInBandDTMF (const SIPX_INST hInst, const bool bEnable)
	Enable/disable sending of in-band DTMF tones. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableRTCP (const SIPX_INST hInst, const bool bEnable)
	Enable or disable sending RTCP reports. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableDnsSrv (const bool bEnable)
	Enables/disables sending of DNS SRV request for all sipXtapi instances. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigIsOutOfBandDTMFEnabled (const SIPX_INST hInst, bool &bEnable)
	Determines if sending of out-of-band DTMF tones is enabled or disabled. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigIsInBandDTMFEnabled (const SIPX_INST hInst, bool &bEnable)
	Determines if sending of in-band DTMF tones is enabled or disabled. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetVersion (char *szVersion, const size_t nBuffer)
	Get the sipXtapi API version string. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetLocalSipUdpPort (SIPX_INST hInst, int *pPort)
	Get the local UDP port for SIP signaling. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigAddContact (const SIPX_INST hInst, const char *szSipContactAddress, const int iSipContactPort, SIPX_CONTACT_ID &iNewContactId)
	Add a contact using the given IP address and port. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetLocalSipTcpPort (SIPX_INST hInst, int *pPort)
	Get the local TCP port for SIP signaling. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetLocalSipTlsPort (SIPX_INST hInst, int *pPort)
	Get the local TLS port for SIP signaling. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetAudioCodecPreferences (const SIPX_INST hInst, SIPX_AUDIO_BANDWIDTH_ID bandWidth)
	Set the preferred bandwidth requirement for codec selection. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetAudioCodecByName (const SIPX_INST hInst, const char *szCodecName)
	Set the codec preference order by name. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetAudioCodecPreferences (const SIPX_INST hInst, SIPX_AUDIO_BANDWIDTH_ID *pBandWidth)
	Get the current codec preference. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetNumAudioCodecs (const SIPX_INST hInst, int *pNumCodecs)
	Get the number of audio codecs. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetAudioCodec (const SIPX_INST hInst, const int index, SIPX_AUDIO_CODEC *pCodec)
	Get the audio codec at a certain index in the list of codecs. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoBandwidth (const SIPX_INST hInst, SIPX_VIDEO_BANDWIDTH_ID bandWidth)
	Set the bandwidth parameters for video codecs.Depending on the bandwidth parameter that is passed in the settings will be set to: More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetVideoCaptureDevices (const SIPX_INST hInst, char **arrSzCaptureDevices, int nDeviceStringLength, int nArrayLength)
	Gets the list of video capture devices. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetVideoCaptureDevice (const SIPX_INST hInst, char *szCaptureDevice, int nLength)
	Gets the current video capture device. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoCaptureDevice (const SIPX_INST hInst, const char *szCaptureDevice)
	Sets the video capture device. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoCodecByName (const SIPX_INST hInst, const char *szCodecName)
	Set the codec by name. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigResetVideoCodecs (const SIPX_INST hInst)
	Reset the codec list if it was modified by sipxConfigSetVideoCodecByName. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetVideoCodecPreferences (const SIPX_INST hInst, SIPX_VIDEO_BANDWIDTH_ID *pBandWidth)
	Get the current codec preference. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetNumVideoCodecs (const SIPX_INST hInst, int *pNumCodecs)
	Get the number of video codecs. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoFormat (const SIPX_INST hInst, SIPX_VIDEO_FORMAT videoFormat)
	Set the supported video format This method will limit the supported video format to either VIDEO_FORMAT_CIF (352x288), VIDEO_FORMAT_QCIF (176x144), VIDEO_FORMAT_SQCIF (128x92), or VIDEO_FORMAT_QVGA (320x240). More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetVideoCodec (const SIPX_INST hInst, const int index, SIPX_VIDEO_CODEC *pCodec)
	Get the video codec at a certain index in the list of codecs. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetLocalContacts (const SIPX_INST hInst, SIPX_CONTACT_ADDRESS addresses[], size_t nMaxAddresses, size_t &nActualAddresses)
	Get the local contact address available for outbound/inbound signaling and audio. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetLocalFeedbackAddress (const SIPX_INST hInst, const char *szRemoteIp, const int iRemotePort, char *szContactIp, size_t nContactIpLength, int &iContactPort, int iTimeoutMs)
	Get our local ip/port combination for the designated remote ip/port. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigGetAllLocalNetworkIps (const char *arrAddresses[], const char *arrAddressAdapter[], int &numAddresses)
	Populates an array of IP Addresses in char* form. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetSecurityParameters (const SIPX_INST hInst, const char *szDbLocation, const char *szMyCertNickname, const char *szDbPassword)
	Set security parameters for an instance of sipXtapi. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableSipShortNames (const SIPX_INST hInst, const bool bEnabled)
	Enables/Disables use of short field names in sip messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableSipDateHeader (const SIPX_INST hInst, const bool bEnabled)
	Enables/Disables use of date header in sip messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableSipAllowHeader (const SIPX_INST hInst, const bool bEnabled)
	Enables/Disables use of allow header in sip messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetSipAcceptLanguage (const SIPX_INST hInst, const char *szLanguage)
	Sets the Accept Language used in sip messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetLocationHeader (const SIPX_INST hInst, const char *szHeader)
	Sets the location header for SIP messages. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetConnectionIdleTimeout (const SIPX_INST hInst, const int idleTimeout)
	Set the connection idle timeout. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigPrepareToHibernate (const SIPX_INST hInst)
	Call this function to prepare a sipXtapi instance for a system hibernation. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigUnHibernate (const SIPX_INST hInst)
	Call this function upon returning from a system hibernation. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigEnableRtpOverTcp (const SIPX_INST hInst, bool bEnable)
	Enables RTP streaming over TCP. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoPreviewDisplay (const SIPX_INST hInst, SIPX_VIDEO_DISPLAY *const pDisplay)
	Sets the display object for the "video preview". More...
SIPXTAPI_API SIPX_RESULT	sipxConfigUpdatePreviewWindow (const SIPX_INST hInst, const SIPX_WINDOW_HANDLE hWnd)
	Updates the Preview window with a new frame buffer. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoQuality (const SIPX_INST hInst, const SIPX_VIDEO_QUALITY_ID quality)
	Sets the video quality. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoParameters (const SIPX_INST hInst, const int bitRate, const int frameRate)
	Sets the bit rate and frame rate parameters for video. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoBitrate (const SIPX_INST hInst, const int bitRate)
	Sets the video bitrate. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoFramerate (const SIPX_INST hInst, const int frameRate)
	Sets the video framerate. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVideoCpuUsage (const SIPX_INST hInst, const int cpuUsage)
	Set the cpu usage. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSubscribe (const SIPX_INST hInst, const SIPX_LINE hLine, const char *szTargetUrl, const char *szEventType, const char *szAcceptType, const SIPX_CONTACT_ID contactId, SIPX_SUB *phSub)
	Subscribe for NOTIFY events which may be published by another end-point or server. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigUnsubscribe (const SIPX_SUB hSub)
	Unsubscribe from previously subscribed NOTIFY events. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigExternalTransportAdd (SIPX_INST const hInst, SIPX_TRANSPORT &hTransport, const bool bIsReliable, const char *szTransport, const char *szLocalIP, const int iLocalPort, SIPX_TRANSPORT_WRITE_PROC writeProc, const char *szLocalRoutingId, const void *pUserData=NULL)
	Associates an external transport mechanism for SIP signaling with the given instance. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigExternalTransportRemove (const SIPX_TRANSPORT hTransport)
	Removes an external transport mechanism from the given instance. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigExternalTransportRouteByUser (const SIPX_TRANSPORT hTransport, bool bRouteByUser)
	The external transport mechanism can be configured to route by user or by destination ip/port. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigExternalTransportHandleMessage (const SIPX_TRANSPORT hTransport, const char *szSourceIP, const int iSourcePort, const char *szLocalIP, const int iLocalPort, const void *pData, const size_t nData)
	Called by the application when it receives a complete SIP message via it's external transport mechanism and want to pass it along to sipXtapi. More...
SIPXTAPI_API SIPX_RESULT	sipxConfigSetVoiceQualityServer (const SIPX_INST hInst, const char *szServer)
	Sets the SIP target URL for voice quality reports. More...
Utility Functions
SIPXTAPI_API SIPX_RESULT	sipxUtilUrlParse (const char *szUrl, char *szUsername, char *szHostname, int *iPort)
	Simple utility function to parse the username, host, and port from a URL. More...
SIPXTAPI_API SIPX_RESULT	sipxUtilUrlGetDisplayName (const char *szUrl, char *szDisplayName, size_t nDisplayName)
	Simple utility function to parse the display name from a SIP URL. More...
SIPXTAPI_API SIPX_RESULT	sipxUtilUrlUpdate (char *szUrl, size_t &nUrl, const char *szNewUsername, const char *szNewHostname, const int iNewPort)
	Simple utility function to update a URL. More...
SIPXTAPI_API SIPX_RESULT	sipxUtilUrlGetUrlParam (const char *szUrl, const char *szParamName, size_t nParamIndex, char *szParamValue, size_t nParamValue)
	Get the Nth named url parameter from the designated url. More...

Variables
const SIPX_INST	SIPX_INST_NULL = 0
	Represents a null instance handle. More...
const SIPX_LINE	SIPX_LINE_NULL = 0
	Represents a null line handle. More...
const SIPX_CALL	SIPX_CALL_NULL = 0
	Represents a null call handle. More...
const SIPX_CONF	SIPX_CONF_NULL = 0
	Represents a null conference handle. More...
const SIPX_PUB	SIPX_PUB_NULL = 0
	Represents a null publisher handle. More...
const SIPX_TRANSPORT	SIPX_TRANSPORT_NULL = 0
	Represents a null transport handle. More...

Detailed Description

sipXtapi main API declarations

Macro Definition Documentation

#define CONF_MAX_CONNECTIONS   32

Max number of conference participants.

#define DEFAULT_BIND_ADDRESS   "0.0.0.0"

Bind to the first physical interface discovered.

#define DEFAULT_CONNECTIONS   32

Default number of max sim.

conns.

#define DEFAULT_IDENTITY   "sipx"

sipx<IP>:UDP_PORT used as identify if lines are not defined.

This define only controls the userid portion of the SIP url.

#define DEFAULT_RTP_START_PORT   9000

Starting RTP port for RTP port range.

The user agent will use ports ranging from the start port to the start port

• (default connections * 2).

#define DEFAULT_STUN_PORT   3478

Default stun server port.

#define DEFAULT_TCP_PORT   5060

Default TCP port.

#define DEFAULT_TLS_PORT   5061

Default TLS port.

#define DEFAULT_UDP_PORT   5060

Default UDP port.

#define GAIN_DEFAULT   70

Nominal gain value.

#define GAIN_MAX   100

Max acceptable gain value.

#define GAIN_MIN   0

Min acceptable gain value.

This gain will mute mic.

#define MAX_AUDIO_DEVICES   64

Max number of input/output audio devices.

#define MAX_PASSWORD_LENGTH   32

maximum password length PKI operations

#define MAX_PKCS12_KEY_LENGTH   4096

pkcs12 key length

#define MAX_SMIME_KEY_LENGTH   2048

s/mime key length

#define MAX_SRTP_KEY_LENGTH   31

srtp key length

#define MAX_VIDEO_DEVICE_LENGTH   256

Max length of video capture device string.

#define MAX_VIDEO_DEVICES   8

Max number of video capture devices.

#define SIPX_CALLING_CONVENTION

The SIPX_CALLING_CONVENTION define controls the default calling convention.

define "SIPX_USE_STDCALL" to use the __stdcall calling convention under MSVC.

#define SIPX_MAX_ADAPTER_NAME_LENGTH   256

Max length of an adapter name.

#define SIPX_MAX_CALLS   64

Maximum number of simultaneous calls.

#define SIPX_MAX_IP_ADDRESSES   32

Maximum number of IP addresses on the host.

#define SIPX_PORT_AUTO   -2

Special value that instructs sipXtapi to automatically select an open port for signaling or audio when passed to sipXinitialize.

#define SIPX_PORT_DISABLE   -1

Special value that disables the transport type (e.g.

UDP, TCP, or TLS) when passed to sipXinitialize

#define SIPXTAPI_API   extern "C"

Assume extern "C" for non-win32 platforms.

#define SIPXTAPI_BUILD_WORD   3,3,0,0

Default build word – automatically filled in during release process.

#define SIPXTAPI_BUILDDATE   "0000-00-00"

Default build date – automatically filled in during release process.

#define SIPXTAPI_BUILDNUMBER   "0"

Default build number – automatically filled in during release process.

#define SIPXTAPI_CODEC_NAMELEN   32

Maximum length for codec name.

#define SIPXTAPI_CODEC_NAMELEN   32

Maximum length for codec name.

#define SIPXTAPI_FULL_VERSION   "3.3.0.0"

Default full version number – automatically filled in during release process.

#define SIPXTAPI_VERSION   "3.3.0"

sipXtapi API version – automatically filled in during release process

#define SIPXTAPI_VERSION_STRING   "sipXtapi SDK %s.%s %s (built %s)"

Version string format string.

#define VOLUME_DEFAULT   70

Nominal volume value.

#define VOLUME_MAX   100

Max acceptable volume value.

#define VOLUME_MIN   0

Min acceptable volume value.

Typedef Documentation

typedef void(* fnMicAudioHook) (const int nSamples, short *pSamples)

Typedef for audio source (microphone) hook procedure.

This typedef coupled with the sipxConfigSetMicAudioHook API allows developers to view, modify or substitute microphone data.

Parameters

nSamples	number of 16 bit unsigned PCM samples
pSamples	pointer to array of samples.

typedef void(* fnSpkrAudioHook) (const int nSamples, short *pSamples)

Typedef for audio target(speaker) hook procedure.

This typedef coupled with the sipxConfigSetSpkrAudioHook API allows developers to intercept and modify audio headed for the speaker.

Parameters

nSamples	number of 16 bit unsigned samples
pSamples	pointer to array of samples

typedef enum MEDIA_TYPE MEDIA_TYPE

Media type.

typedef enum SIPX_AEC_MODE SIPX_AEC_MODE

SIPX_AEC_MODE defines different AEC modes.

Options included DISABLED, SUPPRESS, CANCEL, and CANCEL_AUTO.

NOTE: This functionally is only supported when sipXtapi is bundled with VoiceEngine from Global IP Sound or Speex library.

NOTE: SIPX_AEC_SUPPRESS is not available if Speex library used.

typedef enum SIPX_AUDIO_BANDWIDTH_ID SIPX_AUDIO_BANDWIDTH_ID

Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements.

Bandwidth requirements for supported codecs:

High:     IPCMWB    ~ 80 kbps
Normal:   PCMU      64 kbps
          PCMA      64 kbps
Low:      iLBC      13.33 kbps, 30 ms frame size and
                    15.2 kbps, 20 ms frame size
          GSM       13 kbps, 20 ms frame size
          G729      8 Kbps, 10ms frame size
          SPEEX     8 Kbps
          SPEEX_5   5.95 Kbps
          SPEEX_15  15 Kbps
          SPEEX_24  24.6 Kbps
Variable: ISAC      variable bitrate

typedef enum SIPX_AUDIO_DATA_FORMAT SIPX_AUDIO_DATA_FORMAT

Format definitions for memory resident audio data.

typedef enum SIPX_AUDIO_FILE_FORMAT SIPX_AUDIO_FILE_FORMAT

Audio file format types.

typedef unsigned int SIPX_CALL

The SIPX_CALL handle represents a call or connection between the user agent and another party.

All call operations require the call handle as a parameter.

typedef unsigned long SIPX_CONF

The SIPX_CONF handle represents a collection of CALLs that have bridge (mixed) audio.

Application developers can manipulate each leg of the conference through various conference functions.

typedef int SIPX_CONTACT_ID

Type for storing Contact Record identifiers.

See also

sipxConfigGetLocalContacts

typedef unsigned long SIPX_INFO

The SIPX_INFO handle represents a handle to an INFO message sent by a sipXtapi instance.

INFO messages are useful for communicating information between user agents within a logical call. The SIPX_INFO handle is returned when sending an INFO message via sipxCallSendInfo(...). The handle is references as part of the EVENT_CATEGORY_INFO_STATUS event callback/observer. sipXtapi will automatically deallocate this handle immediately after the status call back.

typedef void* SIPX_INST

The SIPX_INST handle represents an instance of a user agent.

A user agent includes a SIP stack and media processing framework. sipXtapi does support multiple instances of user agents in the same process space, however, certain media processing features become limited or ambiguous. For example, only one user agent should control the local system's input and output audio devices.

typedef unsigned int SIPX_LINE

The SIPX_LINE handle represents an inbound or outbound identity.

When placing outbound the application programmer must define the outbound line. When receiving inbound calls, the application can query the line.

typedef enum SIPX_LOG_LEVEL SIPX_LOG_LEVEL

Various log levels available for the sipxConfigEnableLog method.

Developers can choose the amount of detail available in the log. Each level includes messages generated at lower levels. For example, LOG_LEVEL_EMERG will limit the log to emergency messages, while LOG_LEVEL_ERR includes emergency messages, alert messages, critical messages, and errors. LOG_LEVEL_ERR is probably best for general runtime situations. LOG_LEVEL_INFO or LOG_LEVEL_DEBUG is best for diagnosing problems.

typedef enum SIPX_NOISE_REDUCTION_MODE SIPX_NOISE_REDUCTION_MODE

SIPX_NOISE_REDUCTION_MODE defines the various noise reduction options.

Options include, DISABLED, LOW, MEDIUM, and HIGH. When selecting a noise reduction level, you are trading off reducing back ground noise with the possibility of suppressing speech. We recommend selecting the LOW level.

NOTE: This functionally is only supported when sipXtapi is bundled with VoiceEngine from Global IP Sound or Speex library.

NOTE: When Speex library is used there is no difference between SIPX_NOISE_REDUCTION_LOW, SIPX_NOISE_REDUCTION_MEDIUM and SIPX_NOISE_REDUCTION_HIGH.

typedef unsigned long SIPX_PUB

The SIPX_PUB handle represent a publisher context.

Publisher are used to publish application-data to interested parties (Subscribers). This maps directly to the SIP SUBSCRIBE, and NOTIFY methods. The handle is used to mange the life cycle of the publisher.

SIPX_PUB handles are created by using sipxCreatePublisher. SIPX_PUB handles should be torn down using sipxDestroyPublisher.

typedef enum SIPX_RESULT SIPX_RESULT

SIPX_RESULT is an enumeration with all the possible result/return codes.

typedef unsigned long SIPX_SUB

A SIPX_SUB handle represent a subscription to a remote publisher.

This maps directly to the SIP SUBSCRIBE, and NOTIFY methods. The handle is used to mange the life cycle of the subscription.

SIPX_SUB handles are created by using the sipxCallSubscribe function. SIPX_SUB handles should be destroyed using the sipxCallUnsubscribe function.

typedef enum SIPX_TONE_ID SIPX_TONE_ID

DTMF/other tone ids used with sipxCallStartTone/sipxCallStopTone.

typedef unsigned long SIPX_TRANSPORT

The SIPX_TRANSPORT handle represents a user-implementation of a network transport mechanism to be used for SIP signaling.

typedef bool(SIPX_CALLING_CONVENTION * SIPX_TRANSPORT_WRITE_PROC) (SIPX_TRANSPORT hTransport, const char *szDestinationIp, const int iDestPort, const char *szLocalIp, const int iLocalPort, const void *pData, const size_t nData, const void *pUserData)

External Transport callback definition.

SIPxua will invoke this callback when there is data to write to the external transport layer. A function pointer with this signature is passed into the sipxConfigExternalTransportAdd function.

Parameters

hTransport	Handle to the external transport object. Will match a transport handle obtained via a call to sipxConfigExternalTransportAdd
szDestinationIp	IP address which is the destination for the write.
iDestPort	Port value to which the data will be sent.
szLocalIp	IP address which is the source address for the write.
iLocalPort	Port value from which the data will be sent.
pData	Pointer to the data to be sent.
nData	Size of the data to be sent.

typedef enum SIPX_VIDEO_BANDWIDTH_ID SIPX_VIDEO_BANDWIDTH_ID

Video Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements.

The codec name is a combination of the actual codec name and the video resolution.

Supported codecs are:

VP71, IYUV, I420, and RGB24

Supported resolutions are

CIF (352x288), QCIF (176x144), SQCIF (128x96), and QVGA (320x240)

A VP71 codec in QCIF resolution would be named VP71-QCIF.

typedef enum SIPX_VIDEO_DISPLAY_TYPE SIPX_VIDEO_DISPLAY_TYPE

Enum for specifying the type of display object to be used for displaying video.

typedef enum SIPX_VIDEO_FORMAT SIPX_VIDEO_FORMAT

Enumeration of possible video sizes.

typedef enum SIPX_VIDEO_QUALITY_ID SIPX_VIDEO_QUALITY_ID

Video Codec quality definitions.

Quality is used as a trade off between between CPU usage and the amount of bandwidth used.

typedef void* SIPX_WINDOW_HANDLE

Type for storing a "window object handle" - in Windows, the application should cast their HWND to a SIPX_WINDOW_HANDLE.

typedef void(* sipxLogCallback) (const char *szPriority, const char *szSource, const char *szMsg)

Signature for a log callback function that gets passed three strings, first string is the priority level, second string is the source id of the subsystem that generated the message, and the third string is the message itself.

typedef enum SPEAKER_TYPE SPEAKER_TYPE

Speaker output types are used to differentiate between the logical ringer (used to alert user of in inbound call) and speaker (in call audio device).

typedef SIPX_TONE_ID TONE_ID

Left for backwards compatibility – please use SIPX_TONE_ID.

Enumeration Type Documentation

enum MEDIA_TYPE

Media type.

Enumerator
AUDIO_MEDIA
VIDEO_MEDIA

enum SIPX_AEC_MODE

SIPX_AEC_MODE defines different AEC modes.

Options included DISABLED, SUPPRESS, CANCEL, and CANCEL_AUTO.

NOTE: This functionally is only supported when sipXtapi is bundled with VoiceEngine from Global IP Sound or Speex library.

NOTE: SIPX_AEC_SUPPRESS is not available if Speex library used.

Enumerator
SIPX_AEC_DISABLED	Disabled AEC; do not attempt to cancel or suppress echo.
SIPX_AEC_SUPPRESS	Echo suppression; attempt to suppress echo by effectively forcing a half-duplex audio channel. If you are speaking, the speaker will be silenced to avoid echo. Echo cancellation is consider a better approach/experience, however, requires more CPU consumption.
SIPX_AEC_CANCEL	Full echo cancellation; attempt to cancel echo between the the speaker and microphone. Depending on the quality of your speaker/microphone, this may result in some suppression. For example, if either the speaker or microphone distorts the signal (making it non-linear), it is becomes increasingly difficult to cancel. This is consider a full-duplex solution.
SIPX_AEC_CANCEL_AUTO	Full echo cancellation; attempt to cancel echo between the the speaker and microphone; however, automatically disable echo cancellation if it appears not needed.

enum SIPX_AUDIO_BANDWIDTH_ID

Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements.

Bandwidth requirements for supported codecs:

High:     IPCMWB    ~ 80 kbps
Normal:   PCMU      64 kbps
          PCMA      64 kbps
Low:      iLBC      13.33 kbps, 30 ms frame size and
                    15.2 kbps, 20 ms frame size
          GSM       13 kbps, 20 ms frame size
          G729      8 Kbps, 10ms frame size
          SPEEX     8 Kbps
          SPEEX_5   5.95 Kbps
          SPEEX_15  15 Kbps
          SPEEX_24  24.6 Kbps
Variable: ISAC      variable bitrate

Enumerator
AUDIO_CODEC_BW_VARIABLE	ID for codecs with variable bandwidth requirements.
AUDIO_CODEC_BW_LOW	ID for codecs with low bandwidth requirements.
AUDIO_CODEC_BW_NORMAL	ID for codecs with normal bandwidth requirements.
AUDIO_CODEC_BW_HIGH	ID for codecs with high bandwidth requirements.
AUDIO_CODEC_BW_CUSTOM	Possible return value for sipxConfigGetAudioCodecPreferences. This ID indicates the available list of codecs was overridden by a sipxConfigSetAudioCodecByName call.
AUDIO_CODEC_BW_DEFAULT	Value used to signify the default bandwidth level when calling sipxCallConnect, sipxCallAccept, or sipxConferenceAdd.

enum SIPX_AUDIO_DATA_FORMAT

Format definitions for memory resident audio data.

Enumerator
RAW_PCM_16	Signed 16 bit PCM data, mono, 8KHz, no header.

enum SIPX_AUDIO_FILE_FORMAT

Audio file format types.

Enumerator
SIPX_UNKNOWN_FORMAT
SIPX_WAVE_PCM_16	Wave audio file with PCM 16 bit, signed, little endian encoded audio.
SIPX_WAVE_GSM	Wave audio file with GSM encoded audio.
SIPX_OGG_OPUS	OGG file container using OPUS audio encoder.

enum SIPX_CONTACT_TYPE

SIPX_CONTACT_TYPE is an enumeration of possible address types for use with SIP contacts and SDP connection information.

Application developers and choose to setup calls with specific contact types (e.g. use my local IP address, a stun-derived IP address, turn-derived IP address, etc). Unless you have complete knowledge and control of your network environment, you should likely use CONTACT_AUTO.

Enumerator
CONTACT_LOCAL	Local address for a particular interface.
CONTACT_NAT_MAPPED	NAT mapped address (e.g. STUN)
CONTACT_RELAY	Relay address (e.g. TURN)
CONTACT_CONFIG	Manually configured address.
CONTACT_AUTO	Automatic contact selection; used for API parameters.
CONTACT_ALL

enum SIPX_KEEPALIVE_TYPE

SIPX_KEEPALIVE_TYPEs define different methods of keeping NAT/firewall port open.

These approaches are used for the signaling path of a call and are generally only needed under specific network configurations.

Examples: - When not using a proxy

• When the registration period is longer then NAT bindings timeout

The STUN, and SIP_PING and SIP_OPTIONS events may also give you more information about your network NAT mappings. When you add a keepalive, you may get KEEPALIVE_FEEDBACK events with the IP/port that your peer thinks is you. For STUN, this comes from the STUN response, for the SIP keepalives, this comes from the "via" response if the remote supports rport/symmetric signaling.

Enumerator
SIPX_KEEPALIVE_CRLF	Send a Carriage Return/Line Feed to other side.
SIPX_KEEPALIVE_STUN	Send a Stun request to the other side.
SIPX_KEEPALIVE_SIP_PING	Send a SIP PING method request to the other side.
SIPX_KEEPALIVE_SIP_OPTIONS	Send a SIP OPTIONS method request to the other side.

enum SIPX_LOG_LEVEL

Various log levels available for the sipxConfigEnableLog method.

Developers can choose the amount of detail available in the log. Each level includes messages generated at lower levels. For example, LOG_LEVEL_EMERG will limit the log to emergency messages, while LOG_LEVEL_ERR includes emergency messages, alert messages, critical messages, and errors. LOG_LEVEL_ERR is probably best for general runtime situations. LOG_LEVEL_INFO or LOG_LEVEL_DEBUG is best for diagnosing problems.

Enumerator
LOG_LEVEL_DEBUG	debug-level messages
LOG_LEVEL_INFO	informational messages
LOG_LEVEL_NOTICE	normal, but significant, conditions
LOG_LEVEL_WARNING	warning conditions
LOG_LEVEL_ERR	error conditions
LOG_LEVEL_CRIT	critical conditions
LOG_LEVEL_ALERT	action must be taken immediately
LOG_LEVEL_EMERG	system is unusable
LOG_LEVEL_NONE	disable logging

enum SIPX_NOISE_REDUCTION_MODE

SIPX_NOISE_REDUCTION_MODE defines the various noise reduction options.

Options include, DISABLED, LOW, MEDIUM, and HIGH. When selecting a noise reduction level, you are trading off reducing back ground noise with the possibility of suppressing speech. We recommend selecting the LOW level.

NOTE: This functionally is only supported when sipXtapi is bundled with VoiceEngine from Global IP Sound or Speex library.

NOTE: When Speex library is used there is no difference between SIPX_NOISE_REDUCTION_LOW, SIPX_NOISE_REDUCTION_MEDIUM and SIPX_NOISE_REDUCTION_HIGH.

Enumerator
SIPX_NOISE_REDUCTION_DISABLED	Disable NR; Do not attempt to reduce background noise.
SIPX_NOISE_REDUCTION_LOW	Enable NR with least amount of aggressiveness.
SIPX_NOISE_REDUCTION_MEDIUM	Enable NR with modest amount of aggressiveness.
SIPX_NOISE_REDUCTION_HIGH	Enable NR with highest amount of aggressiveness.

enum SIPX_RESULT

SIPX_RESULT is an enumeration with all the possible result/return codes.

Enumerator
SIPX_RESULT_SUCCESS	Success.
SIPX_RESULT_FAILURE	Generic Failure.
SIPX_RESULT_NOT_IMPLEMENTED	Method/API not implemented.
SIPX_RESULT_OUT_OF_MEMORY	Unable to allocate enough memory to perform operation.
SIPX_RESULT_INVALID_ARGS	Invalid arguments; bad handle, argument out of range, etc.
SIPX_RESULT_BAD_ADDRESS	Invalid SIP address.
SIPX_RESULT_OUT_OF_RESOURCES	Out of resources (hit some max limit)
SIPX_RESULT_INSUFFICIENT_BUFFER	Buffer too short for this operation.
SIPX_RESULT_EVAL_TIMEOUT	The evaluation version of this product has expired.
SIPX_RESULT_BUSY	The operation failed because the system was busy.
SIPX_RESULT_INVALID_STATE	The operation failed because the object was in the wrong state. For example, attempting to split a call from a conference before that call is connected.
SIPX_RESULT_MISSING_RUNTIME_FILES	The operation failed because required runtime dependencies are missing.
SIPX_RESULT_TLS_DATABASE_FAILURE	The operation failed because the certificate database did not initialize.
SIPX_RESULT_TLS_BAD_PASSWORD	The operation failed because the certificate database did not accept the password.
SIPX_RESULT_TLS_TCP_IMPORT_FAILURE	The operation failed because a TCP socket could not be imported by the SSL/TLS module.
SIPX_RESULT_NSS_FAILURE	The operation failed due to an NSS failure.
SIPX_RESULT_NOT_SUPPORTED	The operation is not supported in this build/configuration.
SIPX_RESULT_NETWORK_FAILURE	The network is down or failing.
SIPX_RESULT_LIB_NOT_FOUND	Dependent shared library not found.
SIPX_RESULT_OS_PLATFORM_UNSUPPORTED	libraries or facilities not supported for this specific OS platform

enum SIPX_RTP_TRANSPORT

Possible roles that a Media connection can have.

TCP is currently defined, but not fully implemented. Use at your own risk.

Enumerator
SIPX_RTP_TRANSPORT_UNKNOWN
SIPX_RTP_TRANSPORT_UDP
SIPX_RTP_TRANSPORT_TCP
SIPX_RTP_TCP_ROLE_ACTIVE
SIPX_RTP_TCP_ROLE_PASSIVE
SIPX_RTP_TCP_ROLE_ACTPASS
SIPX_RTP_TCP_ROLE_CONNECTION

enum SIPX_SRTP_LEVEL

Enumeration of the possible levels of SRTP.

Enumerator
SRTP_LEVEL_NONE
SRTP_LEVEL_ENCRYPTION
SRTP_LEVEL_AUTHENTICATION
SRTP_LEVEL_ENCRYPTION_AND_AUTHENTICATION

enum SIPX_TONE_ID

DTMF/other tone ids used with sipxCallStartTone/sipxCallStopTone.

Enumerator
ID_DTMF_INVALID	Invalid/Uninitialized DMTF Id.
ID_DTMF_0	DMTF 0.
ID_DTMF_1	DMTF 1.
ID_DTMF_2	DMTF 2.
ID_DTMF_3	DMTF 3.
ID_DTMF_4	DMTF 4.
ID_DTMF_5	DMTF 5.
ID_DTMF_6	DMTF 6.
ID_DTMF_7	DMTF 7.
ID_DTMF_8	DMTF 8.
ID_DTMF_9	DMTF 9.
ID_DTMF_STAR	DMTF *.
ID_DTMF_POUND	DMTF #.
ID_DTMF_FLASH	DTMF Flash.
ID_TONE_DIALTONE	Dialtone (Not supported with GIPS VoiceEngine)
ID_TONE_BUSY	Call-busy tone (Not supported with GIPS VoiceEngine)
ID_TONE_RINGBACK	Remote party is ringing feedback tone (Not supported with GIPS VoiceEngine)
ID_TONE_RINGTONE	Default ring/alert tone (Not supported with GIPS VoiceEngine)
ID_TONE_CALLFAILED	Fasy Busy / call failed tone (Not supported with GIPS VoiceEngine)
ID_TONE_SILENCE	Silence (Not supported with GIPS VoiceEngine)
ID_TONE_BACKSPACE	Backspace tone (Not supported with GIPS VoiceEngine)
ID_TONE_CALLWAITING	Call waiting alert tone (Not supported with GIPS VoiceEngine)
ID_TONE_CALLHELD	Call held feedback tone (Not supported with GIPS VoiceEngine)
ID_TONE_LOUD_FAST_BUSY	Off hook / fast busy tone (Not supported with GIPS VoiceEngine)

enum SIPX_TRANSPORT_TYPE

SIPX_TRANSPORT_TYPE defines various protocols use for signaling transport.

The SIPX_TRANSPORT_TYPE is return in contact addresses.

Enumerator
TRANSPORT_UDP	Indicator for a UDP socket type.
TRANSPORT_TCP	Indicator for a TCP socket type.
TRANSPORT_TLS	Indicator for a TLS socket type.
TRANSPORT_CUSTOM	Indicator for a custom external transport.

enum SIPX_VIDEO_BANDWIDTH_ID

Video Codec bandwidth ids are used to select a group of codecs with equal or lower bandwidth requirements.

The codec name is a combination of the actual codec name and the video resolution.

Supported codecs are:

VP71, IYUV, I420, and RGB24

Supported resolutions are

CIF (352x288), QCIF (176x144), SQCIF (128x96), and QVGA (320x240)

A VP71 codec in QCIF resolution would be named VP71-QCIF.

Enumerator
VIDEO_CODEC_BW_VARIABLE	ID for codecs with variable bandwidth requirements.
VIDEO_CODEC_BW_LOW	ID for codecs with low bandwidth requirements.
VIDEO_CODEC_BW_NORMAL	ID for codecs with normal bandwidth requirements.
VIDEO_CODEC_BW_HIGH	ID for codecs with high bandwidth requirements.
VIDEO_CODEC_BW_CUSTOM	Possible return value for sipxConfigGetVideoCodecPreferences. This ID indicates the available list of codecs was overridden by a sipxConfigSetVideoCodecByName call.
VIDEO_CODEC_BW_DEFAULT	Value used to signify the default bandwidth level when calling sipxCallLimitCodecPreferences.

enum SIPX_VIDEO_DISPLAY_TYPE

Enum for specifying the type of display object to be used for displaying video.

Enumerator
SIPX_WINDOW_HANDLE_TYPE	A handle to the window for the remote video display.
DIRECT_SHOW_FILTER	A DirectShow render filter object for handling the remote video display.

enum SIPX_VIDEO_FORMAT

Enumeration of possible video sizes.

Enumerator
VIDEO_FORMAT_CIF	352x288
VIDEO_FORMAT_QCIF	176x144
VIDEO_FORMAT_SQCIF	128x96
VIDEO_FORMAT_QVGA	320x240

enum SIPX_VIDEO_QUALITY_ID

Video Codec quality definitions.

Quality is used as a trade off between between CPU usage and the amount of bandwidth used.

Enumerator
VIDEO_QUALITY_LOW	Low quality video.
VIDEO_QUALITY_NORMAL	Normal quality video.
VIDEO_QUALITY_HIGH	High quality video.

enum SPEAKER_TYPE

Speaker output types are used to differentiate between the logical ringer (used to alert user of in inbound call) and speaker (in call audio device).

Enumerator
SPEAKER	Speaker / in call device.
RINGER	Ringer / alerting device.

Function Documentation

SIPXTAPI_API SIPX_RESULT sipxAudioEnableSpeaker	(	const SIPX_INST	hInst,
		const SPEAKER_TYPE	type
	)

Enables one of the speaker outputs.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
type	The type of the speaker either the logical ringer (used to alert user of in inbound call) or speaker (in call audio device).

SIPXTAPI_API SIPX_RESULT sipxAudioGetAECMode	(	const SIPX_INST	hInst,
		SIPX_AEC_MODE &	mode
	)

Get the mode of Acoustic Echo Cancellation (AEC).

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
mode	AEC mode.

See also

SIPX_AEC_MODE

SIPXTAPI_API SIPX_RESULT sipxAudioGetAGCMode	(	const SIPX_INST	hInst,
		bool &	bEnabled
	)

Get the enable/disable state of Automatic Gain Control (AGC).

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bEnabled	true if AGC is enabled; otherwise false.

SIPXTAPI_API SIPX_RESULT sipxAudioGetCurrentInputDevice	(	const SIPX_INST	hInst,
		int &	index,
		const char *&	szDevice
	)

Get the name/identifier for the current input device used by sipX.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
index	Zero based index of the input device.
szDevice	Reference an character string pointer to receive the device name.

SIPXTAPI_API SIPX_RESULT sipxAudioGetCurrentOutputDevice	(	const SIPX_INST	hInst,
		int &	index,
		const char *&	szDevice
	)

Get the name/identifier for the current output device used by sipX.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
index	Zero based index of the input device.
szDevice	Reference an character string pointer to receive the device name.

SIPXTAPI_API SIPX_RESULT sipxAudioGetEnabledSpeaker	(	const SIPX_INST	hInst,
		SPEAKER_TYPE &	type
	)

Gets the enabled speaker selection.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
type	The type of the speaker either the logical ringer (used to alert user of in inbound call) or speaker (in call audio device).

SIPXTAPI_API SIPX_RESULT sipxAudioGetGain	(	const SIPX_INST	hInst,
		int &	iLevel
	)

Get the current microphone gain.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
iLevel	The level of the gain of the microphone

SIPXTAPI_API SIPX_RESULT sipxAudioGetInputDevice	(	const SIPX_INST	hInst,
		const int	index,
		const char *&	szDevice
	)

Get the name/identifier for input device at position index.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
index	Zero based index of the input device to be queried.
szDevice	Reference an character string pointer to receive the device name.

SIPXTAPI_API SIPX_RESULT sipxAudioGetNoiseReductionMode	(	const SIPX_INST	hInst,
		SIPX_NOISE_REDUCTION_MODE &	mode
	)

Get the mode/policy for Noise Reduction (NR).

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
mode	noise reduction mode.

See also

SIPX_NOISE_REDUCTION_MODE

SIPXTAPI_API SIPX_RESULT sipxAudioGetNumInputDevices	(	const SIPX_INST	hInst,
		size_t &	numDevices
	)

Get the number of input devices available on this system.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
numDevices	The number of input devices available on this system.

SIPXTAPI_API SIPX_RESULT sipxAudioGetNumOutputDevices	(	const SIPX_INST	hInst,
		size_t &	numDevices
	)

Get the number of output devices available on this system.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
numDevices	The number of output devices available on this system.

SIPXTAPI_API SIPX_RESULT sipxAudioGetOutputDevice	(	const SIPX_INST	hInst,
		const int	index,
		const char *&	szDevice
	)

Get the name/identifier for output device at position index.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
index	Zero based index of the output device to be queried.
szDevice	Reference an character string pointer to receive the device name.

SIPXTAPI_API SIPX_RESULT sipxAudioGetVolume	(	const SIPX_INST	hInst,
		const SPEAKER_TYPE	type,
		int &	iLevel
	)

Gets the audio level for the designated speaker type.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
type	The type of the speaker either the logical ringer (used to alert user of in inbound call) or speaker (in call audio device).
iLevel	The level of the gain of the microphone

SIPXTAPI_API SIPX_RESULT sipxAudioIsMuted	(	const SIPX_INST	hInst,
		bool &	bMuted
	)

Gets the mute state of the microphone.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bMuted	True if the microphone has been muted and false if it is not mute

SIPXTAPI_API SIPX_RESULT sipxAudioMute	(	const SIPX_INST	hInst,
		const bool	bMute
	)

Mute or unmute the microphone.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bMute	True if the microphone is to be muted and false if it is not to be muted

SIPXTAPI_API SIPX_RESULT sipxAudioSetAECMode	(	const SIPX_INST	hInst,
		const SIPX_AEC_MODE	mode
	)

Enables or disables Acoustic Echo Cancellation (AEC).

By default, sipXtapi assumes SIPX_AEC_CANCEL_AUTO. Change this parameter will modify the policy for both existing and new calls.

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
mode	AEC mode.

See also

SIPX_AEC_MODE

SIPXTAPI_API SIPX_RESULT sipxAudioSetAGCMode	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable/Disable Automatic Gain Control (AGC).

By default, AGC is disabled.

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bEnable	true to enable AGC or false to disable

SIPXTAPI_API SIPX_RESULT sipxAudioSetCallInputDevice	(	const SIPX_INST	hInst,
		const char *	szDevice
	)

Set the call input device (in-call microphone).

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szDevice	Character string pointer to be set to a string name of the output device.

SIPXTAPI_API SIPX_RESULT sipxAudioSetCallOutputDevice	(	const SIPX_INST	hInst,
		const char *	szDevice
	)

Set the call output device (in-call speaker).

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szDevice	The call output device.

SIPXTAPI_API SIPX_RESULT sipxAudioSetGain	(	const SIPX_INST	hInst,
		const int	iLevel
	)

Set the local microphone gain.

If the microphone is muted, resetting the gain will not enable audio – you must unmute the microphone.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
iLevel	The level of the local microphone gain

SIPXTAPI_API SIPX_RESULT sipxAudioSetNoiseReductionMode	(	const SIPX_INST	hInst,
		const SIPX_NOISE_REDUCTION_MODE	mode
	)

Set the noise reduction mode/policy for suppressing background noise.

By default sipXtapi assumes SIPX_NOISE_REDUCTION_LOW. Change this parameter will modify the policy for both existing and new calls.

Note: This API is only supported when bundled with VoiceEngine from Global IP Sound or Speex library.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
mode	noise reduction mode.

See also

SIPX_NOISE_REDUCTION_MODE

SIPXTAPI_API SIPX_RESULT sipxAudioSetRingerOutputDevice	(	const SIPX_INST	hInst,
		const char *	szDevice
	)

Set the call ringer/alerting device.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szDevice	The call ringer/alerting device.

SIPXTAPI_API SIPX_RESULT sipxAudioSetVolume	(	const SIPX_INST	hInst,
		const SPEAKER_TYPE	type,
		const int	iLevel
	)

Sets the audio level for the designated speaker type.

If the speaker type is enabled, the change it audio will be heard instantly.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
type	The type of the speaker either the logical ringer (used to alert user of in inbound call) or speaker (in call audio device).
iLevel	The level of the gain of the microphone 8 (see VOLUME_MIN, VOLUME_MAX and VOLUME_DEFAULT)

SIPXTAPI_API SIPX_RESULT sipxCallAccept	(	const SIPX_CALL	hCall,
		SIPX_VIDEO_DISPLAY *const	pDisplay = NULL,
		SIPX_SECURITY_ATTRIBUTES *const	pSecurity = NULL,
		SIPX_CALL_OPTIONS *	options = NULL,
		bool	bSendEarlyMedia = false
	)

Accepts an inbound call and proceed immediately to alerting.

This method is invoked in response to a NEWCALL event. Whenever a new call is received, the application developer should ACCEPT (proceed to ringing), REJECT (send back busy), or REDIRECT the call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
pDisplay	Pointer to an object describing the display object for rendering remote video.
pSecurity	Pointer to an object describing the security attributes for the call.
options	Pointer to a SIPX_CALL_OPTIONS structure.
bSendEarlyMedia	flag to send early media (RTP) upon accepting the call

See also

sipxConfigSetLocationHeader

sipxConfigSetAudioCodecPreferences

SIPXTAPI_API SIPX_RESULT sipxCallAnswer	(	const SIPX_CALL	hCall,
		bool	bTakeFocus = true
	)

Answer an alerting call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
bTakeFocus	Should SIPxua place the newly answered call in focus (engage local microphone and speaker). In some cases, application developer may want to answer the call in the background and play audio while the user finishes up with their active (in focus) call.

SIPXTAPI_API SIPX_RESULT sipxCallAudioPlayFileStart	(	const SIPX_CALL	hCall,
		const char *	szFile,
		const bool	bRepeat,
		const bool	bLocal,
		const bool	bRemote,
		const bool	bMixWithMicrophone = false,
		const float	fVolumeScaling = 1.0
	)

Play the designated file.

The file may be a raw 16 bit signed PCM at 8000 samples/sec, mono, little endian or a .WAV file. If a sipxCallDestroy is attempted while an audio file is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. Call sipxCallAudioPlayFileStop before making the call to sipxCallDestroy.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface. Audio files can only be played in the context of a call.
szFile	Filename for the audio file to be played.
bRepeat	True if the file is supposed to be played repeatedly
bLocal	True if the audio file is to be rendered locally.
bRemote	True if the audio file is to be rendered by the remote endpoint.
bMixWithMicrophone	True to mix the audio with the microphone data or false to replace it. This option is only supported when sipXtapi is bundled with GIPS VoiceEngine.
fVolumeScaling	Volume down scaling for the audio file. Valid values are between 0 and 1.0, where 1.0 is the no down-scaling. This option is only supported when sipXtapi is bundled with GIPS VoiceEngine.

SIPXTAPI_API SIPX_RESULT sipxCallAudioPlayFileStop

(

const SIPX_CALL

hCall

)

Stop playing a file started with sipxCallPlayFileStart If a sipxCallDestroy is attempted while an audio file is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code.

Call sipxCallAudioPlayFileStop before making the call to sipxCallDestroy.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface. Audio files can only be played and stopped in the context of a call.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordBufferStart	(	const SIPX_CALL	hCall,
		const char *	pBuffer,
		const int	bufferSize,
		const int	bufferType = RAW_PCM_16,
		const int	maxRecordTime = -1,
		const int	maxSilence = -1
	)

Record a call session (including other parties if this is a multi-party call / conference) to a buffer.

Conference join operation on this call will cause the recording to stop. In case of conference recording, this function should be called for single conference call only. Conference recording will continue even after the original call had been dropped/split, as long as there is at least one call left in the conference. If the original call has been dropped, use handle of other call in the conference to stop recording.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
pBuffer	Buffer to record data to.
bufferSize	Size of the buffer in bytes.
bufferType	The audio encoding format for the data as specified by the SIPX_AUDIO_DATA_FORMAT enumerations. Currently only RAW_PCM_16 is supported.
maxRecordTime	Maximum time to record in milliseconds. Recording stops automatically when this amount of time is recorded. Pass -1 to disable time limit.
maxSilence	Maximum silence time before recording is stopped automatically. Pass -1 to disable stop on silence.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordBufferStop

(

const SIPX_CALL

hCall

)

Stop recording a call to buffer.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordFileStart	(	const SIPX_CALL	hCall,
		const char *	szFile,
		const SIPX_AUDIO_FILE_FORMAT	recordFormat = SIPX_WAVE_PCM_16,
		const bool	appendToFile = false,
		const int	numChannels = 1
	)

Record a call session (including other parties if this is a multi-party call / conference) to a file.

The resulting file will be a PCM WAV file. Conference join operation on this call will cause the recording to stop. In case of conference recording, this function should be called for single conference call only. Conference recording will continue even after the original call had been dropped/split, as long as there is at least one call left in the conference. If the original call has been dropped, use handle of other call in the conference to stop recording.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szFile	Filename for the resulting audio file.
recordFormat	- audio file format to use when creating recording file
appendToFile	- true/false if file exists, append to it as opposed to replacing it. Note: existing file must be of the same audio format requested or start record will fail.
numChannels	- the number of channels to record in the wav file. Must be less than compile time maximum channels for recording.

Note: If recording with append in rapid stop, start situations, be sure to wait for the MEDIA_RECORDFILE_STOP media event before starting to append to the same record file or you risk having multiple file handles open on the same file with unpredictable results.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordFileStop

(

const SIPX_CALL

hCall

)

Stop recording a call to file.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordPause

(

const SIPX_CALL

hCall

)

Pause recording call to file or buffer.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallAudioRecordResume

(

const SIPX_CALL

hCall

)

Resume paused recording call to file or buffer.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallBlindTransfer	(	const SIPX_CALL	hCall,
		const char *	szAddress
	)

Blind transfer the specified call to another party.

Monitor the TRANSFER state events for details on the transfer attempt. If the call is not already on hold, the party will be placed on hold.

Assuming that all parties are using sipXtapi, all of the calls are active (not held), and the transfer works, you should expect the following event sequence:

Transferee (party being transfered):

The transferee's original call will be placed on remote hold from the transfer controller and will then put itself on full hold before starting the transfer. The transfer is implemented as a new call – this allows the developer to reclaim the original call if the transfer fails. The NEWCALL event will include a cause for of CALLSTATE_CAUSE_TRANSFER and the hAssociatedCall member of the SIPX_CALLSTATE_INFO structure will include the handle of the original call.

Original Call: MEDIA_LOCAL_STOP
Original Call: MEDIA_REMOTE_STOP
Original Call: CALLSTATE_REMOTE_HELD
Original Call: CALLSTATE_HELD

New Call: CALLSTATE_NEWCALL
New Call: CALLSTATE_REMOTE_OFFERING
New Call: CALLSTATE_REMOTE_ALERTING
New Call: CALLSTATE_CONNECTED
New Call: MEDIA_LOCAL_START
New Call: MEDIA_REMOTE_START

After the transfer completes, the application developer must destroy the original call using sipxCallDestroy. If the new call fails for any reason, the application layer should resume the original call by taking it off hold.

Transfer Controller (this user agent):

The transfer controller will automatically take the call out of focus and place it on hold. Afterwards, the transfer controller will receive CALLSTATE_TRANSFER_EVENTs indicating the status of the transfer.

Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_INITIATED
Source Call: CALLSTATE_BRIDGED
Source Call: CALLSTATE_HELD
Source Call: MEDIA_CAUSE_HOLD
Source Call: MEDIA_CAUSE_HOLD
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_ACCEPTED
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_RINGING
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_SUCCESS
Source Call: CALLSTATE_DISCONNECTED

Upon success, the call will automatically be disconnected, however, the application layer needs to call sipXcallDestroy to free the handle and call processing resources.

Transfer Target (identified by szAddress):

The transferee will go through the normal event progression for an incoming call:

New Call: CALLSTATE_NEWCALL
New Call: CALLSTATE_OFFERING
New Call: CALLSTATE_ALERTING
New Call: CALLSTATE_CONNECTED

If the transfer target rejects the call or fails to answer, the transfer will fail.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szAddress	SIP url identifing the transfer target (who the call identified by hCall will be transfered to).

See also

SIPX_CALLSTATE_EVENT

SIPX_CALLSTATE_CAUSE

SIPXTAPI_API SIPX_RESULT sipxCallConnect	(	const SIPX_CALL	hCall,
		const char *	szAddress,
		SIPX_CONTACT_ID	contactId = 0,
		SIPX_VIDEO_DISPLAY *const	pDisplay = NULL,
		SIPX_SECURITY_ATTRIBUTES *const	pSecurity = NULL,
		bool	bTakeFocus = true,
		SIPX_CALL_OPTIONS *	options = NULL,
		const char *	szCallId = NULL
	)

Connects an idle call to the designated target address.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szAddress	SIP url of the target party
contactId	Id of the desired contact record to use for this call. The id refers to a Contact Record obtained by a call to sipxConfigGetLocalContacts. The application can choose a contact record of type LOCAL, NAT_MAPPED, CONFIG, or RELAY. The Contact Type allows you to control whether the user agent and media processing advertises the local address (e.g. LOCAL contact of 10.1.1.x or 192.168.x.x), the NAT-derived address to the target party, or, local contact addresses of other types. The Contact Record's eTransportType field indicates the type of network transport to be used for call signalling. this can be UPD, TCP, or TLS. If the eTransportType field value is greater than 3, this indicates that a custom EXTERNAL TRANSPORT mechanism is to be used, and the value of the eTransportType field indicates the SIPX_TRANSPORT handle associated with the EXTERNAL TRANSPORT.
pDisplay	Pointer to an object describing the display object for rendering remote video.
pSecurity	Pointer to an object describing the security attributes for the call.
bTakeFocus	Should SIPxua place the this call in focus (engage local microphone and speaker). In some cases, application developer may want to place the call in the background and play audio while the user finishes up with their active (in focus) call.
options	Pointer to a SIPX_CALL_OPTIONS structure.
szCallId	A call-id for the session, if NULL, one is generated.

See also

sipxConfigSetLocationHeader

sipxConfigSetAudioCodecPreferences

SIPXTAPI_API SIPX_RESULT sipxCallCreate	(	const SIPX_INST	hInst,
		const SIPX_LINE	hLine,
		SIPX_CALL *	phCall
	)

Create a new call for the purpose of creating an outbound connection/call.

As a side effect, a DIALTONE event is fired to simulate the PSTN world. Generally an application would simulate dialtone in reaction to that event.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
hLine	Line Identity for the outbound call. The line identity helps defines the "From" caller-id. The line identity can't be SIPX_LIINE_NULL for an outbound call.
phCall	Pointer to a call handle. Upon success, this value is replaced with a valid call handle. Success is determined by the SIPX_RESULT result code.

SIPXTAPI_API SIPX_RESULT sipxCallDestroy

(

SIPX_CALL &

hCall

)

Drop/Destroy the specified call.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallGetAudioRtcpStats	(	const SIPX_CALL	hCall,
		SIPX_RTCP_STATS *	pStats
	)

Obtain RTCP stats for the specified call.

This API is only supported when sipXtapi is bundled with VoiceEngine from GIPS.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
pStats	Structure to place call stats, the structure's cbSize member must be filled out prior to calling this API. For example: myStats.cbSize = sizeof(SIPX_RTCP_STATS);

SIPXTAPI_API SIPX_RESULT sipxCallGetAudioRtpSourceIds	(	const SIPX_CALL	hCall,
		unsigned int &	iSendSSRC,
		unsigned int &	iReceiveSSRC
	)

Gets the sending and receiving Audio RTP SSRC IDs.

The SSRC ID is used to identify the RTP/audio stream. The call must be in the connected state for this request to succeed.

This API is only supported when sipXtapi is bundled with VoiceEngine from GIPS.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
iSendSSRC	The RTP SSRC used when sending audio
iReceiveSSRC	The RTP SSRC used by the remote party to sending audio

SIPXTAPI_API SIPX_RESULT sipxCallGetConference	(	const SIPX_CALL	hCall,
		SIPX_CONF &	hConf
	)

Get the conference handle for the specified call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
hConf	Conference handle for this call (if the call is part of a conference)

SIPXTAPI_API SIPX_RESULT sipxCallGetConnectionId	(	const SIPX_CALL	hCall,
		int &	connectionId
	)

Gets the media interface connection ID.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
connectionId	Reference to the returned connection identifier.

SIPXTAPI_API SIPX_RESULT sipxCallGetContactID	(	const SIPX_CALL	hCall,
		char *	szId,
		const size_t	iMaxLength
	)

Get the SIP identity of the contact connection.

The identity represents the originator of the message.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szId	Buffer to store the ID. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the ID buffer.

SIPXTAPI_API SIPX_RESULT sipxCallGetEnergyLevels	(	const SIPX_CALL	hCall,
		int &	iInputEnergyLevel,
		int &	iOutputEnergyLevel,
		const size_t	nMaxContributors,
		unsigned int	CCSRCs[],
		int	iEnergyLevels[],
		size_t &	nActualContributors
	)

Gets energy levels for a call.

The call must be in the connected state for this request to succeed.

This API is only supported when sipXtapi is bundled with VoiceEngine from GIPS.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
iInputEnergyLevel	Input/Microphone energy level ranging from 0 to 9.
iOutputEnergyLevel	Output/Speaker energy level ranging from 0 to 9. The output energy level is pre-mixed (before mixing any files/tones or other parties). sipxConferenceGetEnergyLevels provides an API to obtain post-mixed energy levels.
nMaxContributors	Max number of contributors/energy levels. Contributors are derived by looking at the contributing RTP source IDs from the RTP stream.
CCSRCs	Array of contributing source ids. This array will be filled in up to a max of nMaxContributors. See nActualContributors for the actual number of elements returned.
iEnergyLevels	Energy level for each contributing source id ranging from 0 to 9. This array in up to a max of nMaxContributors. See nActualContributors for the actual number of elements returned.
nActualContributors	The actual number of contributing source ids and energy levels returned.

SIPXTAPI_API SIPX_RESULT sipxCallGetID	(	const SIPX_CALL	hCall,
		char *	szId,
		const size_t	iMaxLength
	)

Get the SIP call ID of the call represented by the specified call handle.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szId	Buffer to store the ID. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the ID buffer

SIPXTAPI_API SIPX_RESULT sipxCallGetInviteHeader	(	const SIPX_CALL	hCall,
		const char *	headerName,
		char *	headerValue,
		const size_t	maxValueLength,
		bool *	inviteFromRemote,
		const size_t	headerInstanceIndex = 0
	)

Get the named header value from the last SIP INVITE request (sent or received) in the context of the given call handle.

Parameters

hCall	handle to a call.
headerName	- name of the header field for which the value is to be retrieved (case insensative)
headerValue	- SIP header field value retrieved from the INVITE if it exist and this is a SIP call
maxValueLength	- maximum number of characters including null terminator that may be set in headerValue
inviteFromRemote	- TRUE/FALSE the SIP INVITE was received from the remote end of the call (as opposed to sent by this end)
headerInstanceIndex	- some SIP header fields may have multiple instances. This index indicates which instance of the header field of the given name. The first instance has index = 0.

Returns

SIPX_RESULT_SUCCESS if the named header field was found SIPX_RESULT_FAILURE if the INVITE message exists, but the named header was not set in the INVITE SIPX_RESULT_INVALID_STATE if this is not a SIP call or the call has not yet been initiated by an INVITE

SIPXTAPI_API SIPX_RESULT sipxCallGetLine	(	const SIPX_CALL	hCall,
		SIPX_LINE &	hLine
	)

Get the line handle for the given call if it has one.

Parameters

hCall	- Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
hLine	- handle to call for local side of call. Note: not all calls have a line handle. Set to SIPX_LINE_NULL if line handle does not exist for call.

SIPXTAPI_API SIPX_RESULT sipxCallGetLocalID	(	const SIPX_CALL	hCall,
		char *	szId,
		const size_t	iMaxLength
	)

Get the SIP identity of the local connection.

The identity represents either 1) who was called in the case of a inbound call, or 2) the line identity used in an outbound call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szId	Buffer to store the ID. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the ID buffer.

SIPXTAPI_API SIPX_RESULT sipxCallGetRemoteContact	(	const SIPX_CALL	hCall,
		char *	szContact,
		const size_t	iMaxLength
	)

Get the SIP remote contact.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szContact	Buffer to store the remote contact. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the remote contact buffer.

SIPXTAPI_API SIPX_RESULT sipxCallGetRemoteID	(	const SIPX_CALL	hCall,
		char *	szId,
		const size_t	iMaxLength
	)

Get the SIP identity of the remote connection.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szId	Buffer to store the ID. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the ID buffer.

SIPXTAPI_API SIPX_RESULT sipxCallGetRemoteUserAgent	(	const SIPX_CALL	hCall,
		char *	szAgent,
		const size_t	iMaxLength
	)

Get the remote user agent of the call represented by the specified call handle.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szAgent	Buffer to store the user agent name. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the buffer

SIPXTAPI_API SIPX_RESULT sipxCallGetRequestURI	(	const SIPX_CALL	hCall,
		char *	szUri,
		const size_t	iMaxLength
	)

Get the SIP request URI.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szUri	Buffer to store the request URI. A zero-terminated string will be copied into this buffer on success.
iMaxLength	Max length of the request URI buffer.

SIPXTAPI_API SIPX_RESULT sipxCallHold	(	const SIPX_CALL	hCall,
		bool	bStopRemoteAudio = true
	)

Place the specified call on hold.

When placing calls on hold or having a remote party place you on hold the event sequences will differ. In this documentation, we refer to "local" hold and/or "focus" and "remote" and/or "full" hold. A call is on local hold when that call is taken out of focus and is no longer connected to the local microphone and speaker. Remote hold is used to indicate that RTP is no longer flowing between parties. The "bRemoteStopAudio" flags to this method controls whether the party is placed on local hold or full hold. See the table below for expected events:

                    RTP Flowing  RTP Stopped
                    ---------    -----------
        In Focus    CONNECTED    REMOTE_HELD
    Out of Focus     BRIDGED        HELD
*

CONNECTED indicates that both RTP is flowing and the call is attached to the local. This is the normal state for a connected call.

BRIDGED indicates that RTP is flowing, but the call is out of focus. This event is generally caused by holding a conference (conference will bridge by default) or if you accept/place a new call without explicitly holding the active call.

REMOTE_HELD indicates that RTP has stopped flowing. This is generally caused when the remote side places you on hold. The call is still locally in focus and audio will automatically resume once your are take off remote hold.

HELD indicates that both RTP has stopped flowing and the call is out of focus.

Developers can also expect media events (e.g. MEDIA_LOCAL_STOP) whenever RTP is stopped (REMOTE_HELD and HELD). Since media is still flowing for CONNECTED and BRIDGED, no media stop events are sent.

NOTE: If this call is part of a conference, sipxCallHold will only change the remote held state (RTP). You must use sipxConferenceHold to change call focus.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

bStopRemoteAudio

Flag which controls whether sipXtapi takes the call out of focus (stops engaging local audio microphone and speaker) or stops sending/receiving audio. Specify true to stop audio (default) or false to take the call out of focus. To play audio or generate tones to a remote connection while on hold, please specify false. This parameter is ignored (and assumed true) if the call is part of a conference.

SIPXTAPI_API SIPX_RESULT sipxCallLimitCodecPreferences	(	const SIPX_CALL	hCall,
		const SIPX_AUDIO_BANDWIDTH_ID	audioBandwidth,
		const SIPX_VIDEO_BANDWIDTH_ID	videoBandwidth,
		const char *	szVideoCodecName
	)

Limits the codec preferences on a per-call basis.

This API will force a codec renegotiation with the specified call regardless if the codecs changed. A renegotiation includes sending a new INVITE with an updated SDP list. Local audio will be stopped and restarted during this process, however, hold events are not sent to the application.

NOTE: If a call is on remote hold, it will be taken off remote hold.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
audioBandwidth	A bandwidth id to limit audio codecs. Pass in AUDIO_CODEC_BW_DEFAULT to leave audio codecs unchanged.
videoBandwidth	A bandwidth id to limit video bitrate and framerate. (see sipxConfigSetVideoBandwidth for an explanation on how bandwidth ids affect bitrate and framerate). Pass in AUDIO_CODEC_BW_DEFAULT to leave these parameters unchanged.
szVideoCodecName	Codec name that limits the supported video codecs to this one video codec.

See also

sipxConfigSetVideoBandwidth

SIPXTAPI_API SIPX_RESULT sipxCallLimitCodecs	(	const SIPX_CALL	hCall,
		const char *	codecNames
	)

Restrict the set of codecs that are allowed to be used for the given call.

This method does not force signalling to occur. Codecs must be restricted before the offer or answer SDP is sent from this side.

Parameters

hCall	- call handle for call on which to restrict codec set
codecNames	- space delimited list of codec names. Note: codecNames MUST be a subset of the codecs enabled via sipxConfigSetAudioCodecByName TELEPHONE-EVENT is implicitly included

SIPXTAPI_API SIPX_RESULT sipxCallPlayBufferStart	(	const SIPX_CALL	hCall,
		const char *	szBuffer,
		const int	bufSize,
		const int	bufType,
		const bool	bRepeat,
		const bool	bLocal,
		const bool	bRemote
	)

Play the specified audio data.

Currently the only data format that is supported is raw 16 bit signed PCM at 8000 samples/sec, mono, little endian. If a sipxCallDestroy is attempted while an audio buffer is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. Call sipxCallPlayBufferStop before making the call to sipxCallDestroy.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface. Audio can only be played in the context of a call.
szBuffer	Pointer to the audio data to be played.
bufSize	Length, in bytes, of the audio data.
bufType	The audio encoding format for the data as specified by the SIPX_AUDIO_DATA_FORMAT enumerations. Currently only RAW_PCM_16 is supported.
bRepeat	True if the audio is supposed to be played repeatedly
bLocal	True if the audio is to be rendered locally.
bRemote	True if the audio is to be rendered by the remote endpoint.

SIPXTAPI_API SIPX_RESULT sipxCallPlayBufferStop

(

const SIPX_CALL

hCall

)

Stop playing the audio started with sipxCallPlayBufferStart If a sipxCallDestroy is attempted while an audio buffer is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code.

Call sipxCallPlayBufferStop before making the call to sipxCallDestroy.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface. Audio can only be played and stopped in the context of a call.

SIPXTAPI_API SIPX_RESULT sipxCallRedirect	(	const SIPX_CALL	hCall,
		const char *	szForwardURL
	)

Redirect an inbound call (prior to alerting the user).

This method must be invoked before the end user is alerted (before sipxCallAccept). Whenever a new call is received, the application developer should ACCEPT (proceed to ringing), REJECT (send back busy), or REDIRECT the call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szForwardURL	SIP url to forward/redirect the call to.

SIPXTAPI_API SIPX_RESULT sipxCallReject	(	const SIPX_CALL	hCall,
		const int	errorCode = 400,
		const char *	szErrorText = "Bad Request"
	)

Reject an inbound call (prior to alerting the user).

This method must be invoked before the end user is alerted (before sipxCallAccept). Whenever a new call is received, the application developer should ACCEPT (proceed to ringing), REJECT (send back busy), or REDIRECT the call.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
errorCode	RFC specified error code.
szErrorText	null terminated text string to explain the error code.

SIPXTAPI_API SIPX_RESULT sipxCallResizeWindow	(	const SIPX_CALL	hCall,
		const SIPX_WINDOW_HANDLE	hWnd
	)

Resizes the video window.

Should be called when the window receives a SIZE message.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
hWnd	Window handle of the video window.

SIPXTAPI_API SIPX_RESULT sipxCallSendInfo	(	SIPX_INFO *	phInfo,
		const SIPX_CALL	hCall,
		const char *	szContentType,
		const char *	szContent,
		const size_t	nContentLength
	)

Sends an INFO event to the specified call.

This method will fail with an SIPX_RESULT_INVALID_STATE return code if an existing INFO message transaction is still outstanding (sipXtapi has not received a final response to the initial request).

Parameters

phInfo	Pointer to an INFO message handle, whose value is set by this method.
hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
szContentType	String representation of the INFO content type
szContent	Pointer to the INFO messasge's content
nContentLength	Size of the INFO content

SIPXTAPI_API SIPX_RESULT sipxCallSetMediaPassThrough	(	const SIPX_CALL	hCall,
		MEDIA_TYPE	mediaType,
		int	mediaTypeStreamIndex,
		const char *	streamReceiveAddress,
		int	rtpPort,
		int	rtcpPort
	)

Set the address and port to which the given media stream is to be recieved.

This is generally used when a separate process is to recieve the media stream.

Parameters

hCall	- call handle for the RTP stream
mediaType	- audio or video
mediaTypeStreamIndex	- connection Id
streamReceiveAddress	- IP address on which RTP stream is to be received
rtpPort	- RTP port on which to receive stream
rtcpPort	- RTCP port on which to recieve stream

SIPXTAPI_API SIPX_RESULT sipxCallSetMicGain	(	const SIPX_CALL	hCall,
		float	gain
	)

Sets the gain of the microphone for a given call.

This gain is applied digitally inside of the media engine.

Note

Right now this is implemented for sipXmediaLib with Topology enabled only.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
gain	Floating point gain to apply. Gain should be >0. Internally, inside of a media engine, it may be represented as fixed point value. (values for gain recommended in the 0.001-10.0 range) The default value is 1.0. Clipping may occur for values greater than 1.0. Also

See also

sipxCallHold with the bStopRemoteAudio=false may be useful.

SIPXTAPI_API SIPX_RESULT sipxCallSetMixOutputGain	(	const SIPX_CALL	hCall,
		int	bridgeOutputIndex,
		float	gain
	)

Sets the gain for all inputs to the given bridge output, except for the input of the same index which is set to 0.0f gain.

Parameters

hCall	- handle to call using the bridge mixer.
bridgeOutputIndex	- output index on the bridge whose input gains are to be set.
gain	- new gain value to set for inputs. (values for gain recommended in the 0.001-10.0 range) The default value is 1.0. Clipping may occur for values greater than 1.0

SIPXTAPI_API SIPX_RESULT sipxCallStartTone	(	const SIPX_CALL	hCall,
		const SIPX_TONE_ID	toneId,
		const bool	bLocal,
		const bool	bRemote
	)

Play a tone (DTMF, dialtone, ring back, etc) to the local and/or remote party.

See the DTMF_ constants for built-in tones. If a sipxCallDestroy is attempted while a tone is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. Call sipxCallStopTone before making the call to sipxConferenceDestroy. DTMF is sent via RFC 2833 method.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
toneId	ID of the tone to play
bLocal	Should the tone be played locally?
bRemote	Should the tone be played to the remote party?

SIPXTAPI_API SIPX_RESULT sipxCallStopTone

(

const SIPX_CALL

hCall

)

Stop playing a tone (DTMF, dialtone, ring back, etc).

to local and remote parties. Under a GIPS VoiceEngine build, this method is a NOP – VoiceEngine only plays tones for a specific time interval. If a sipxCallDestroy is attempted while a tone is playing, sipxCallDestroy will fail with a SIPX_RESULT_BUSY return code. Call sipxCallStopTone before making the call to sipxConferenceDestroy.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallSubscribe	(	const SIPX_CALL	hCall,
		const char *	szEventType,
		const char *	szAcceptType,
		SIPX_SUB *	phSub,
		bool	bRemoteContactIsGruu = false
	)

Subscribe for NOTIFY events which may be published by the other end-point of the call.

This API differs from sipxConfigSubscribe in that it allows you to use the contact address from the remote party as the subscription target (see the bRemoteContactIsGruu parameter).

sipXtapi will automatically refresh subscriptions until sipxCallUnsubscribe is called. Please make sure you call sipxCallUnsubscribe before tearing down your call.

Parameters

hCall	The call handle of the call associated with the subscription.
szEventType	A string representing the type of event that can be published. This string is used to populate the "Event" header in the SIP SUBSCRIBE request. For example, if checking voicemail status, your would use "message-summary".
szAcceptType	A string representing the types of NOTIFY events that this client will accept. This string is used to populate the "Accept" header in the SIP SUBSCRIBE request. For example, if checking voicemail status, you would use "application/simple-message-summary"
phSub	Pointer to a subscription handle whose value is set by this function. This handle allows you to cancel the subscription and differentiate between NOTIFY events.
bRemoteContactIsGruu	indicates whether the Contact for the remote side of the call can be assumed to be a Globally Routable Unique URI (GRUU). Normally one cannot assume that a contact is a GRUU and the To or From address for the remote side is assumed to be an Address Of Record (AOR) that is globally routable.

SIPXTAPI_API SIPX_RESULT sipxCallTransfer	(	const SIPX_CALL	hSourceCall,
		const SIPX_CALL	hTargetCall
	)

Transfer the source call to the target call.

This method can be used to implement consultative transfer (transfer initiator can speak with the transfer target prior to transferring). If you wish to consult privately, create a new call to the transfer target. If you wish consult and allow the source (transferee) to participant in the conversation, create a conference and then transfer one leg to another.

If not already on hold, the transferee (hSourceCall) is placed on hold as part of the transfer operation.

The event sequence may differ slightly depending on whether the calls are part of a conference (attended transfer) or individual calls (semi- attended transfer).

Assuming the calls are part of a conference and not on hold, the event sequences are as follows:

Transfer Controller (this user agent):

Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_INITIATED
Source Call: CALLSTATE_REMOTE_HELD
Source Call: MEDIA_LOCAL_STOP
Source Call: MEDIA_REMOTE_STOP
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_ACCEPTED
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_CAUSE_TRANSFER_SUCCESS
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_DISCONNECTED
Source Call: CALLSTATE_TRANSFER_EVENT::CALLSTATE_DESTROYED

The source call will automatically be disconnected if the transfer is successful. Also, if the source call is part of a conference, the call will automatically be destroyed. If not part of a conference, the application must destroy the call using sipxCallDestroy.

Target Call: CALLSTATE_REMOTE_HELD
Target Call: MEDIA_LOCAL_STOP
Target Call: MEDIA_REMOTE_STOP
Target Call: CALLSTATE_DISCONNECTED
Target Call: CALLSTATE_DESTROYED

The target call is remote held as part of the transfer operation. If the target call is part of a conference, it will automatically be destroyed. Otherwise, the application layer is responsible for calling sipxCallDestroy.

Transferee (user agent on other side of hSourceCall):

The transferee will create a new call to the transfer target and automatically disconnect the original call upon success. The new call will be created with a cause of CALLSTATE_CAUSE_TRANSFER in the SIPX_CALLSTATE_INFO event data. The application layer can look at the hAssociatedCall member to connect the new call to the original call.

Original Call: MEDIA_LOCAL_STOP
Original Call: MEDIA_REMOTE_STOP
Original Call: CALLSTATE_REMOTE_HELD
Original Call: CALLSTATE_HELD
Original Call: CALLSTATE_DISCONNECTED

New Call: CALLSTATE_NEWCALL::CALLSTATE_CAUSE_TRANSFER
New Call: CALLSTATE_REMOTE_OFFERING
New Call: CALLSTATE_CONNECTED
New Call: MEDIA_LOCAL_START
New Call: MEDIA_REMOTE_START

The application is responsible for calling sipxCallDestroy on the original call after the CALLSTATE_DISCONNECT event.

Transfer Target (user agent on other side of hTargetCall):

The transfer target will automatically receive and answer the inbound call from the transferee. After this completes, the original call is disconnected.

CALLSTATE_NEWCALL::CALLSTATE_CAUSE_TRANSFERRED
CALLSTATE_CONNECTED
MEDIA_LOCAL_START
MEDIA_REMOTE_START

Please note that no offering event was fired. The target does not have the option to accept or reject the call. If this call was part of a conference, the new call is automatically added to the same conference.

Parameters

hSourceCall	Handle to the source call (transferee).
hTargetCall	Handle to the target call (transfer target).

See also

SIPX_CALLSTATE_EVENT

SIPX_CALLSTATE_CAUSE

SIPXTAPI_API SIPX_RESULT sipxCallUnhold

(

const SIPX_CALL

hCall

)

Take the specified call off hold.

This API will take the call off both local hold and remote/full hold.

See also

sipxCallHold for a description of expected events associated with hold events.

Parameters

hCall

Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.

SIPXTAPI_API SIPX_RESULT sipxCallUnsubscribe

(

const SIPX_SUB

hSub

)

Unsubscribe from previously subscribed NOTIFY events.

This method will send another subscription request with an expires time of 0 (zero) to end your subscription.

Parameters

hSub	The subscription handle obtained from the call to sipxCallSubscribe.

SIPXTAPI_API SIPX_RESULT sipxCallUpdateVideoWindow	(	const SIPX_CALL	hCall,
		const SIPX_WINDOW_HANDLE	hWnd
	)

Updates the Video window with a new frame buffer.

Should be called when the window receives a PAINT message.

Parameters

hCall	Handle to a call. Call handles are obtained either by invoking sipxCallCreate or passed to your application through a listener interface.
hWnd	Window handle of the video preview window.

SIPXTAPI_API SIPX_RESULT sipxConferenceAdd	(	const SIPX_CONF	hConf,
		const SIPX_LINE	hLine,
		const char *	szAddress,
		SIPX_CALL *	phNewCall,
		SIPX_CONTACT_ID	contactId = 0,
		SIPX_VIDEO_DISPLAY *const	pDisplay = NULL,
		SIPX_SECURITY_ATTRIBUTES *const	pSecurity = NULL,
		bool	bTakeFocus = true,
		SIPX_CALL_OPTIONS *	options = NULL
	)

Add a new party to an existing conference.

A connection is automatically initiated for the specified address.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
hLine	Line Identity for the outbound call. The line identity helps defines the "From" caller-id. The line identity can't be SIPX_LIINE_NULL, as it's an outbound call.
szAddress	SIP URL of the conference participant to add
phNewCall	Pointer to a call handle to store new call.
contactId	Id of the desired contact record to use for this call. The id refers to a Contact Record obtained by a call to sipxConfigGetLocalContacts. The application can choose a contact record of type LOCAL, NAT_MAPPED, CONFIG, or RELAY. The Contact Type allows you to control whether the user agent and media processing advertises the local address (e.g. LOCAL contact of 10.1.1.x or 192.168.x.x), the NAT-derived address to the target party, or, local contact addresses of other types.
pDisplay	Pointer to an object describing the display object for rendering remote video.
pSecurity	Pointer to an object describing the security attributes for the call.
bTakeFocus	Should SIPxua place the newly answered call in focus (engage local microphone and speaker). In some cases, application developer may want to answer the call in the background and play audio while the user finishes up with their active (in focus) call.
options	Pointer to a SIPX_CALL_OPTIONS structure.

See also

sipxConferenceCreate

sipxConfigSetLocationHeader

sipxConfigGetLocalContacts

sipxConfigSetAudioCodecPreferences

SIPXTAPI_API SIPX_RESULT sipxConferenceCreate	(	const SIPX_INST	hInst,
		SIPX_CONF *	phConference
	)

Create a conference handle.

Conferences are an association of calls where the audio media is mixed. sipXtapi supports conferences up to 4 (CONF_MAX_CONNECTIONS) parties in its default configuration.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
phConference	Pointer to a conference handle. Upon success, this value is replaced with a valid conference handle. Success is determined by the SIPX_RESULT result code.

SIPXTAPI_API SIPX_RESULT sipxConferenceDestroy

(

SIPX_CONF

hConf

)

Destroys a conference.

All participants within a conference are dropped.

Parameters

hConf

Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.

SIPXTAPI_API SIPX_RESULT sipxConferenceGetCalls	(	const SIPX_CONF	hConf,
		SIPX_CALL	calls[],
		const size_t	iMax,
		size_t &	nActual
	)

Gets all of the calls participating in a conference.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
calls	An array of call handles filled in by the API.
iMax	The maximum number of call handles to return.
nActual	The actual number of call handles returned.

SIPXTAPI_API SIPX_RESULT sipxConferenceGetEnergyLevels	(	const SIPX_CONF	hConf,
		int &	iInputEnergyLevel,
		int &	iOutputEnergyLevel
	)

Gets energy levels for a conference.

The conference must be in the connected state (not held or bridged) for this request to succeed.

This API is only supported when sipXtapi is bundled with VoiceEngine from GIPS.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
iInputEnergyLevel	Input/Microphone energy level ranging from 0 to 9.
iOutputEnergyLevel	Output/Speaker energy level ranging from 0 to 9. The output energy level is post-mixed (after mixing all files/tones and other parties. sipxCallGetEnergyLevel provides an API to obtain pre-mixed energy levels.

SIPXTAPI_API SIPX_RESULT sipxConferenceHold	(	const SIPX_CONF	hConf,
		bool	bBridging = true
	)

Places a conference on hold.

This API can be used to place a conference on local hold (continue to bridge participants) or full hold (remaining participants cannot talk to each other). The default is local hold/bridged. The bBridged flag can be used to change this behavior (false for full hold).

Developers may also hold/unhold individual conference participants by calling sipxCallHold and sipxCallUnhold on individual call handles. The sipxConferenceGetCalls API can be used to enumerate conference participants.

See also

sipxCallHold for a description of the expected events.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
bBridging	true for a bridging conference hold, false for a non-bridging conference hold.

SIPXTAPI_API SIPX_RESULT sipxConferenceJoin	(	const SIPX_CONF	hConf,
		const SIPX_CALL	hCall
	)

Join (add) an existing held call into a conference.

An existing call can be added to a virgin conference without restriction. Additional calls, must be connected and on remote hold for this operation to succeed. A remote hold can be accomplished by calling sipxCallHold on the joining party. The application layer must wait for the CALLSTATE_HELD event prior to calling join. No events are fired as part of the operation and the newly joined call is left on hold. The application layer should call sipxCallUnhold on the new participant to finalize the join.

Parameters

hConf	Conference handle obtained by calling sipxConferenceCreate.
hCall	Call handle of the call to join into the conference.

SIPXTAPI_API SIPX_RESULT sipxConferenceLimitCodecPreferences	(	const SIPX_CONF	hConf,
		const SIPX_AUDIO_BANDWIDTH_ID	audioBandwidth,
		const SIPX_VIDEO_BANDWIDTH_ID	videoBandwidth,
		const char *	szVideoCodecName
	)

Limits the codec preferences on a conference.

This API will force a codec renegotiation with the specified calls regardless if the codecs changed. A renegotiation includes sending a new INVITE with an updated SDP list. Local audio will be stopped and restarted during this process, however, hold events are not sent to the application.

NOTE: If any calls are on remote hold, they will be taken off hold.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
audioBandwidth	A bandwidth id to limit audio codecs. Pass in AUDIO_CODEC_BW_DEFAULT to leave audio codecs unchanged.
videoBandwidth	A bandwidth id to limit video bitrate and framerate. (see sipxConfigSetVideoBandwidth for an explanation on how bandwidth ids affect bitrate and framerate). Pass in AUDIO_CODEC_BW_DEFAULT to leave these parameters unchanged.
szVideoCodecName	Codec name that limits the supported video codecs to this one video codec.

See also

sipxConfigSetVideoBandwidth

SIPXTAPI_API SIPX_RESULT sipxConferencePlayAudioFileStart	(	const SIPX_CONF	hConf,
		const char *	szFile,
		const bool	bRepeat,
		const bool	bLocal,
		const bool	bRemote,
		const bool	bMixWithMicrophone = false,
		const float	fVolumeScaling = 1.0
	)

Play the designated audio file to all conference participants and/or the local speaker.

The file may be a raw 16 bit signed PCM at 8000 samples/sec, mono, little endian or a .WAV file. If a sipxConferenceDestroy is attempted while an audio file is playing, sipxConferenceDestroy will fail with a SIPX_RESULT_BUSY return code. Call sipxConferencePlayAudioFileStop before making the call to sipxConferenceDestroy.

Parameters

hConf	Conference handle obtained by calling sipxConferenceCreate.
szFile	Filename for the audio file to be played.
bRepeat	True if the file is supposed to be played repeatedly
bLocal	True if the audio file is to be rendered locally.
bRemote	True if the audio file is to be rendered by the remote endpoint.
bMixWithMicrophone	True to mix the audio with the microphone data or false to replace it. This option is only supported when sipXtapi is bundled with GIPS VoiceEngine.
fVolumeScaling	Volume down scaling for the audio file. Valid values are between 0 and 1.0, where 1.0 is the no down-scaling. This option is only supported when sipXtapi is bundled with GIPS VoiceEngine.

SIPXTAPI_API SIPX_RESULT sipxConferencePlayAudioFileStop

(

const SIPX_CONF

hConf

)

SIPXTAPI_API SIPX_RESULT sipxConferenceRemove	(	const SIPX_CONF	hConf,
		const SIPX_CALL	hCall
	)

Removes a participant from conference by hanging up on them.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
hCall	Call handle identifying which call to remove from the conference by hanging up.

SIPXTAPI_API SIPX_RESULT sipxConferenceSplit	(	const SIPX_CONF	hConf,
		const SIPX_CALL	hCall
	)

Split (remove) a held call from a conference.

This method will remove the specified call from the conference.

The call must be connected and on remote hold for this operation to succeed. A remote hold can be accomplished by calling sipxCallHold on the conference participant or by placing the entire conference on hold with bridging disabled. The application layer must wait for the CALLSTATE_HELD event prior to calling split. No events are fired as part of the operation and the split call is left on hold.

Parameters

hConf	Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.
hCall	Call handle of the call that should be removed from the the conference.

SIPXTAPI_API SIPX_RESULT sipxConferenceUnhold

(

const SIPX_CONF

hConf

)

Removes conference members from a held state.

This method will take a call off either local or remote/full hold.

See also

sipxConferenceHold for details on holding conferences.

sipxCallHold for a description of the expected events.

Parameters

hConf

Handle to a conference. Conference handles are obtained by invoking sipxConferenceCreate.

SIPXTAPI_API SIPX_RESULT sipxConfigAddContact	(	const SIPX_INST	hInst,
		const char *	szSipContactAddress,
		const int	iSipContactPort,
		SIPX_CONTACT_ID &	iNewContactId
	)

Add a contact using the given IP address and port.

This can be used for IP address spoofing in the SIP Contact and SDP/RTP c address.

Parameters

szSipContactAddress	- IP address to use in SIP Contact in initial REGISTER and INVITE requests as well as the c field in the SDP.
iSipContactPort	- port to use in the SIP Contact headers.
iNewContactId	- contactId to use to specify this contact should be used (see , , ).

SIPXTAPI_API SIPX_RESULT sipxConfigDisableIce

(

const SIPX_INST

hInst

)

Disable the use of ICE.

See sipxConfigEnableICE for details.

Parameters

hInst

Instance pointer obtained by sipxInitialize.

SIPXTAPI_API SIPX_RESULT sipxConfigDisableStun

(

const SIPX_INST

hInst

)

Disable the use of STUN.

See sipxConfigEnableStun for details on STUN.

Parameters

hInst

Instance pointer obtained by sipxInitialize.

SIPXTAPI_API SIPX_RESULT sipxConfigDisableTurn

(

const SIPX_INST

hInst

)

Disable the use of TURN.

See sipxConfigEnableTurn for details TURN.

Parameters

hInst

Instance pointer obtained by sipxInitialize.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableDnsSrv

(

const bool

bEnable

)

Enables/disables sending of DNS SRV request for all sipXtapi instances.

DNS SRV is enabled by default.

Parameters

bEnable

Enable or disable DNS SRV resolution.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableGIPSTracing	(	SIPX_INST	hInst,
		bool	bEnable
	)

Enables GIPS tracing in sipXtapi (if bundled with sipXtapi).

Log entries are under the facility "VIDEOENGINE" and "VOICEENGINE" and are Base64 encoded (to avoid issues with binary data.

NOTE: Tracing must be enabled prior to setting up a call.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bEnable	true to enable GIPS tracing or false to disable.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableIce

(

const SIPX_INST

hInst

)

Enables an ICE-like mechanism for determining connectivity of remote parties dynamically.

By default, ICE is disabled.

The current sipXtapi implementation is a bastardization of draft-ietf-mmusic-ice-04. In subsequent release, this will conform to draft-ietf-mmusic-ice-05 or the latest draft.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableInBandDTMF	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable/disable sending of in-band DTMF tones.

In-band DTMF is disabled by default.

This function is currently not implemented.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	Enable or disable in-band DTMF tones.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableOutOfBandDTMF	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable/disable sending of RFC 2833 DTMF tones.

If disabled the tones will be sent in-band (if in-band DTMF is enabled). RFC 2833 DTMF is enabled by default.

Generally, out-of-band DTMF should always be enabled. In-band DTMF can be distorted and unrecognized by gateways/IVRs/ACDs when using compressed codecs such as G729. By nature, many compressed codecs are lossy and cannot regenerate DTMF signals. If you find that you need to disable out-of-band DTMF (due to duplicate DTMF signals) on another device, please consider reconfiguring that other device.

This function is currently not implemented.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	Enable or disable out-of-band DTMF tones.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableRport	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable or disable the use of "rport".

If rport is included on a via, responses should be sent back to the originating port – not what is advertised as part of via. Additionally, the sip stack will not receive messages sent to the originating port unless this is enabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
bEnable	Enable or disable the use of rport.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableRportMapping	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable or diable the use of IP and port provided in the response via (rport) for mapping of NAT/public IP address and port in the contact database.

By default rport mapping is enabled.

Parameters

bEnable

- true/false if IP address and port provided via rport should be used for contact database NAT address mapping.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableRTCP	(	const SIPX_INST	hInst,
		const bool	bEnable
	)

Enable or disable sending RTCP reports.

By default, RTCP is enabled and sends reports every ~5 seconds. RTCP is enabled by default.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	Enable or disable sending of RTCP reports.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableRtpOverTcp	(	const SIPX_INST	hInst,
		bool	bEnable
	)

Enables RTP streaming over TCP.

Enabling this feature allows the application to use RTP streaming over TCP or UDP.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	True allows RTP-over-TCP, false disallows it.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableSipAllowHeader	(	const SIPX_INST	hInst,
		const bool	bEnabled
	)

Enables/Disables use of allow header in sip messages.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnabled	True if allow header, false if no allow header

SIPXTAPI_API SIPX_RESULT sipxConfigEnableSipDateHeader	(	const SIPX_INST	hInst,
		const bool	bEnabled
	)

Enables/Disables use of date header in sip messages.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnabled	True if date header, false if no date header

SIPXTAPI_API SIPX_RESULT sipxConfigEnableSipShortNames	(	const SIPX_INST	hInst,
		const bool	bEnabled
	)

Enables/Disables use of short field names in sip messages.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnabled	True if short names, false if long names

SIPXTAPI_API SIPX_RESULT sipxConfigEnableStun	(	const SIPX_INST	hInst,
		const char *	szServer,
		int	iServerPort,
		int	iKeepAliveInSecs
	)

Enables STUN (Simple Traversal of UDP through NAT) support for both UDP SIP signaling and UDP audio/video (RTP).

STUN helps user agents determine thier external IP address from the inside of NAT/Firewall. This method should be invoked immediately after calling sipxInitialize and before creating any lines or calls. Enabling STUN while calls are setup should not effect the media path of existing calls. The "contact" address uses for UDP signaling may change on the next request.

TODO :: STUN conforms to IETF RFC/DRAFT XXXX with the following exceptions:

Parameters

hInst	Instance pointer obtained by sipxInitialize
szServer	The stun server that should be used for discovery.
iServerPort	The port of the stun server that should be used for discovery.
iKeepAliveInSecs	This setting controls how often to refresh the stun binding. The most aggressive NAT/Firewall solutions free port mappings after 30 seconds of non-use. We recommend a value of 28 seconds to be safe.

SIPXTAPI_API SIPX_RESULT sipxConfigEnableTurn	(	const SIPX_INST	hInst,
		const char *	szServer,
		const int	iServerPort,
		const char *	szUsername,
		const char *	szPassword,
		const int	iKeepAliveInSecs
	)

Enable TURN for support for UDP audio/video (RTP).

TURN allows VoIP communications while operating behind a symmetric NAT or firewall (cannot only receive data from the same IP/port that you have sent to).

This implementation is based on draft-rosenberg-midcom-turn-04.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szServer	The TURN server that should be used for relaying.
iServerPort	The TURN sever port that should be used for relaying
szUsername	TURN username for authentication
szPassword	TURN password for authentication
iKeepAliveInSecs	This setting controls how often to refresh the TURN binding. The recommended value is 28 seconds.

SIPXTAPI_API SIPX_RESULT sipxConfigExternalTransportAdd	(	SIPX_INST const	hInst,
		SIPX_TRANSPORT &	hTransport,
		const bool	bIsReliable,
		const char *	szTransport,
		const char *	szLocalIP,
		const int	iLocalPort,
		SIPX_TRANSPORT_WRITE_PROC	writeProc,
		const char *	szLocalRoutingId,
		const void *	pUserData = NULL
	)

Associates an external transport mechanism for SIP signaling with the given instance.

Parameters

hInst	An instance handle obtained from sipxInitialize.
hTransport	Reference to a SIPX_TRANSPORT handle. This function will assign a value to the referenced handle.
bIsReliable	If false, indicates that SIPxua should retry external transport writes in case of response timeouts. If true, SIPxua will not attempt retries in case of a response timeouts. For connection oriented transport (like TCP), bIsReliable should be true, for packet oriented transport (like UDP), bIsReliable should be false.
szTransport	Transport type string to be used in SIP URIs. For example, passing in a szTransport of "flibble" will cause the transport tag to be added to the URI like so: "sip:mickey\@example.com;transport=flibble"
szLocalIP	IP address which is the source address for the write.
iLocalPort	Port value from which the data will be sent.
writeProc	Function pointer to the callback for writing data using the transport mechanism.
szLocalRoutingId	A local routing id pass back to the write proc callback.
pUserData	User data passed back to the write proc.

SIPXTAPI_API SIPX_RESULT sipxConfigExternalTransportHandleMessage	(	const SIPX_TRANSPORT	hTransport,
		const char *	szSourceIP,
		const int	iSourcePort,
		const char *	szLocalIP,
		const int	iLocalPort,
		const void *	pData,
		const size_t	nData
	)

Called by the application when it receives a complete SIP message via it's external transport mechanism and want to pass it along to sipXtapi.

The application is responsible for preparing the Data so that it is composed of a single and complete SIP message.

Parameters

hTransport	Handle to the external transport object. Obtained via a call to sipxConfigExternalTransportAdd
szSourceIP	IP address which was the source of the data which was read.
iSourcePort	Port value from which the data was sent.
szLocalIP	Local IP address on which the data was received.
iLocalPort	Local port value on which the data was received.
pData	Pointer to the data which was received, which should point to a single and complete SIP message.
nData	Size of the data which was received.

SIPXTAPI_API SIPX_RESULT sipxConfigExternalTransportRemove

(

const SIPX_TRANSPORT

hTransport

)

Removes an external transport mechanism from the given instance.

Will fail if transport in use.

Parameters

hTransport

Handle to the external transport object. Obtained via a call to sipxConfigExternalTransportAdd

SIPXTAPI_API SIPX_RESULT sipxConfigExternalTransportRouteByUser	(	const SIPX_TRANSPORT	hTransport,
		bool	bRouteByUser
	)

The external transport mechanism can be configured to route by user or by destination ip/port.

User is the default.

Parameters

hTransport	Handle to the external transport object. Obtained via a call to sipxConfigExternalTransportAdd
bRouteByUser	true to route by user (default), false to route by destination ip/port.

SIPXTAPI_API SIPX_RESULT sipxConfigGetAllLocalNetworkIps	(	const char *	arrAddresses[],
		const char *	arrAddressAdapter[],
		int &	numAddresses
	)

Populates an array of IP Addresses in char* form.

The array must be preallocated to contain MAX_IP_ADDRESSES elements.

WARNING: THIS API IS DEPRECATED AND WILL CHANGE IN FUTURE REVISIONS

Parameters

arrAddresses	Pre-allocated array to be popluated with ip addresses.
arrAddressAdapter	For each record in arrAddresses, there is a corresponding record, with the same index, in arrAddressAdpater which represents the "sipx adapter name" for that address
numAddresses	Input: Size of the preallocated arrays. Output: Number of IPs found by the system.

SIPXTAPI_API SIPX_RESULT sipxConfigGetAudioCodec	(	const SIPX_INST	hInst,
		const int	index,
		SIPX_AUDIO_CODEC *	pCodec
	)

Get the audio codec at a certain index in the list of codecs.

Use this function in conjunction with sipxConfigGetNumAudioCodecs to enumerate the list of audio codecs. This method will return SIPX_RESULT_SUCCESS if able to set the audio codec preferences. SIPX_RESULT_FAILURE is returned if the audio codec can not be retrieved.

Parameters

hInst	Instance pointer obtained by sipxInitialize
index	Index in the list of codecs
pCodec	SIPX_AUDIO_CODEC structure that holds information (name, bandwidth requirement) about the codec.

SIPXTAPI_API SIPX_RESULT sipxConfigGetAudioCodecPreferences	(	const SIPX_INST	hInst,
		SIPX_AUDIO_BANDWIDTH_ID *	pBandWidth
	)

Get the current codec preference.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pBandWidth	pointer to an integer that will contain AUDIO_CODEC_BW_LOW, AUDIO_CODEC_BW_NORMAL, or AUDIO_CODEC_BW_HIGH. AUDIO_CODEC_BW_CUSTOM will be returned if a specific codec was et using the sipxConfigSetAudioCodecByName function.

SIPXTAPI_API SIPX_RESULT sipxConfigGetLocalContacts	(	const SIPX_INST	hInst,
		SIPX_CONTACT_ADDRESS	addresses[],
		size_t	nMaxAddresses,
		size_t &	nActualAddresses
	)

Get the local contact address available for outbound/inbound signaling and audio.

The local contact addresses will always include the local IP addresses. The local contact addresses may also include external NAT- derived addresses (e.g. STUN). See the definition of SIPX_CONTACT_ADDRESS for more details on the structure.

Determining which contact address to use depends on your network topology. If you have a proxy/edge proxy within the same firewall/NAT space, you can use the LOCAL UDP, TCP, or TLS contact type for your calls. If your proxy resides outside of the firewall/NAT space, you should use the NAT_MAPPED or RELAY contact type (UDP only). Both NAT_MAPPED and RELAY use your STUN-derived IP address, however RELAY requests TURN for media paths.

Parameters

hInst	Instance pointer obtained by sipxInitialize
addresses	A pre-allocated list of SIPX_CONTACT_ADDRESS structures. This data will be filled in by the API call.
nMaxAddresses	The maximum number of addresses supplied by the addresses parameter.
nActualAddresses	The actual number of addresses populated in the addresses parameter.

SIPXTAPI_API SIPX_RESULT sipxConfigGetLocalFeedbackAddress	(	const SIPX_INST	hInst,
		const char *	szRemoteIp,
		const int	iRemotePort,
		char *	szContactIp,
		size_t	nContactIpLength,
		int &	iContactPort,
		int	iTimeoutMs
	)

Get our local ip/port combination for the designated remote ip/port.

This API will look at all of the stun and/or SIP message results to see if a NAT binding exists for this particular host. If using a proxy server, this is generally never needed, however, in peer-to-peer modes this can sometimes help you get through NATs when using out-of-band registrars / signaling helpers (not recommended – use a proxy instead).

For this API to be useful, you need to add a keepalive to the remote host prior to calling this API. This may optionally block if a keep-alive request has been started, but we are waiting for a response.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szRemoteIp	IP of remote party
iRemotePort	port or remote party
szContactIp	Buffer to place local contact IP if successful
nContactIpLength	Length of szContactIp buffer
iContactPort	Int to place contact port
iTimeoutMs	Timeout in MS. Values of 0 (or less) signal not to block. Any other value is rounded up to multiple of 50ms. For VoIP, a value of 500ms seems plenty (latency longer than 300ms will result in a fairly bad audio experience).

SIPXTAPI_API SIPX_RESULT sipxConfigGetLocalSipTcpPort	(	SIPX_INST	hInst,
		int *	pPort
	)

Get the local TCP port for SIP signaling.

The port is supplied in the call to sipXinitialize; however, the port may be allocated dynamically. This method will return SIPX_RESULT_SUCCESS if able to return the port value. SIPX_RESULT_FAILURE is returned if the protocol is not enabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pPort	Pointer to a port number. This value must not be NULL.

SIPXTAPI_API SIPX_RESULT sipxConfigGetLocalSipTlsPort	(	SIPX_INST	hInst,
		int *	pPort
	)

Get the local TLS port for SIP signaling.

The port is supplied in the call to sipXinitialize; however, the port may be allocated dynamically. This method will return SIPX_RESULT_SUCCESS if able to return the port value. SIPX_RESULT_FAILURE is returned if the protocol is not enabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pPort	Pointer to a port number. This value must not be NULL.

SIPXTAPI_API SIPX_RESULT sipxConfigGetLocalSipUdpPort	(	SIPX_INST	hInst,
		int *	pPort
	)

Get the local UDP port for SIP signaling.

The port is supplied in the call to sipXinitialize; however, the port may be allocated dynamically. This method will return SIPX_RESULT_SUCCESS if able to return the port value. SIPX_RESULT_FAILURE is returned if the protocol is not enabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pPort	Pointer to a port number. This value must not be NULL.

SIPXTAPI_API SIPX_RESULT sipxConfigGetNumAudioCodecs	(	const SIPX_INST	hInst,
		int *	pNumCodecs
	)

Get the number of audio codecs.

This method will return SIPX_RESULT_SUCCESS if able to set the audio codec preferences. SIPX_RESULT_FAILURE is returned if the number of codecs can no be retrieved.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pNumCodecs	Pointer to the number of codecs. This value must not be NULL.

SIPXTAPI_API SIPX_RESULT sipxConfigGetNumVideoCodecs	(	const SIPX_INST	hInst,
		int *	pNumCodecs
	)

Get the number of video codecs.

This method will return SIPX_RESULT_SUCCESS if able to set the audio codec preferences. SIPX_RESULT_FAILURE is returned if the number of codecs can no be retrieved.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pNumCodecs	Pointer to the number of codecs. This value must not be NULL.

SIPXTAPI_API SIPX_RESULT sipxConfigGetVersion	(	char *	szVersion,
		const size_t	nBuffer
	)

Get the sipXtapi API version string.

Parameters

szVersion	Buffer to store the version string. A zero-terminated string will be copied into this buffer on success.
nBuffer	Size of szBuffer in bytes (not to exceed). A size of 48 bytes should be sufficient in most cases.

SIPXTAPI_API SIPX_RESULT sipxConfigGetVideoCaptureDevice	(	const SIPX_INST	hInst,
		char *	szCaptureDevice,
		int	nLength
	)

Gets the current video capture device.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szCaptureDevice	Character array to be populated by this function call.
nLength	Max length of szCaptureDevice buffer.

SIPXTAPI_API SIPX_RESULT sipxConfigGetVideoCaptureDevices	(	const SIPX_INST	hInst,
		char **	arrSzCaptureDevices,
		int	nDeviceStringLength,
		int	nArrayLength
	)

Gets the list of video capture devices.

Parameters

hInst	Instance pointer obtained by sipxInitialize
arrSzCaptureDevices	Array of character arrays to be populated by this function call.
nDeviceStringLength	Length of buffer in arrSzCaptureDevice array.
nArrayLength	Number of strings (of length nDeviceStringLength) in the arrSzCaptureDevice array.

SIPXTAPI_API SIPX_RESULT sipxConfigGetVideoCodec	(	const SIPX_INST	hInst,
		const int	index,
		SIPX_VIDEO_CODEC *	pCodec
	)

Get the video codec at a certain index in the list of codecs.

Use this function in conjunction with sipxConfigGetNumVideoCodecs to enumerate the list of video codecs. This method will return SIPX_RESULT_SUCCESS if able to set the video codec preferences. SIPX_RESULT_FAILURE is returned if the video codec can not be retrieved.

Parameters

hInst	Instance pointer obtained by sipxInitialize
index	Index in the list of codecs
pCodec	SIPX_VIDEO_CODEC structure that holds information (name, bandwidth requirement) about the codec.

SIPXTAPI_API SIPX_RESULT sipxConfigGetVideoCodecPreferences	(	const SIPX_INST	hInst,
		SIPX_VIDEO_BANDWIDTH_ID *	pBandWidth
	)

Get the current codec preference.

Parameters

hInst	Instance pointer obtained by sipxInitialize
pBandWidth	pointer to an integer that will contain AUDIO_CODEC_BW_LOW, AUDIO_CODEC_BW_NORMAL, or AUDIO_CODEC_BW_HIGH. AUDIO_CODEC_BW_CUSTOM will be returned if a specific codec was set using the sipxConfigSetVideoCodecByName function.

SIPXTAPI_API SIPX_RESULT sipxConfigIsInBandDTMFEnabled	(	const SIPX_INST	hInst,
		bool &	bEnable
	)

Determines if sending of in-band DTMF tones is enabled or disabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	in-band DTMF tones enabled or disabled.

SIPXTAPI_API SIPX_RESULT sipxConfigIsOutOfBandDTMFEnabled	(	const SIPX_INST	hInst,
		bool &	bEnable
	)

Determines if sending of out-of-band DTMF tones is enabled or disabled.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bEnable	Out-of-band DTMF tones enabled or disabled.

SIPXTAPI_API SIPX_RESULT sipxConfigKeepAliveAdd	(	const SIPX_INST	hInst,
		SIPX_CONTACT_ID	contactId,
		SIPX_KEEPALIVE_TYPE	type,
		const char *	remoteIp,
		int	remotePort,
		int	keepAliveSecs
	)

Add a signaling keep alive to a remote ip address.

Parameters

hInst	Instance pointer obtained by sipxInitialize
contactId	Contact ID used for the keep alive. sipXtapi will send keep alives from the interface identified by the contactId. Specify a contactId of -1 to send keep alives from all interfaces.
type	Designates the method of keep alives.
remoteIp	Remote IP address used to send keep alives. The caller is responsible for converting hostnames to IP address.
remotePort	Remote port used to send keep alives.
keepAliveSecs	The number of seconds to wait before resending. If the value is <= 0, only one keep alive will be sent (calling sipxConfigKeepAliveRemove will fail).

See also

sipxConfigGetLocalContacts

SIPXTAPI_API SIPX_RESULT sipxConfigKeepAliveRemove	(	const SIPX_INST	hInst,
		SIPX_CONTACT_ID	contactId,
		SIPX_KEEPALIVE_TYPE	type,
		const char *	remoteIp,
		int	remotePort
	)

Remove a signaling keepalive.

Parameters

hInst	Instance pointer obtained by sipxInitialize
contactId	Contact ID used for the keep alive. sipXtapi will remove keep alives from the interface identified by the contactId. Specify a contactId of -1 to remove keep alives from all interfaces.
type	Designates the method of keep alives.
remoteIp	Remote IP address used to send keep alives. The caller is responsible for converting hostnames to IP address. This value must match what was specified in sipxConfigKeepAliveAdd.
remotePort	Remote port used to send keep alives.

See also

sipxConfigGetLocalContacts

SIPXTAPI_API SIPX_RESULT sipxConfigPrepareToHibernate

(

const SIPX_INST

hInst

)

Call this function to prepare a sipXtapi instance for a system hibernation.

This function is not thread-safe.

Parameters

hInst

Instance pointer obtained by sipxInitialize

SIPXTAPI_API SIPX_RESULT sipxConfigResetVideoCodecs

(

const SIPX_INST

hInst

)

Reset the codec list if it was modified by sipxConfigSetVideoCodecByName.

This resets the selection to a full codec list. This method will return SIPX_RESULT_SUCCESS if able to set the audio codec. SIPX_RESULT_FAILURE is returned if the codec is not set.

Parameters

hInst

Instance pointer obtained by sipxInitialize

SIPXTAPI_API SIPX_RESULT sipxConfigSetAudioCodecByName	(	const SIPX_INST	hInst,
		const char *	szCodecName
	)

Set the codec preference order by name.

The codec name(s) must match one of the supported codecs otherwise this function will fail. This method will return SIPX_RESULT_SUCCESS if able to set the audio codec. SIPX_RESULT_FAILURE is returned if the codec is not set.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szCodecName	space delimited list of codec names. Note: TELEPHONE-EVENT is implicitly included

SIPXTAPI_API SIPX_RESULT sipxConfigSetAudioCodecPreferences	(	const SIPX_INST	hInst,
		SIPX_AUDIO_BANDWIDTH_ID	bandWidth
	)

Set the preferred bandwidth requirement for codec selection.

Whenever possible a codec matching that requirement will be selected for a call. This method will return SIPX_RESULT_SUCCESS if able to set the audio codec preferences. SIPX_RESULT_FAILURE is returned if the preference is not set.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bandWidth	Valid bandwidth requirements are AUDIO_CODEC_BW_LOW, AUDIO_CODEC_BW_NORMAL, and AUDIO_CODEC_BW_HIGH.

SIPXTAPI_API SIPX_RESULT sipxConfigSetConnectionIdleTimeout	(	const SIPX_INST	hInst,
		const int	idleTimeout
	)

Set the connection idle timeout.

If a media connection is idle for this threshold, a SILENCE event will be fired to the application layer.

Applications may decide to tear down the call after receiving this event under the assumption that the remote party is gone away. Be careful when using codecs that support silence suppression – Some implementations continue to send RTP heartbeats, however, others will not send any data and may appear to be dead.

Parameters

hInst	Instance pointer obtained by sipxInitialize
idleTimeout	The time in seconds that a socket is idle before a MEDIA_REMOTE_SILENT event is fired.

SIPXTAPI_API SIPX_RESULT sipxConfigSetDnsSrvFailoverTimeout	(	const SIPX_INST	hInst,
		const int	failoverTimeoutInSecs
	)

Specifies the time to wait before trying the next DNS SRV record.

The user agent will attempt to obtain DNS SRV resolutions for the child DNS SRV records. This setting is the time allowed for attempting a lookup - if the time expires without a lookup, then next child is attempted.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
failoverTimeoutInSecs	Number of seconds until the next DNS SRV record is tried.

SIPXTAPI_API SIPX_RESULT sipxConfigSetDnsSrvTimeouts	(	const int	initialTimeoutInSecs,
		const int	retries
	)

Modifies the timeout values used for DNS SRV lookups.

In generally, you shouldn't need to modified these, however, if you find yourself in a situation where a router/network fails to send responses to DNS SRV requests these values can be tweaked. Note, failing to send responses is different then a receiving an no-such-animal response.

The default values are initialTimeout = 5 seconds, and 4 retries. The time waited is doubled after each timeout, so with the default settings, a single DNS SRV can block for 75 seconds (5 + 10 + 20 + 40). In general, 4 DNS SRV requests are made for each hostname (e.g. domain.com):

• _sip._udp.domain.com
• _sip._tcp.domain.com
• _sip._udp.domain.com.domain.com
• _sip._tcp.domain.com.domain.com

If DNS response are dropped in the network (or your DNS server is down), the API will block for 3 minutes.

SIPXTAPI_API SIPX_RESULT sipxConfigSetEnergyLevelNotificationPeriod	(	const SIPX_INST	hInst,
		int	periodMs
	)

SIPXTAPI_API SIPX_RESULT sipxConfigSetLocationHeader	(	const SIPX_INST	hInst,
		const char *	szHeader
	)

Sets the location header for SIP messages.

The location header will be included in SIP requests and responses. You can disable the location header on a call-by-call basis in the by changing the bEnableLocationHeader flag on sipxCallAccept, sipxCallConnect, and sipxConferenceAdd methods.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szHeader	- Location header

See also

sipxCallAccept

sipxCallConnect

sipxConferenceAdd

SIPXTAPI_API SIPX_RESULT sipxConfigSetLogCallback

(

sipxLogCallback

pCallback

)

Set a callback function to collect logging information.

This function directs logging output to the specified function.

Parameters

pCallback

is a pointer to a callback function. This callback function gets passed three strings, first string is the priority level, second string is the source id of the subsystem that generated the message, and the third string is the message itself.

SIPXTAPI_API SIPX_RESULT sipxConfigSetLogFile

(

const char *

szFilename

)

The sipxConfigSetlogFile method sets the filename of the log file and directs output to that file.

NOTE: At this time no validation is performed on the specified filename. Please make sure the directories exist and the appropriate permissions are available.

Parameters

szFilename

The filename for the log file. Designated a NULL filename will disable logging, however, threads/resources will not be deallocated.

SIPXTAPI_API SIPX_RESULT sipxConfigSetLogLevel

(

SIPX_LOG_LEVEL

logLevel

)

The sipxConfigEnableLog method enables logging for the sipXtapi API, media processing, call processing, SIP stack, and OS abstraction layer.

Logging is disabled by default. The underlying framework makes no attempts to bound the log file to a fixed size.

Log Format: time:event id:facility:priority:host name:task name:task id:process id:log message

Parameters

logLevel

Designates the amount of detail includes in the log. See SIPX_LOG_LEVEL for more details.

SIPXTAPI_API SIPX_RESULT sipxConfigSetMicAudioHook

(

fnMicAudioHook

hookProc

)

Designate a callback routine as a microphone replacement or supplement.

The callback is invoked with microphone data and the data can be reviewed, modified, replaced, or discarded.

This callback proc must NOT block and must return data quickly. Additionally, the method should not call any blocking function (i.e. IO operations, malloc, new, etc).

Data must be formatted as mono 16-bit signed PCM, little endian, 8000 samples per second. The callback is handed 80 samples (10ms) of data at a time.

SIPXTAPI_API SIPX_RESULT sipxConfigSetOutboundProxy	(	const SIPX_INST	hInst,
		const char *	szProxy
	)

Defines the SIP proxy used for outbound requests.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szProxy	the new outbound proxy

SIPXTAPI_API SIPX_RESULT sipxConfigSetRegisterExpiration	(	const SIPX_INST	hInst,
		const int	nRegisterExpirationSecs
	)

Specifies the expiration period for registration.

After setting this configuration, all subsequent REGISTER messages will be sent with the new registration period.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
nRegisterExpirationSecs	Number of seconds until the expiration of a REGISTER message

SIPXTAPI_API SIPX_RESULT sipxConfigSetRegisterResponseWaitSeconds	(	const SIPX_INST	hInst,
		const int	seconds
	)

Specifies the time to wait for a REGISTER response before sending a LINESTATE_REGISTER_FAILED (or LINESTATE_UNREGISTER_FAILED) message.

If not set, the user-agent will use a 4 second timeout.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
seconds	The number of seconds to wait for a REGISTER response, before it is considered a failure.

SIPXTAPI_API SIPX_RESULT sipxConfigSetSecurityParameters	(	const SIPX_INST	hInst,
		const char *	szDbLocation,
		const char *	szMyCertNickname,
		const char *	szDbPassword
	)

Set security parameters for an instance of sipXtapi.

Deprecated:

These should be set using sipxInitialize.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szDbLocation	The directory in which the certificate database resides.
szMyCertNickname	The local user's certificate nickname, for database lookup.
szDbPassword	The password for the certificated database.

SIPXTAPI_API SIPX_RESULT sipxConfigSetSipAcceptLanguage	(	const SIPX_INST	hInst,
		const char *	szLanguage
	)

Sets the Accept Language used in sip messages.

e.g. - "EN"

Parameters

hInst	Instance pointer obtained by sipxInitialize
szLanguage	- Accept Language string

SIPXTAPI_API SIPX_RESULT sipxConfigSetSpkrAudioHook

(

fnSpkrAudioHook

hookProc

)

Designate a callback routine for post-mixing audio data (e.g.

to speaker data). The hook may review, modify, replace, or discard data.

This callback proc must NOT block and must return data quickly. Additionally, the method should not call any blocking function (i.e. IO operations, malloc, new, etc).

Data must be formatted as mono 16-bit signed PCM, little endian, 8000 samples per second. The callback is handed 80 samples (10ms) of data at a time.

SIPXTAPI_API SIPX_RESULT sipxConfigSetSubscribeExpiration	(	const SIPX_INST	hInst,
		const int	nSubscribeExpirationSecs
	)

Specifies the expiration period for subscription.

After setting this configuration, all subsequent SUBSCRIBE messages will be sent with the new subscribe period.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
nSubscribeExpirationSecs	Number of seconds until the expiration of a SUBSCRIBE message

SIPXTAPI_API SIPX_RESULT sipxConfigSetUserAgentName	(	const SIPX_INST	hInst,
		const char *	szName,
		const bool	bIncludePlatformName = true
	)

Sets the User-Agent name to be used with outgoing SIP messages.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szName	The user-agent name.
bIncludePlatformName	Indicates whether or not to append the platform description onto the user agent name.

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoBandwidth	(	const SIPX_INST	hInst,
		SIPX_VIDEO_BANDWIDTH_ID	bandWidth
	)

Set the bandwidth parameters for video codecs.Depending on the bandwidth parameter that is passed in the settings will be set to:

VIDEO_CODEC_BW_LOW bitrate 5 kbps, framerate 10 fps VIDEO_CODEC_BW_NORMAL bitrate 70 kbps, framerate is what it was set to with sipxConfigSetVideoParameters or 30 if not set. VIDEO_CODEC_BW_HIGH bitrate 400 kbps, framerate is what it was set to with sipxConfigSetVideoParameters or 30 if not set.

This method will return SIPX_RESULT_SUCCESS if able to set the video codec preferences. SIPX_RESULT_FAILURE is returned if the preference is not set.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bandWidth	Valid bandwidth requirements are VIDEO_CODEC_BW_LOW, VIDEO_CODEC_BW_NORMAL, and VIDEO_CODEC_BW_HIGH.

See also

sipxConfigSetVideoParameters

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoBitrate	(	const SIPX_INST	hInst,
		const int	bitRate
	)

Sets the video bitrate.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bitRate	Bit rate parameter

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoCaptureDevice	(	const SIPX_INST	hInst,
		const char *	szCaptureDevice
	)

Sets the video capture device.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szCaptureDevice	Pointer to a character array containing the name of the desired video capture device.

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoCodecByName	(	const SIPX_INST	hInst,
		const char *	szCodecName
	)

Set the codec by name.

The name must match one of the supported codecs otherwise this function will fail. This method will return SIPX_RESULT_SUCCESS if able to set the video codec. SIPX_RESULT_FAILURE is returned if the codec is not set.

Parameters

hInst	Instance pointer obtained by sipxInitialize
szCodecName	codec name

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoCpuUsage	(	const SIPX_INST	hInst,
		const int	cpuUsage
	)

Set the cpu usage.

Parameters

hInst	Instance pointer obtained by sipxInitialize
cpuUsage	CPU usage in percent

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoFormat	(	const SIPX_INST	hInst,
		SIPX_VIDEO_FORMAT	videoFormat
	)

Set the supported video format This method will limit the supported video format to either VIDEO_FORMAT_CIF (352x288), VIDEO_FORMAT_QCIF (176x144), VIDEO_FORMAT_SQCIF (128x92), or VIDEO_FORMAT_QVGA (320x240).

The method will return SIPX_RESULT_SUCCESS if it is able to set the video format, SIPX_RESULT_FAILURE is returned if the video format can not be set.

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoFramerate	(	const SIPX_INST	hInst,
		const int	frameRate
	)

Sets the video framerate.

Parameters

hInst	Instance pointer obtained by sipxInitialize
frameRate	Frame rate parameter

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoParameters	(	const SIPX_INST	hInst,
		const int	bitRate,
		const int	frameRate
	)

Sets the bit rate and frame rate parameters for video.

Parameters

hInst	Instance pointer obtained by sipxInitialize
bitRate	Bit rate parameter in kbps
frameRate	Frame rate parameter frames per second

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoPreviewDisplay	(	const SIPX_INST	hInst,
		SIPX_VIDEO_DISPLAY *const	pDisplay
	)

Sets the display object for the "video preview".

Parameters

hInst	Instance pointer obtained by sipxInitialize
pDisplay	Pointer to a video preview display object.

SIPXTAPI_API SIPX_RESULT sipxConfigSetVideoQuality	(	const SIPX_INST	hInst,
		const SIPX_VIDEO_QUALITY_ID	quality
	)

Sets the video quality.

Parameters

hInst	Instance pointer obtained by sipxInitialize
quality	Id setting the video quality.

SIPXTAPI_API SIPX_RESULT sipxConfigSetVoiceQualityServer	(	const SIPX_INST	hInst,
		const char *	szServer
	)

Sets the SIP target URL for voice quality reports.

Voice Quality reports are sent at the completion of each call and give details on the voice quality (latency, noise, MOS scores, etc). Presently, this is not implemented in the open source version.

This must be enabled prior to creating a call or receiving a new call indiciation. Likewise, changes will not take effect for existing calls.

Parameters

hInst	An instance handle obtained from sipxInitialize.
szServer	Target SIP URL for the voice quality reports. A value of NULL will disable voice quality reports.

SIPXTAPI_API SIPX_RESULT sipxConfigSubscribe	(	const SIPX_INST	hInst,
		const SIPX_LINE	hLine,
		const char *	szTargetUrl,
		const char *	szEventType,
		const char *	szAcceptType,
		const SIPX_CONTACT_ID	contactId,
		SIPX_SUB *	phSub
	)

Subscribe for NOTIFY events which may be published by another end-point or server.

sipXtapi will automatically refresh subscriptions until sipxConfigUnsubscribe is called. Please make sure you call sipxCallUnsubscribe before tearing down your instance of sipXtapi.

Parameters

hInst	Instance pointer obtained by sipxInitialize
hLine	Line Identity for the outbound call. The line identity helps defines the "From" caller-id.
szTargetUrl	The Url of the publishing end-point.
szEventType	A string representing the type of event that can be published. This string is used to populate the "Event" header in the SIP SUBSCRIBE request. For example, if checking voicemail status, your would use "message-summary".
szAcceptType	A string representing the types of NOTIFY events that this client will accept. This string is used to populate the "Accept" header in the SIP SUBSCRIBE request. For example, if checking voicemail status, you would use "application/simple-message-summary"
contactId	Id of the desired contact record to use for this call. The id refers to a Contact Record obtained by a call to sipxConfigGetLocalContacts. The application can choose a contact record of type LOCAL, NAT_MAPPED, CONFIG, or RELAY. The Contact Type allows you to control whether the user agent and media processing advertises the local address (e.g. LOCAL contact of 10.1.1.x or 192.168.x.x), the NAT-derived address to the target party, or, local contact addresses of other types.
phSub	Pointer to a subscription handle whose value is set by this funtion. This handle allows you to cancel the subscription and differeniate between NOTIFY events.

SIPXTAPI_API SIPX_RESULT sipxConfigUnHibernate

(

const SIPX_INST

hInst

)

Call this function upon returning from a system hibernation.

This function is not thread-safe.

Parameters

hInst

Instance pointer obtained by sipxInitialize

SIPXTAPI_API SIPX_RESULT sipxConfigUnsubscribe

(

const SIPX_SUB

hSub

)

Unsubscribe from previously subscribed NOTIFY events.

This method will send another subscription request with an expires time of 0 (zero) to end your subscription.

Parameters

hSub	The subscription handle obtained from the call to sipxConfigSubscribe.

SIPXTAPI_API SIPX_RESULT sipxConfigUpdatePreviewWindow	(	const SIPX_INST	hInst,
		const SIPX_WINDOW_HANDLE	hWnd
	)

Updates the Preview window with a new frame buffer.

Should be called when the window receives a PAINT message.

Parameters

hInst	Instance pointer obtained by sipxInitialize
hWnd	Window handle of the video preview window.

SIPXTAPI_API SIPX_RESULT sipxInitialize	(	SIPX_INST *	phInst,
		const int	udpPort = DEFAULT_UDP_PORT,
		const int	tcpPort = DEFAULT_TCP_PORT,
		const int	tlsPort = DEFAULT_TLS_PORT,
		const int	rtpPortStart = DEFAULT_RTP_START_PORT,
		const int	maxConnections = DEFAULT_CONNECTIONS,
		const char *	szIdentity = DEFAULT_IDENTITY,
		const char *	szBindToAddr = DEFAULT_BIND_ADDRESS,
		bool	bUseSequentialPorts = false,
		const char *	szTLSCertificateNickname = NULL,
		const char *	szTLSCertificatePassword = NULL,
		const char *	szDbLocation = NULL,
		bool	bEnableLocalAudio = true,
		const int	internalSamplerate = 8000,
		const int	devicesSamplerate = 48000,
		const int	internalFrameSizeMs = 10,
		const char *	callInputDeviceName = "",
		const char *	callOutputDeviceName = ""
	)

Initialize the sipX TAPI-like API layer.

This method initialized the basic SIP stack and media process resources and must be called before any other sipxXXX methods. Additionally, this method fills in a SIPX_INST parameter which must be passed to a number of sipX methods.

Parameters

phInst	A pointer to a hInst that must be various other sipXtapi routines.
udpPort	The default UDP port for the SIP protocol stack. The port cannot be changed after initialization. Right now, the UDP port and TCP port numbers MUST be equal. Pass a value of SIPX_PORT_DISABLE (-1) to disable disable UDP or a value of SIPX_PORT_AUTO (-2) to automatically select an open UDP port.
tcpPort	The default TCP port for the SIP protocol stack. The port cannot be changed after initialization. Right now, the UDP port and TCP port numbers MUST be equal. Pass a value of SIPX_PORT_DISABLE (-1) to disable disable TCP or a value of SIPX_PORT_AUTO (-2) to automatically select an open TCP port.
tlsPort	NOT YET SUPPORTED
rtpPortStart	The starting port for inbound RTP traffic. The sipX layer will use ports starting at rtpPortStart and ending at (rtpPortStart + 2 * maxConnections) - 1. Pass a value of SIPX_PORT_AUTO (-2) to automatically select an open port.
maxConnections	The maximum number of simultaneous connections that the sipX layer will support.
szIdentity	The default outbound identity used by the SIP stack if no lines are defined. Generally, the szIdentity is only used for inbound calls since all of sipXtapi APIs required a line ID for outbound calls. The identity will be used to form the "From" field (caller-id) and the username/URL parameters are may be used as part of the "Contact" header. In other words, this field does not impact any routing aspects of the call session.
szBindToAddr	Defines which IP/address the user agent / rtp stack will listen on. The default "0.0.0.0" listens on all interfaces. The address must be in dotted decimal form – hostnames will not work.
bUseSequentialPorts	If unable to bind to the udpPort, tcpPort, or tlsPort, try sequential ports until a successful port is found. If enabled, sipXtapi will try 10 sequential port numbers after the initial port.
szTLSCertificateNickname	Nickname of the certificate to use as an SSL server.
szTLSCertificatePassword	Password for the SSL server certificate.
szDbLocation	Path to the certificate database.
bEnableLocalAudio	Enable use of local microphone and speaker. Set to false if you're running on a server hardware without a sound card.
internalSamplerate	Samplerate at which all internal processing is performed. E.g. if you use G.722 codec which has 16K samplerate, but set inernal samplerate to 8000, then you will hear little difference from G.711, becase incoming audio will be downsampled to 8000 and outgoing audio will be upsampled from 8000. Note however, that working with 8000 internal samplerate requires less CPU power, and thus it is set by default to 8000.
devicesSamplerate	Samplerate used by audio input/output drivers. It is set to 48000 by default, because it's the native samplerate for the most modern soundcards and they produce less jitter and drift at this samplerate. We downsample/upsample to internal samplerate transparently under the hood.
internalFrameSizeMs	Size of an audio frame (in milliseconds), used for internal processing. Default value of 10ms is a recommended value unless you really know what you're doing. If you want to run this on a slow CPU, you may want to set this to 20ms, but it will increase latency and thus is not generally recommended.
callInputDeviceName	Name of the audio device to use as input device during a call. Can be changed later with sipxAudioSetCallInputDevice(). Use empty string to select default (OS-dependent) device.
callOutputDeviceName	Name of the audio device to use as output device during a call. Can be changed later with sipxAudioSetCallOutputDevice(). Use empty string to select default (OS-dependent) device.

SIPXTAPI_API SIPX_RESULT sipxLineAdd	(	const SIPX_INST	hInst,
		const char *	szLineURL,
		SIPX_LINE *	phLine,
		SIPX_CONTACT_ID	contactId = 0
	)

Adds a line appearance.

A line appearance defines your address of record and is used both as your "From" caller-id and as the public identity to which you will receive calls for. Directing calls to a particular user agent is achieved using registrations.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szLineURL	The address of record for the line identity. Can be prepended with a Display Name. e.g. - "Zaphod Beeblebrox" <sip:zaphb@fourty-two.net>
phLine	Pointer to a line handle. Upon success, a handle to the newly added line is returned.
contactId	Id of the desired contact record to use for this line. The id refers to a Contact Record obtained by a call to sipxConfigGetLocalContacts. The application can choose a contact record of type LOCAL, NAT_MAPPED, CONFIG, or RELAY. The Contact Type allows you to control whether the user agent and media processing advertises the local address (e.g. LOCAL contact of 10.1.1.x or 192.168.x.x), the NAT-derived address to the target party, or, local contact addresses of other types.

See also

sipxConfigGetLocalContacts

SIPXTAPI_API SIPX_RESULT sipxLineAddAlias	(	const SIPX_LINE	hLine,
		const char *	szLineURL
	)

Adds an alias for a line definition.

Line aliases are used to map an inbound call request to an existing line definition. You should only need to an a alias if your network infrastructure directs calls to this user agent using multiple identities. For example, if user agent registers as "sip:bandreasen\@example.com"; however, calls can also be directed to you via an exention (e.g. sip:122@example.com).

If sipXtapi receives a call with an unknown line, you can still answer and interact wtih the call; however, the line handle will be SIPX_LINE_NULL in all event callbacks. Adding an aliases allows you to correlate another line url with your line definition and receive real line handles with event callbacks.

Line aliases are not used for outbound calls.

See also

sipxConfigGetLocalContacts

SIPXTAPI_API SIPX_RESULT sipxLineAddCredential	(	const SIPX_LINE	hLine,
		const char *	szUserID,
		const char *	szPasswd,
		const char *	szRealm
	)

Adds authentication credentials to the designated line appearance.

Credentials are often required by registration services to verify that the line is being used by the line appearance/address of record owner.

Parameters

hLine	Handle to a line appearance. Line handles are obtained by creating a line using the sipxLineAdd function or by receiving a line event notification.
szUserID	user id used for the line appearance.
szPasswd	passwd used for the line appearance.
szRealm	realm for which the user and passwd are valid. Supply NULL for automatic realm.

SIPXTAPI_API SIPX_RESULT sipxLineFindByURI	(	const SIPX_INST	hInst,
		const char *	szURI,
		SIPX_LINE &	hLine
	)

Find a line definition given a URI.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
szURI	URI used to search for a line definition
hLine	line handle if successful

SIPXTAPI_API SIPX_RESULT sipxLineGet	(	const SIPX_INST	hInst,
		SIPX_LINE	lines[],
		const size_t	max,
		size_t &	actual
	)

Gets the active list of line identities.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
lines	Pre-allocated array of line handles.
max	Maximum number of lines to return.
actual	Actual number of valid lines returned.

SIPXTAPI_API SIPX_RESULT sipxLineGetContactId	(	const SIPX_LINE	hLine,
		SIPX_CONTACT_ID &	contactId
	)

Get the contact ID for the designated line handle.

Parameters

hLine	Handle to a line appearance. Line handles are obtained by creating a line using the sipxLineAdd function or by receiving a line event notification.
contactId	contact Id specified during line creation

SIPXTAPI_API SIPX_RESULT sipxLineGetURI	(	const SIPX_LINE	hLine,
		char *	szBuffer,
		const size_t	nBuffer,
		size_t &	nActual
	)

Get the Line URI for the designated line handle.

Parameters

hLine	Handle to a line appearance. Line handles are obtained by creating a line using the sipxLineAdd function or by receiving a line event notification.
szBuffer	Buffer to place line URL. A NULL value will return the amount of storage needed in nActual.
nBuffer	Size of szBuffer in bytes (not to exceed)
nActual	Actual number of bytes written

SIPXTAPI_API SIPX_RESULT sipxLineRegister	(	const SIPX_LINE	hLine,
		const bool	bRegister
	)

Registers a line with the proxy server.

Registrations will be re-registered automatically, before they expire.

Unless your user agent is designated a static IP address or DNS name and that routing information is provisioned into a SIP server, you should register the line by calling this function.

Please unregister your line before calling sipxLineRemove.

Parameters

hLine	Handle to a line appearance. Line handles are obtained by creating a line using the sipxLineAdd function or by receiving a line event notification.
bRegister	true if Registration is desired, otherwise, an Unregister is performed.

SIPXTAPI_API SIPX_RESULT sipxLineRemove

(

SIPX_LINE

hLine

)

Remove the designated line appearance.

If the line was previous registered using the sipxLineRegister API, please unregister the line and wait for the unregistered event before calling sipxLineRemove. Otherwise, the line will be removed without unregistering.

Parameters

hLine

Handle to a line appearance. Line handles are obtained by creating a line using the sipxLineAdd function or by receiving a line event notification.

SIPXTAPI_API SIPX_RESULT sipxMediaConnectionCreate	(	const SIPX_CONF	hConf,
		int &	connectionId
	)

Create a new media connetion that is indpendent of any existing call legs in the conference.

This media connection is for independent control independent of SIP call control. RTP streams may be stopped and started manually via the sipXtapi methods: sipxConferenceRtpSetDestination sipxConferenceRtpStartSend sipxConferenceRtpStopSend

Parameters

[in]	hConf	- conference handle indicating which media interface in which the new media connection is to be created.
[out]	connectionId	- handle/id for the media connection created.

SIPXTAPI_API SIPX_RESULT sipxMediaConnectionRtpSetDestination	(	const SIPX_CONF	hConf,
		int	connectionId,
		MEDIA_TYPE	mediaType,
		int	mediaTypeStreamIndex,
		const char *	streamSendAddress,
		int	rtpPort,
		int	rtcpPort
	)

Set the destination to which RTP streams are to be sent.

Parameters

[in]	hConf	- conference handle indicating which media interface the connection belongs.
[in]	connectionId	- handle/id for media connection
[in]	mediaType	- media stream type (e.g. VIDEO_MEDIA, AUDIO_MEDIA)
[in]	mediaTypeStreamIndex	- index to which stream for the given type of media for now limited to 1 stream (i.e. index = 0)
[in]	streamSendAddress	- remote IP address to send RTP to
[in]	rtpPort	- port at remote address to send RTP stream to
[in]	rtcpPort	- port at remote address to send RTCP stream to

SIPXTAPI_API SIPX_RESULT sipxMediaConnectionRtpStartSend	(	const SIPX_CONF	hConf,
		int	connectionId,
		int	numTokens,
		const char *	codecTokens[],
		int	payloadIds[]
	)

Start sending the RTP/RTCP stream.

Parameters

[in]	hConf	- conference handle indicating which media interface the connection belongs.
[in]	connectionId	- handle/id for media connection
[in]	numTokens	- number of codec tokens in codecTokens array
[in]	codecTokens	- array of codec names
[in]	payloadIds	- payload ID to use for corresponding codec in codecTokens array

SIPXTAPI_API SIPX_RESULT sipxMediaConnectionRtpStopSend	(	const SIPX_CONF	hConf,
		int	connectionId
	)

Stop sending RTP/RTCP.

Parameters

[in]	hConf	- conference handle indicating which media interface the connection belongs.
[in]	connectionId	- handle/id for media connection

SIPXTAPI_API SIPX_RESULT sipxPublisherCreate	(	const SIPX_INST	hInst,
		SIPX_PUB *	phPub,
		const char *	szResourceId,
		const char *	szEventType,
		const char *	szContentType,
		const char *	pContent,
		const size_t	nContentLength
	)

Creates a publishing context, which performs the processing necessary to accept SUBSCRIBE requests, and to publish NOTIFY messages to subscribers.

The resource may be specific to a single call, conference or global to this user agent. The naming of the resource ID determines the scope.

Parameters

hInst	Instance pointer obtained by sipxInitialize.
phPub	Pointer to a publisher handle - this method modifies the value to refer to the newly created publishing context.
szResourceId	The resourceId to the state information being published. This must match the request URI of the incoming SUBSCRIBE request (only the user ID, host and port are significant in matching the request URI). Examples: fred@10.0.0.1:5555, sip:conference1@192.160.0.1, sip:kate@example.com
szEventType	A string representing the type of event that can be published.
szContentType	String representation of the content type being published.
pContent	Pointer to the NOTIFY message's body content.
nContentLength	Size of the content to be published.

Returns

If the resource already has a a publisher created for the given event type, SIPX_RESULT_INVALID_ARGS is returned.

SIPXTAPI_API SIPX_RESULT sipxPublisherDestroy	(	const SIPX_PUB	hPub,
		const char *	szContentType,
		const char *	pFinalContent,
		const size_t	nContentLength
	)

Tears down the publishing context.

Any existing subscriptions are sent a final NOTIFY request. If pFinalContent is not NULL and nContentLength > 0 the given publish state is given otherwise the final NOTIFY requests are sent with no body or state.

Parameters

hPub	Handle of the publishing context to destroy (returned from a call to sipxCreatePublisher)
szContentType	String representation of the content type being published
pFinalContent	Pointer to the NOTIFY message's body content
nContentLength	Size of the content to be published

SIPXTAPI_API SIPX_RESULT sipxPublisherUpdate	(	const SIPX_PUB	hPub,
		const char *	szContentType,
		const char *	pContent,
		const size_t	nContentLength
	)

Publishes an updated state to specific event via NOTIFY to its subscribers.

Parameters

hPub	Handle of the publishing context (returned from a call to sipxCreatePublisher)
szContentType	String representation of the content type being published
pContent	Pointer to the NOTIFY message's body content
nContentLength	Size of the content to be published

SIPXTAPI_API SIPX_RESULT sipxReInitialize	(	SIPX_INST *	phInst,
		const int	udpPort = DEFAULT_UDP_PORT,
		const int	tcpPort = DEFAULT_TCP_PORT,
		const int	tlsPort = DEFAULT_TLS_PORT,
		const int	rtpPortStart = DEFAULT_RTP_START_PORT,
		const int	maxConnections = DEFAULT_CONNECTIONS,
		const char *	szIdentity = DEFAULT_IDENTITY,
		const char *	szBindToAddr = DEFAULT_BIND_ADDRESS,
		bool	bUseSequentialPorts = false,
		const char *	szTLSCertificateNickname = NULL,
		const char *	szTLSCertificatePassword = NULL,
		const char *	szDbLocation = NULL
	)

Re-initialize the sipX TAPI-like API layer.

This method will remove all lines, conferences, calls, publishers, and subscribers, and listeners. Before calling this function, the application should unregister all registered lines.

Your listeners are NOT maintained. You should also reset any of your configuration settings (rport, outbound proxy, etc).

Parameters

phInst	A pointer to a SIPX_INST variable. Your old SIPX_INST handle will be invalid after this call. Calling any routines with the old SIPX_INST variable is undefined and may result in an exception.
udpPort	The default UDP port for the SIP protocol stack. The port cannot be changed after initialization. Right now, the UDP port and TCP port numbers MUST be equal. Pass a value of SIPX_PORT_DISABLE (-1) to disable disable UDP or a value of SIPX_PORT_AUTO (-2) to automatically select an open UDP port.
tcpPort	The default TCP port for the SIP protocol stack. The port cannot be changed after initialization. Right now, the UDP port and TCP port numbers MUST be equal. Pass a value of SIPX_PORT_DISABLE (-1) to disable disable TCP or a value of SIPX_PORT_AUTO (-2) to automatically select an open TCP port.
tlsPort	NOT YET SUPPORTED
rtpPortStart	The starting port for inbound RTP traffic. The sipX layer will use ports starting at rtpPortStart and ending at (rtpPortStart + 2 * maxConnections) - 1. Pass a value of SIPX_PORT_AUTO (-2) to automatically select an open port.
maxConnections	The maximum number of simultaneous connections that the sipX layer will support.
szIdentity	The default outbound identity used by the SIP stack if no lines are defined. Generally, the szIdentity is only used for inbound calls since all of sipXtapi APIs required a line ID for outbound calls. The identity will be used to form the "From" field (caller-id) and the username/URL parameters are may be used as part of the "Contact" header. In other words, this field does not impact any routing aspects of the call session.
szBindToAddr	Defines which IP/address the user agent / rtp stack will listen on. The default "0.0.0.0" listens on all interfaces. The address must be in dotted decimal form – hostnames will not work.
bUseSequentialPorts	If unable to bind to the udpPort, tcpPort, or tlsPort, try sequential ports until a successful port is found. If enabled, sipXtapi will try 10 sequential port numbers after the initial port.
szTLSCertificateNickname	Nickname of the certificate to use as an SSL server.
szTLSCertificatePassword	Password for the SSL server certificate.
szDbLocation	Path to the certificate database.
bEnableLocalAudio	Should local sound card be engaged or not. Setting this parameter to FALSE is useful for server applications which should work without a sound card installed.

SIPXTAPI_API SIPX_RESULT sipxUnInitialize	(	SIPX_INST	hInst,
		bool	bForceShutdown = false
	)

Uninitialize the sipX TAPI-like API layer.

This method tears down the basic SIP stack and media process resources and should be called before exiting the process. Users are responsible for ending all calls and unregistering line appearances before calling sipxUnInitialize. Failing to end calls/conferences or remove lines will result in a SIPX_RESULT_BUSY return code.

Parameters

hInst	An instance handle obtained from sipxInitialize.
bForceShutdown	forces sipXtapi to shutdown regardless of live calls/unregistered lines. Enabling this in NOT RECOMMENDED, please tear down all calls and lines prior to calling sipxUnitialize.

SIPXTAPI_API SIPX_RESULT sipxUtilUrlGetDisplayName	(	const char *	szUrl,
		char *	szDisplayName,
		size_t	nDisplayName
	)

Simple utility function to parse the display name from a SIP URL.

SIPXTAPI_API SIPX_RESULT sipxUtilUrlGetUrlParam	(	const char *	szUrl,
		const char *	szParamName,
		size_t	nParamIndex,
		char *	szParamValue,
		size_t	nParamValue
	)

Get the Nth named url parameter from the designated url.

Parameters

szUrl	The url to parse
szParamName	Name of the url parameter
nParamIndex	Index of the url parameter (zero-based). This is used if you have multiple url parameters with the same name – otherwise, you should use 0.
szParamValue	Buffer to place parameter value
nParamValue	size of parameter value buffer (szParamValue)

SIPXTAPI_API SIPX_RESULT sipxUtilUrlParse	(	const char *	szUrl,
		char *	szUsername,
		char *	szHostname,
		int *	iPort
	)

Simple utility function to parse the username, host, and port from a URL.

All url, field, and header parameters are ignored. You may also specify NULL for any parameters (except szUrl) which are not needed. Lastly, the caller must allocate enough storage space for each url component – if in doubt use the length of the supplied szUrl.

SIPXTAPI_API SIPX_RESULT sipxUtilUrlUpdate	(	char *	szUrl,
		size_t &	nUrl,
		const char *	szNewUsername,
		const char *	szNewHostname,
		const int	iNewPort
	)

Simple utility function to update a URL.

If the szUrl isn't large enough, or is NULL, this function will fail, however, the nUrl will contained the required size in bytes.

To leave an existing component unchanged, use NULL for strings and -1 for ports.

If the szUrl isn't large enough, this function will fail. Specify a NULL szUrl to request required length. To leave an existing component unchanged, use NULL for strings and -1 for ports.

Variable Documentation

const SIPX_CALL SIPX_CALL_NULL = 0

Represents a null call handle.

const SIPX_CONF SIPX_CONF_NULL = 0

Represents a null conference handle.

const SIPX_INST SIPX_INST_NULL = 0

Represents a null instance handle.

const SIPX_LINE SIPX_LINE_NULL = 0

Represents a null line handle.

const SIPX_PUB SIPX_PUB_NULL = 0

Represents a null publisher handle.

const SIPX_TRANSPORT SIPX_TRANSPORT_NULL = 0

Represents a null transport handle.