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PortSIP VoIP SDK Manual for Mac

1. audioRecordMode The audio record mode aviFileCodecType The codec which using for compress the video data to save into video record file videoRecordMode_ Allow set video record mode support record received video send video both 38 received and send Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int stopRecord long sessionld Stop record Parameters sessionld The session ID of call conversation Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Play audio and video file to remoe functions Functions int PortSIPSDK playVideoFileToRemote aviFile loop play Audio Play an AVI file to remote party int PortSIPSDK stopPlayVideoFileToRemote Stop play video file to remote side int PortSIPSDK playAudioFileToRemote filename fileSamplesPerSec loop Play an wave file to remote party int PortSIPSDK stopPlayAudioFileToRemote Stop play wave file to remote side int PortSIPSDK playAudioFileToRemoteAsBackground filename fileSamplesPerSec Play an wave file to remote party as conversation background sound 6 int PortSIPSDK stopPlayAudioFileToRemoteAsBackground Stop play an wave file to remote party as conversation background sound Detailed Description
2. int attendedRefer replaceSessionId referTo Make an attended refer long acceptRefer referSignaling Accept the REFER request a new call will be make if called this function usuall called after onReceivedRefer callback event int rejectRefer Reject the REFER request int enableSendPcmStreamToRemote state streamSamplesPerSec Enable the SDK send PCM stream data to remote side from another source to instread of microphone int sendPcmStreamToRemote data Send the audio stream in PCM format from another source to instead of audio device capture microphone int enableSend VideoStreamToRemote state Enable the SDK send video stream data to remote side from another source to instead of camera int send VideoStreamToRemote data width height Send the video stream to remote int setRtpCallback Set the RTP callbacks to allow access the sending and received RTP packets int enableAudioStreamCallback enable callbackMode Enable disable the audio stream callback int enable VideoStreamCallback callbackMode Enable disable the video stream callback int startRecord recordFilePath recordFileName appendTimeStamp audioFileFormat audioRecordMode av iFileCodecType videoRecordMode Start record the call int stopRecord Stop record int play VideoFileToRemote aviFile loop playAudio Play an AVI file to remote party int stopPlay VideoFileToRemote Stop play vi
3. http www PortSIP com PortSIP VoIP SDK Video Render View class description Method Documentation void updateVideoRenderFrame NSRect frameRect change the Video Render size Remarks Example NSRect rect videoRenderView frame rect size width 20 rect size height 20 videoRenderView frame rect videoRenderView setNeedsDisplay YES NSRect renderRect videoRenderView bounds videoRenderView updateVideoRenderFrame renderRect 80 The documentation for this class was generated from the following file PortSIPVideoRenderView h 81 Index lt PortSIPEventDelegate gt 70 acceptRefer referSignaling Refer functions 33 Access SIP message header functions 21 addExtensionHeader headerValue 22 clearAddExtensionHeaders 22 clearModifyHeaders 23 getExtensionHeaderValue headerName 21 modifyHeaderValue headerValue 22 addAudioCodec Audio and video codecs functions 12 addExtensionHeader headerValue Access SIP message header functions 22 Additional setting functions 14 addSupportedMimeType mimeType subMi meType 20 detectMwi 19 disableCallForward 18 disableSessionTimer 19 enable3GppTags 16 enableCallbackSendingSignaling 16 enableCallForward forwardTo 18 enableCheckMwi 19 enableReliableProvisional 16 enableSessionTimer refreshMode 18 getVersion minorVersion 16 setAudioSamples maxPtime 20 setDisplayName 15 setDoNotDisturb 19 setKeepAliveTime 20
4. BOOL existsAudio existsVideo BOOL existsVideo When the call is coming this event was triggered Parameters sessionId The session ID of the call callerDisplayNam The display name of caller e caller The caller calleeDisplayNam The display name of callee e callee The callee audioCodecs The matched audio codecs it s separated by if have more than one codec videoCodecs The matched video codecs it s separated by if have more than one codec existsAudio If it s true means this call include the audio exists Video If it s true means this call include the video void onInviteTrying long sessionid optional If the outgoing call was processing this event triggered Parameters sessionld The session ID of the call void onInviteSessionProgress long sessionid audioCodecs char audioCodecs videoCodecs char videoCodecs existsEarlyMedia BOOL existsEarlyMedia existsAudio BOOL existsAudio existsVideo BOOL existsVideo optional Once the caller received the 183 session progress message this event will be triggered Parameters sessionId The session ID of the call audioCodecs The matched audio codecs it s separated by if have more than one codec videoCodecs The matched video codecs it s separated by if have more than one codec existsEarlyMedia If it s true means the call has early media existsAudio If it s
5. Send audio and video stream functions 34 enableSessionTimer refreshMode Additional setting functions 18 enableVAD Audio effect functions 46 enable VideoStreamCallback callbackMode RTP packets Audio stream and video stream callback functions 36 forwardCall forwardTo Call functions 30 getAudioRtcpStatistics bytesSent packetsSent by tesReceived packetsReceived sendFractionLost s endCumulativeLost recvFractionLost recvCumul ativeLost RTP statistics functions 45 getAudioRtpStatistics averageJitterMs maxJitter Ms discardedPackets RTP statistics functions 44 getDynamic VolumeLevel microphone Volume Audio and video functions 27 getExtensionHeaderValue headerName Access SIP message header functions 21 getLocalIpAddress NIC and local IP functions 11 83 getMic Volume Device Manage functions 54 getNetworkStatistics currentBufferSize preferred BufferSize currentPacketLossRate currentDiscar dRate currentExpandRate currentPreemptiveRat e currentAccelerateRate RTP statistics functions 44 getNICNums NIC and local IP functions 11 getNumOfPlayoutDevices Device Manage functions 52 getNumOfRecordingDevices Device Manage functions 52 getNumOfVideoCaptureDevices Device Manage functions 52 getPlayoutDeviceName Device Manage functions 53 getRecordingDeviceName Device Manage functions 53 getSpeaker Volume Device Manage functions 53 getSystemInputMute Device Manage functions 54 getSystemOut
6. lt PortSIPEventDelegate gt onPresenceOnline from stateText void lt PortSIPEventDelegate gt onPresenceOffline from 64 Detailed Description Function Documentation void onPresenceRecvSubscribe long subscribeld fromDisplayName char fromDisplayName from char from subject char subject This event will be triggered when received the SUBSCRIBE request from a contact Parameters subscribeld The id of SUBSCRIBE request fromDisplayName_ The display name of contact from The contact who send the SUBSCRIBE request subject The subject of the SUBSCRIBE request void onPresenceOnline char fromDisplayName from char from stateText char stateText When the contact is online or changed presence status this event will be triggered Parameters fromDisplayName_ The display name of contact from The contact who send the SUBSCRIBE request stateText The presence status text void onPresenceOffline char fromDisplayName from char from When the contact is went offline then this event will be triggered Parameters fromDisplayName_ The display name of contact from The contact who send the SUBSCRIBE request MESSAGE message events Functions void lt PortSIPEventDelegate gt onRecvMessage mimeT ype subMimeType messageData messageDataLengt h void lt PortSIPEventDelegate gt onRecvOu
7. void PortSIPSDK clearAudioCodec Remove all enabled audio codecs void PortSIPSDK clearVideoCodec Remove all enabled video codecs 6 int PortSIPSDK setAudioCodecParameter parameter Set the codec parameter for audio codec 6 int PortSIPSDK setVideoCodecParameter parameter Set the codec parameter for video codec Detailed Description Function Documentation int addAudioCodec AUDIOCODEC_TYPE codecType Enable an audio codec it will be appears in SDP Parameters codecType Audio codec type Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int addVideoCodec VIDEOCODEC_TYPE codecType Enable a video codec it will be appears in SDP Parameters codecType Video codec type 13 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code BOOL isAudioCodecEmpty Detect enabled audio codecs is empty or not Returns If no audio codec was enabled the return value is true otherwise is false BOOL isVideoCodecEmpty Detect enabled video codecs is empty or not Returns If no video codec was enabled the return value is true otherwise is false int setAudioCodecPayloadType AUDIOCODEC_TYPE codecType payloadType int payloadType Set the RTP payload type for dynamic audio codec Paramet
8. Gets the number of audio devices available for audio playout Returns The return value is number of playout devices If the function fails the return value is a specific error code less than 0 53 NSString getRecordingDeviceName int index Gets the name of a specific recording device given by an index Parameters index Device index 0 1 2 N 1 where N is given by getNumOfRecordingDevices Also 1 is a valid value and will return the name of the default recording device Returns A NSString to which the device name will be copied as a null terminated string in UTF8 format NSString getPlayoutDeviceName int index Gets the name of a specific playout device given by an index Parameters index Device index 0 1 2 N 1 where N is given by getNumOfPlayoutDevices U Also 1 is a valid value and will return the name of the default playout device Returns A NSString to which the device name will be copied as a null terminated string in UTF8 format int setSpeakerVolume int volume Set the speaker volume level Parameters volume Volume level of speaker valid range is 0 255 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getSpeakerVolume Gets the speaker volume level Returns If the function succeeds the return value is speaker volume valid range is
9. void onRecvMessage mimeType subMimeType messageData messageDataLength 71 void onRecvOutOfDialogMessage from toDisplayName to mimeType subMimeType messageData messag eDataLength void onSendMessageSuccess messageld void onSendMessageFailure messageld reason code void onSendOutOfDialogMessageSuccess fromDisplayName from toDisplayName to void onSendOutOfDialogMessageFailure fromDisplayName from toDisplayName to reason code void onPlayAudioFileFinished fileName void onPlayVideoFileFinished void onReceivedRTPPacket isAudio RTPPacket packetSize void onSendingRTPPacket isAudio RTPPacket packetSize void onAudioRawCallback audioCallbackMode data dataLength samplingFreqHz void onVideoRawCallback videoCallbackMode width height data dataLength Detailed Description PortSIP SDK Callback events Delegate Author Copyright c 2006 2014 PortSIP Solutions Inc All rights reserved Version 11 2 See also http www PortSIP com PortSIP SDK Callback events Delegate description The documentation for this protocol was generated from the following file PortSIPEventDelegate h 72 PortSIPSDK Class Reference PortSIP VoIP SDK functions class import lt PortSIPSDK h gt Inherits lt NSObject gt Instance Methods int initialize logleve
10. Function Documentation int playVideoFileToRemote long sessionid aviFile NSString aviFile loop BOOL loop playAudio BOOL playAudio Play an AVI file to remote party 39 Parameters sessionId Session ID of the call aviFile The file full path name such as test avi loop Set to false to stop play video file when it is end Set to true to play it as repeat playAudio If set to true then play audio and video together set to false just play video only Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int stopPlayVideoFileToRemote long sessionld Stop play video file to remote side Parameters sessionld Session ID of the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int play AudioFileToRemote long sessionid filename NSSiring filename fileSamplesPerSec int fileSamplesPerSec loop BOOL loop Play an wave file to remote party Parameters sessionld Session ID of the call filename The file full path name such as test wav fileSamplesPerSec_ The wave file sample in seconds should be 8000 or 16000 or 32000 Set to false to stop play audio file when it is end Set to true to play it as repeat loop Returns If the function succeeds the return value is 0 If the
11. If the function fails the return value is a specific error code int enableAudioStreamCallback long sessionld enable BOOL enable callbackMode AUDIOSTREAM_CALLBACK_MODE callbackMode Enable disable the audio stream callback Parameters sessionId The session ID of call enable Set to true to enable audio stream callback false to stop the callback callbackMode The audio stream callback mode Type Description AUDIOSTREAM_LOCAL_MIX Callback the audio stream from microphone for all channels AUDIOSTREAM_LOCAL_PER_CHANNEL Callback the audio stream from microphone for one channel base on the given sessionId AUDIOSTREAM_REMOTE_MIX Callback the received audio stream that mixed including all channels AUDIOSTREAM_REMOTE_PER_CHANNE L Callback the received audio stream for one channel base on the given sessionId Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks the onAudioRawCallback event will be triggered if the callback is enabled int enableVideoStreamCallback long session d callbackMode VIDEOSTREAM_CALLBACK_MODE callbackMode Enable disable the video stream callback Parameters sessionId The session ID of call callbackMode The video stream callback mode Mode Description VIDEOSTREAM_NONE Disable video stream callback VIDEOSTR
12. PortSIPSDK setVideoNackStatus Enable disable the NACK feature rfc6642 which help to improve the video quatliy void PortSIPSDK muteMicrophone Mute the device microphone it s unavailable for Android and iOS void PortSIPSDK muteSpeaker 24 Mute the device speaker it s unavailable for Android and iOS int PortSIPSDK setAudioDeviceld outputDeviceld Set the audio device that will use for audio call void PortSIPSDK getDynamicVolumeLevel microphoneVolume Obtain the dynamic microphone volume level from current call Detailed Description Function Documentation int setVideoDeviceld int deviceld Set the video device that will use for video call Parameters deviceld Device ID index for video device camera Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoResolution VIDEO RESOLUTION resolution Set the video capture resolution Parameters resolution Video resolution defined in PortSIPType file Note Some cameras don t support SVGA and XVGA 720P please read your camera manual Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoBitrate int bitrateKbps Set the video bit rate Parameters bitrateKbps The video bit rate in KBPS Returns If the function s
13. 28 rejectRefer Refer functions 33 removeFromConference Conference functions 41 RTP and RTCP QOS functions 42 setAudioQos DSCPValue priority 43 setAudioRtcpBandwidth BitsRR BitsRS K BitsAS 42 set VideoQos DSCPValue 43 set VideoRtcpBandwidth BitsRR BitsRS K BitsAS 42 RTP callback events 67 onReceivedRTPPacket isAudio RTPPacket packetSize 67 onSendingRTPPacket isAudio RTPPacket p acketSize 67 RTP packets Audio stream and video stream callback functions 35 enableAudioStreamCallback enable callbac kMode 36 enableVideoStreamCallback callbackMode 36 setRtpCallback 35 RTP statistics functions 43 getAudioRtcpStatistics bytesSent packetsSe nt bytesReceived packetsReceived sendFrac tionLost sendCumulativeLost recvFraction Lost recvCumulativeLost 45 getAudioRtpStatistics averageJitterMs max JitterMs discardedPackets 44 getNetworkStatistics currentBufferSize pref erredBufferSize currentPacketLossRate curr entDiscardRate currentExpandRate current PreemptiveRate currentAccelerateRate 44 getVideoRtpStatistics bytesSent packetsSen t bytesReceived packetsReceived 45 SDK Callback events 55 SDK functions 8 Send audio and video stream functions 33 enableSendPcmStreamToRemote state strea mSamplesPerSec 34 enableSendVideoStreamToRemote state 34 sendPcmStreamToRemote data 34 send VideoStreamToRemote data width hei ght 35 Send OPTIONS INFO MESSAGE functions 47 sendInfo mimeType subMimeType info
14. Audio Play audio and video file to remoe functions 38 PortSIPSDK 72 PortSIPVideoRenderView 79 update VideoRenderFrame 79 Presence events 63 onPresenceOffline from 64 onPresenceOnline from stateText 64 onPresenceRecvSubscribe fromDisplayNa me from subject 64 Presence functions 49 presenceAcceptSubscribe 50 presenceOffline 50 presenceOnline statusText 50 presenceRejectSubscribe 50 presenceSubscribeContact subject 49 presenceAcceptSubscribe Presence functions 50 presenceOffline Presence functions 50 presenceOnline statusText Presence functions 50 presenceRejectSubscribe Presence functions 50 presenceSubscribeContact subject Presence functions 49 Record functions 37 startRecord recordFilePath recordFileName appendTimeStamp audioFileFormat audio RecordMode aviFileCodecType videoRecor dMode 37 stopRecord 38 Refer events 59 onACTVTransferFailure reason code 60 onACTVTransferSuccess 60 onReceivedRefer referld to from referSipM essage 59 onReferAccepted 60 onReferRejected reason code 60 onTransferRinging 60 onTransferTrying 60 Refer functions 31 acceptRefer referSignaling 33 attendedRefer replaceSessionId referTo 32 refer referTo 32 rejectRefer 33 refer referTo Refer functions 32 Register events 55 onRegisterFailure statusCode 56 onRegisterSuccess statusCode 56 registerServer retryTimes Initialize and register functions 10 rejectCall code Call functions
15. MESSAGE message to remote side 77 int presenceSubscribeContact subject Send a SUBSCRIBE message for presence to a contact int presenceAcceptSubscribe Accept the presence SUBSCRIBE request which received from contact int presenceRejectSubscribe Reject a presence SUBSCRIBE request which received from contact int presenceOnline statusText Send a NOTIFY message to contact to notify that presence status is online changed int presenceOffline Send a NOTIFY message to contact to notify that presence status is offline int getNumOfVideoCaptureDevices Gets the number of available capture devices int get VideoCaptureDeviceName uniqueld deviceName Gets the name of a specific video capture device given by an index int getNumOfRecordingDevices Gets the number of audio devices available for audio recording int getNumOfPlayoutDevices Gets the number of audio devices available for audio playout NSString getRecordingDeviceName Gets the name of a specific recording device given by an index NSString getPlayoutDeviceName Gets the name of a specific playout device given by an index int setSpeakerVolume Set the speaker volume level int getSpeakerVolume Gets the speaker volume level int setSystemOutputMute Mutes the speaker device completely in the OS BOOL getSystemOutputMute Retrieves the output device mute state in
16. The ACTV means Active For example A established the call with B A transfer B to C C accepted the refer call A received this event Parameters sessionld The session ID of the call void onACTVTransferFailure long sessionid reason char reason code int code When the refer call is fails this event will be triggered The ACTV means Active For example A established the call with B A transfer B to C C rejected this refer call A will received this event Parameters sessionld The session ID of the call reason The error reason code The error code Signaling events Functions void lt PortSIPEventDelegate gt onReceivedSignaling message void lt PortSIPEventDelegate gt onSendingSignaling message 61 Detailed Description Function Documentation void onReceivedSignaling long sessionid message char message This event will be triggered when received a SIP message Parameters sessionId The session ID of the call message The SIP message which is received void onSendingSignaling long sessionid message char message This event will be triggered when sent a SIP message Parameters sessionld The session ID of the call message The SIP message which is sent MWI events Functions void lt PortSIPEventDelegate gt o0nWaiting VoiceMessage urgentNewMessageCount urgentOl
17. The DTMF tone 11 The DTMF tone 12 The DTMF tone A 13 The DTMF tone B 14 The DTMF tone C 15 The DTMF tone D 16 The DTMF tone FLASH Parameters dtmfDuration The DTMF tone samples recommend 160 playDtmfTone Set to true the SDK play local DTMF tone sound during send DTMF Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Refer functions Functions int PortSIPSDK refer referTo Refer the currently call to another one int PortSIPSDK attendedRefer replaceSessionId referTo Make an attended refer long PortSIPSDK acceptRefer referSignaling Accept the REFER request a new call will be make if called this function usuall called after onReceivedRefer callback event int PortSIPSDK rejectRefer Reject the REFER request Detailed Description Function Documentation int refer long sessionid referTo NSString referTo Refer the currently call to another one Parameters sessionld The session ID of the call referTo Target of the refer it can be sip number sipserver com or number only Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks myVoIPSdk refer sessionId referTo sip testuserl2 sip portsip com You can download the demo AVI at http www portsip com downloads video bl
18. delegate self 8 More details please read the Sample project source code 4 How to test the P2P call without SIP server 1 Download and extract the SDK sample project zip file compile and run the P2PSample project 2 Run the P2Psample on two devices for example run it on device A and device B A IP address ie 192 163 ES B IP ailes ag 192 1168 iil 3 Enter a user name and password on A for example user name is 111 password is aaa you can enter anything for the password the SDK will ignore it Enter a user name and password on B for example user name is 222 password is aaa 4 Click the Initialize button on A and B If the default port 5060 in using the P2PSample will said Initialize failure In case please click the Uninitialize button and change the locale port Clack them i nikteaka lise Uut COn macgad me 5 The log box will appears Initialized if the SDK initialize succeeded 6 Make call from A to B enter sip 222 192 168 1 11 and click Dial button Make call from Bee AW T enter eip al el SVs Are 5 il 5 AL Note If changed the local sip port to other port for example the A using local port 5080 and the B using local port 6021 make call from A to By enter sip 2220192 168 1 11 6021 and daalk Makekcal rron B tor A enter sip lll Glo eG Smell Or 5 0 8 0 5 Does the SDK is thread safe Yes the SDK is thread safe you can call all the API functions don t need to consider
19. function fails the return value is a specific error code int stopPlayAudioFileToRemote long sessionld Stop play wave file to remote side Parameters sessionId Session ID of the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code 40 int play AudioFileToRemoteAsBackground long sessionid filename NSSiring filename fileSamplesPerSec int fileSamplesPerSec Play an wave file to remote party as conversation background sound Parameters sessionId Session ID of the call filename The file full path name such as test wav fileSamplesPerSec_ The wave file sample in seconds should be 8000 or 16000 or 32000 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int stopPlayAudioFileToRemoteAsBackground long sessionld Stop play an wave file to remote party as conversation background sound Parameters sessionId Session ID of the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Conference functions Functions 6 int PortSIPSDK createConference videoResolution displayLocal Video Create a conference It s failures if the exists conference isn t destroy yet void PortSIPSDK destroyConference Destroy the ex
20. image from camera int setRemoteVideoWindow long sessionid remoteVideoWindow PortSIPVideoRenderView remoteVideoWindow 26 Set the window for a session that using to display the received remote video image Parameters sessionId The session ID of the call remoteVideoWindo The PortSIPVideoRenderView to display received remote video image w Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int displayLocalVideo BOOL state Start stop to display the local video image Parameters state state Set to true to display local video iamge Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoNackStatus BOOL state Enable disable the NACK feature rfc6642 which help to improve the video quatliy Parameters state state Set to true to enable Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code void muteMicrophone BOOL mute Mute the device microphone it s unavailable for Android and iOS Parameters mute If the value is set to true the microphone is muted set to false to un mute it void muteSpeaker BOOL mute Mute the device speaker it s unavailable for Android and iOS Parameters mute If the val
21. int PortSIPSDK unHold Take off hold int PortSIPSDK muteSession muteIncomingAudio muteOutgoingAudio muteIncoming Video muteOutgo ingVideo Mute the specified session audio or video 28 int PortSIPSDK forwardCall forwardTo Forward call to another one when received the incoming call int PortSIPSDK sendDtmf dtmfMethod code dtmfDration playDtmfTone Send DTMF tone Detailed Description Function Documentation long call NSString callee sendSdp BOOL sendSdp videoCall BOOL videoCall Make a call Parameters callee The callee it can be name only or full SIP URI for example user001 or sip user001 sip iptel org or sip user002 sip yourdomain com 5068 sendSdp If set to false then the outgoing call doesn t include the SDP in INVITE message videoCall If set the true and at least one video codec was added then the outgoing call include the video codec into SDP Returns If the function succeeds the return value is the session ID of the call greater than 0 If the function fails the return value is a specific error code Note the function success just means the outgoing call is processing you need to detect the call final state in onInviteTrying onInviteRinging onInviteFailure callback events int rejectCall long sessionld code int code rejectCall Reject the incoming call Parameters sessionId The sessionId
22. parameters as the RFC3556 Parameters sessionId The session ID of call conversation BitsRR The bits for the RR parameter BitsRS The bits for the RS parameter KBitsAS The Kbits for the AS parameter Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoRtcpBandwidth long sessionid BitsRR int BitsRR BitsRS int BitsRS KBitsAS int KBitsAS Set the video RTCP bandwidth parameters as the RFC3556 Parameters sessionld The session ID of call conversation 43 BitsRR The bits for the RR parameter BitsRS The bits for the RS parameter KBitsAS The Kbits for the AS parameter Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setAudioQos BOOL enable DSCPValue int DSCPValue priority int priority Set the DSCP differentiated services code point value of QoS Quality of Service for audio channel Parameters enable Set to true to enable audio QoS DSCPValue The six bit DSCP value Valid range is 0 63 As defined in RFC 2472 the DSCP value is the high order 6 bits of the IP version 4 IPv4 TOS field and the IP version 6 IPv6 Traffic Class field priority The 802 1p priority PCP field in a 802 1Q VLAN tag Values 0 7 set the priority value 1 leaves the priority setting unchang
23. performance SDKs provide unified API layers for full user control and flexibility Changes in this release This release is a major upgrade see Release Notes_ for more information Getting Started You can download the PortSIP VoIP SDK Sample projects at our Website the samples include for VC C VB NET Delphi XE XCode for iOS and Mac OS Eclipse Java for Android the sample project source code is provided not include SDK source code The sample projects demonstrate how to create a SIP application base on our SDK powerful easy and quick Contents The download sample package contains almost all of PortSIP SDK documentation Dynamic Static libraries sources headers datasheet and everything else a SDK user might need SDK User Manual The starting point for the documentation of PortSIP VoIP SDK is the SDK User Manual page which gives a brief description of each API functions Web Site Some general interest or often changing PortSIP SDK information lives only on the PortSIP web site The release contains links to the site so while browsing it you ll see occasional broken links if you aren t connected to the Internet But everything needed to use the PortSIP VoIP SDK is contained within the release Background Read the Overview to help you understand what PortSIP is about and to help in educating your organization about PortSIP Support Please send email to Our support if you need any helps Machine Requi
24. return value is 0 If the function fails the return value is a specific error code Remarks MUST called this function first if want to send the PCM stream data to another side int sendPcmStreamToRemote long sessionid data NSData data Send the audio stream in PCM format from another source to instead of audio device capture microphone Parameters sessionId Session ID of the call conversation data The PCM audio stream data must is 16bit mono Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Usually we should use it like below myVoIPSdk enableSendPcmStreamToRemote sessionId state YES streamSamplesPerSec 16000 myVoIPSdk sendPcmStreamToRemote sessionId data data int enableSendVideoStreamToRemote long sessionid state BOOL state Enable the SDK send video stream data to remote side from another source to instead of camera Parameters sessionld The session ID of call 35 state Set to true to enable the send stream false to disable Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int sendVideoStreamToRemote long sessionid data NSData data width int width height int height Send the video stream to remote Parameters sessionld Session I
25. setRtcpPortRange maximumRtcpAudioPort minimumRtcp VideoPort maximumRtcpVi deoPort 17 setRtpKeepAlive keepAlivePayloadType de ltaTransmitTimeMS 19 setRtpPortRange maximumRtpAudioPort m inimumRtp VideoPort maximumRtpVideoP ort 17 setSrtpPolicy 17 addSupportedMimeT ype mimeType subMimeTy pe Additional setting functions 20 addVideoCodec Audio and video codecs functions 12 answerCall videoCall Call functions 29 attendedRefer replaceSessionId referTo Refer functions 32 Audio and video codecs functions 11 addAudioCodec 12 addVideoCodec 12 isAudioCodecEmpty 13 isVideoCodecEmpty 13 setAudioCodecParameter parameter 13 setAudioCodecPayloadType payloadType 13 setVideoCodecParameter parameter 14 setVideoCodecPayloadType payloadType 13 Audio and video functions 23 displayLocalVideo 26 getDynamic VolumeLevel microphone Volu me 27 muteMicrophone 26 muteSpeaker 26 send Video sendState 25 setAudioDeviceld outputDeviceld 27 setLocal Video Window 25 setRemote Video Window remote VideoWin dow 25 setVideoBitrate 24 setVideoDeviceld 24 setVideoFrameRate 25 setVideoNackStatus 26 setVideoOrientation 25 setVideoResolution 24 Audio and video stream callback events 68 onAudioRawCallback audioCallbackMode data dataLength samplingFreqHz 68 onVideoRawCallback videoCallbackMode width height data dataLength 68 Audio effect functions 46 enableAEC 46 enableAGC 47 enableANS 47
26. the multiple threads Note the SDK allows call API functions in callback events directly except the onAudioRawCallback onVideoRawCallback onReceivedRtpPacket onSendingRtpPacket Sexe 6 Does the SDK support native 64 bits Yes the SDK support 64 bits Module Index Modules Here is a list of all modules SDK functions ee et edt ie bi ees te Rete esl ee tl need e 9 Initialize and register functions sss sees eee eee 9 NIG and JocallP functions ees acetates Beta Ee neha 12 Audio and video Codecs TuncHons erea e a aeea ee e EEE EEEE E Ea a Eae ORE i E 12 Additional setting Tunetong sese cS easZ5 2592 rhe is Esie ENEE SRTR eaS TRT OD EET 15 Access SIP message header Tunctons esse sese eee 22 Audio and VideO UC sle 24 Call gt reel EE 28 Refer functions Z aa TZ saaTad eect ats haha Te TT a ket Ate T heat ed hee 32 Send audio and video stream functions sss sees eee 34 RTP packets Audio stream and video stream callback functions elect eee eee 36 REGOPGAUNCHONS sde pete E EO titans Mae E hie See ee 38 Play audio and video file to remoe functions sss sese esse ee eee eee 39 Conference ele e EE 41 RTP and RTCP QOS functions ceeccecccssececssccecessececseacececeeecsesaececseaeeeeeseeeesesaecesseeeeensseeseneeas 43 RTP statistics TT TTT 44 Audio effect TUnctions eee eiee a ER En EE AOE R OPEREN ESE ATEENA EEEE EE s 47 Send OPTIONS INFO MESSAGE functions esse ee ee eee eee 48 Presence Tuncpong a r a E R E OE E E E E
27. true means this call include the audio exists Video If it s true means this call include the video void onInviteRinging long sessionid statusText char statusText statusCode int statusCode optional If the out going call was ringing this event triggered Parameters sessionld The session ID of the call statusText The status text status Code The status code void onInviteAnswered long sessionid callerDisplayName char callerDisplayName caller char caller calleeDisplayName char calleeDisplayName callee char callee audioCodecs char audioCodecs videoCodecs char videoCodecs existsAudio BOOL existsAudio existsVideo BOOL existsVideo optional If the remote party was answered the call this event triggered Parameters sessionld The session ID of the call 58 callerDisplayNam_ The display name of caller e caller The caller calleeDisplayNam_ The display name of callee e callee The callee audioCodecs The matched audio codecs it s separated by if have more than one codec videoCodecs The matched video codecs it s separated by if have more than one codec existsAudio If it s true means this call include the audio exists Video If it s true means this call include the video void onlnviteFailure long sessionid reason char reason code int code optional If the outgoin
28. value is 0 If the function fails the return value is a specific error code int getVideoRtpStatistics long sessionld bytesSent int bytesSent packetsSent int packetsSent bytesReceived int bytesReceived packetsReceived int packetsReceived Obtain the RTP statisics of video Parameters sessionId The session ID of call conversation bytesSent The number of sent bytes packetsSent The number of sent packets bytesReceived The number of received bytes packetsReceived The number of received packets Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code 46 Audio effect functions Functions void PortSIPSDK enableVAD Enable disable Voice Activity Detection VAD void PortSIPSDK enableAEC Enable disable AEC Acoustic Echo Cancellation void PortSIPSDK enableCNG Enable disable Comfort Noise Generator CNG void PortSIPSDK enableAGC Enable disable Automatic Gain Control AGC void PortSIPSDK enableANS Enable disable Audio Noise Suppression ANS Detailed Description Function Documentation void enableVAD BOOL state Enable disable Voice Activity Detection VAD Parameters state Set to true to enable VAD false to disable void enableAEC EC_MODES state Enable disable AEC Acoustic Echo Cancellation Para
29. 0 255 If the function fails the return value is a specific error code int setSystemOutputMute BOOL enable Mutes the speaker device completely in the OS 54 Parameters enable If set to true the device output is muted If set to false the output is unmuted Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code BOOL getSystemOutputMute Retrieves the output device mute state in the operating system Returns If return value is true the output device is muted If false the output device is not muted int setMicVolume int volume Sets the microphone volume level Parameters volume The microphone volume level the valid value is 0 255 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getMicVolume Retrieves the current microphone volume Returns If the function succeeds the return value is the microphone volume If the function fails the return value is a specific error code int setSystemInputMute BOOL enable Mute the microphone input device completely in the OS Parameters enable If set to true the input device is muted Set to false is unmuted Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code BOOL getSystemInput
30. 1 createConference videoResolution displayLocal Video Conference functions 41 detectMwi Additional setting functions 19 Device Manage functions 51 audioPlayLoopbackTest 55 getMicVolume 54 getNumOfPlayoutDevices 52 getNumOfRecordingDevices 52 getNumOfVideoCaptureDevices 52 getPlayoutDeviceName 53 getRecordingDeviceName 53 getSpeakerVolume 53 getSystemInputMute 54 getSystemOutputMute 54 getVideoCaptureDeviceName uniqueld dev iceName 52 setMicVolume 54 setSpeakerVolume 53 setSystemInputMute 54 setSystemOutputMute 53 disableCallForward Additional setting functions 18 disableSessionTimer Additional setting functions 19 displayLocalVideo Audio and video functions 26 DTMF events 62 onRecvDtmfTone tone 62 enable3GppTags Additional setting functions 16 enableAEC Audio effect functions 46 enableAGC Audio effect functions 47 enableANS Audio effect functions 47 enableAudioStreamCallback enable callbackMo de RTP packets Audio stream and video stream callback functions 36 enableCallbackSendingSignaling Additional setting functions 16 enableCallForward forwardTo Additional setting functions 18 enableCheckMwi Additional setting functions 19 enableCNG Audio effect functions 46 enableReliableProvisional Additional setting functions 16 enableSendPcmStreamToRemote state streamSa mplesPerSec Send audio and video stream functions 34 enableSend VideoStreamToRemote state
31. Con tents 48 sendMessage mimeType subMimeType me ssage messageLength 48 sendOptions sdp 47 sendOutOfDialogMessage mimeType subM imeType message messageLength 48 86 sendDtmf dtmfMethod code dtmfDration playDt mfTone Call functions 31 sendInfo mimeType subMimeType infoContents Send OPTIONS INFO MESSAGE functions 48 sendMessage mimeType subMimeType message messageLength Send OPTIONS INFO MESSAGE functions 48 sendOptions sdp Send OPTIONS INFO MESSAGE functions 47 sendOutOfDialogMessage mimeType subMime Type message messageLength Send OPTIONS INFO MESSAGE functions 48 sendPcmStreamToRemote data Send audio and video stream functions 34 send Video sendState Audio and video functions 25 send VideoStreamToRemote data width height Send audio and video stream functions 35 setAudioCodecParameter parameter Audio and video codecs functions 13 setAudioCodecPayloadType payloadType Audio and video codecs functions 13 setAudioDeviceld outputDeviceld Audio and video functions 27 setAudioQos DSCPValue priority RTP and RTCP QOS functions 43 setAudioRtcpBandwidth BitsRR BitsRS KBitsA S RTP and RTCP QOS functions 42 setAudioSamples maxPtime Additional setting functions 20 setConference Video Window Conference functions 41 setDisplayName Additional setting functions 15 setDoNotDisturb Additional setting functions 19 setKeepAliveTime Additional setting functions 20 setLicense
32. D of the call conversation data The video video stream data must is 1420 format width The video image width height The video image height Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Send the video stream in i420 from another source to instead of video device capture camera Before called this function you MUST call the enableSend VideoStreamToRemote function Usually we should use it like below myVoIPSdk enableSendVideoStreamToRemote sessionId state YES myVoIPSdk sendVideoStreamToRemote sessionId data dara width 352 height 288 RTP packets Audio stream and video stream callback functions Functions int PortSIPSDK setRtpCallback Set the RTP callbacks to allow access the sending and received RTP packets int PortSIPSDK enableAudioStreamCallback enable callbackMode Enable disable the audio stream callback int PortSIPSDK enable VideoStreamCallback callbackMode Enable disable the video stream callback Detailed Description Function Documentation int setRtpCallback BOOL enable 36 Set the RTP callbacks to allow access the sending and received RTP packets Parameters enable Set to true to enable the RTP callback for received and sending RTP packets the onSendingRtpPacket and onReceivedRtpPacket events will be triggered Returns If the function succeeds the return value is 0
33. EAM_LOCAL VIDEOSTREAM_REMOTE Local video stream callback Remote video stream callback 37 VIDEOSTREAM_ BOTH Both of local and remote video stream callback Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks the onVideoRawCallback event will be triggered if the callback is enabled Record functions Functions int PortSIPSDK startRecord recordFilePath recordFileName appendTimeStamp audioFileFormat audioR ecordMode aviFileCodecType videoRecordMode Start record the call int PortSIPSDK stopRecord Stop record Detailed Description Function Documentation int startRecord long sessionid recordFilePath NSString recordFilePath recordFileName NSString recordFileName appendTimeStamp BOOL appendTimeStamp audioFileFormat AUDIO_FILE_FORMAT audioFileFormat audioRecordMode RECORD_MODE audioRecordMode aviFileCodecType VIDEOCODEC_TYPE aviFileCodecType videoRecordMode RECORD_MODE videoRecordMode Start record the call Parameters sessionId The session ID of call conversation recordFilePath The file path to save record file it s must exists recordFileName The file name of record file for example audiorecord wav or videorecord avi appendTimeStamp Set to true to append the timestamp to the recording file name audioFileFormat The audio record file format
34. Key Initialize and register functions 10 setLocalVideoWindow Audio and video functions 25 setMicVolume Device Manage functions 54 setRemote VideoWindow remote Video Window Audio and video functions 25 setRtcpPortRange maximumRtcpAudioPort mini mumRtcpVideoPort maximumRtcp VideoPort Additional setting functions 17 setRtpCallback RTP packets Audio stream and video stream callback functions 35 setRtpKeepAlive keepAlivePayloadType deltaTr ansmitTimeMS Additional setting functions 19 setRtpPortRange maximumRtpAudioPort minim umRtpVideoPort maximumRtp VideoPort Additional setting functions 17 setSpeaker Volume Device Manage functions 53 setSrtpPolicy Additional setting functions 17 setS ystemInputMute Device Manage functions 54 Set ystemOutputMute Device Manage functions 53 setUser displayName authName password locall P localSIPPort userDomain SIPServer SIPServer Port STUNServer STUNServerPort outboundSer ver outboundServerPort Initialize and register functions 9 setVideoBitrate Audio and video functions 24 setVideoCodecParameter parameter Audio and video codecs functions 14 setVideoCodecPayloadType payloadT ype Audio and video codecs functions 13 setVideoDeviceld Audio and video functions 24 setVideoFrameRate Audio and video functions 25 setVideoNackStatus Audio and video functions 26 setVideoOrientation Audio and video functions 25 setVideoQos DSCPValue RTP and RTC
35. Mute Gets the mute state of the input device in the operating system 55 Returns If return value is true the input device is muted If false the input device is not muted void audioPlayLoopbackTest BOOL enable Use to do the audio device loop back test Parameters enable Set to true start audio look back test Set to fase to stop SDK Callback events Modules Register events Call events Refer events Signaling events MWI events DTMF events INFO OPTIONS message events Presence events MESSAGE message events Play audio and video file finished events RTP callback events Audio and video stream callback events Detailed Description SDK Callback events Register events Functions void lt PortSIPEventDelegate gt onRegisterSuccess statusCode void lt PortSIPEventDelegate gt onRegisterFailure statusCode Detailed Description Register events 56 Function Documentation void onRegisterSuccess char statusText statusCode int statusCode When successfully register to server this event will be triggered Parameters statusText The status text status Code The status code void onRegisterFailure char statusText statusCode int statusCode If register to SIP server is fail this event will be triggered Parameters statusText The status text status Code The status code Call events Functi
36. P QOS functions 43 setVideoResolution Audio and video functions 24 setVideoRtcpBandwidth BitsRR BitsRS KBitsA S RTP and RTCP QOS functions 42 Signaling events 60 onReceivedSignaling message 61 onSendingSignaling message 61 startRecord recordFilePath recordFileName appe ndTimeStamp audioFileFormat audioRecordMo de aviFileCodecType videoRecordMode Record functions 37 stopPlayAudioFileToRemote Play audio and video file to remoe functions 39 stopPlayAudioFileToRemoteAsBackground Play audio and video file to remoe functions 40 87 stopPlay VideoFileToRemote Play audio and video file to remoe functions 39 stopRecord Record functions 38 unHold Call functions 30 unRe gisterServer Initialize and register functions 10 updateCall enableAudio enable Video Call functions 29 update VideoRenderFrame PortSIPVideoRenderView 79 88
37. PortSIP VoIP SDK Manual for Mac Version 11 2 2 4 4 2015 Table of Contents Welcome to the PortSIP VoIP SDK sese eee 3 Module index TTT 6 Hire rare hi Gal ne TTT 7 LS ASS LINER TTT 8 Module Documentation T 9 KIIRCH 9 Initialize and register FUNCTIONS oo eee eee eeecetecesecnseceeecneecaeeeeeeeaeeeeeesecsecssecsaecsaecsaecseecseeeaeseaeeeaeeatens 9 NIC and local IP functions 20 0 cece ce cecececececececececesececeeececececacececscasecacasacasassseuaesseesesuseseeeseanaes 12 Audio and video Codecs FUNCTIONS eseis ccc ceceesesssesseseescssssssssscseeesesesssesesesesesesesesesesssessseseseseseseseeseseeees 12 Additional SettinS Det eng shese erneten eere eeoa E esheets ap e ye ee SEEE HOOKER dyd eerste 15 Access SIP message header functions eeceeeesecesecesecsseceecaeecseseneeeeeeeeeeseeeaeeesecssecsaecsaecsaesaaeeneeeaes 22 Alidio and vid o E Te e TTT 24 CallsfinCt Ons TTT 28 Refer RST La 32 Send audio and video stream functions sss sese sese 34 RTP packets Audio stream and video stream callback functions sss eee cee eee eee 36 Record Ele Te 38 Play audio and video file to remoe functions 0 0 eect eee eee 39 Conference FUNCTIONS see 41 RTP and RECEP QOS TunctiOns s sda shee ech aR ZES 2 TT te eds Taar a aaa T 2 aR a REE A T itn 43 RTP statistics functions see 44 ERT eE G BN 01810 TT 47 Send OPTIONS INFO MESSAGE functions sss eee e eee 48 Presence MUINCHIONS e caccssdeceececeecedcaneececs coeeoecsasuececdesGectcedacecc
38. QoS Quality of Service for video channel int getNetworkStatistics currentBufferSize preferredBufferSize currentPacketLossRate currentDiscardRat e currentExpandRate currentPreemptiveRate currentAccelerateRate Get the in call statistics The statistics are reset after the query int getAudioRtpStatistics averageJitterMs maxJitterMs discardedPackets Obtain the RTP statisics of audio channel int getAudioRtcpStatistics bytesSent packetsSent bytesReceived packetsReceived sendFractionLost send CumulativeLost recvFractionLost recvCumulativeLost Obtain the RTCP statisics of audio channel int get VideoRtpStatistics bytesSent packetsSent bytesReceived packetsReceived Obtain the RTP statisics of video void enableVAD Enable disable Voice Activity Detection VAD void enableAEC Enable disable AEC Acoustic Echo Cancellation void enableCNG Enable disable Comfort Noise Generator CNG void enableAGC Enable disable Automatic Gain Control AGC void enableANS Enable disable Audio Noise Suppression ANS int sendOptions sdp Send OPTIONS message int sendInfo mimeType subMimeType infoContents Send a INFO message to remote side in dialog long sendMessage mimeType subMimeType message messageLength Send a MESSAGE message to remote side in dialog long sendOutOfDialogMessage mimeT ype subMimeT ype message messageLength Send a out of dialog
39. Server STUNServerPort int stunServerPort outboundServer NSString outboundServer outboundServerPort int outboundServerPort Set user account info Parameters userName Account User name of the SIP usually provided by an IP Telephony service provider displayName The display name of user you can set it as your like such as James Kend It s optional authName Authorization user name usually equals the username password The password of user it s optional localIP The local computer IP address to bind for example 192 168 1 108 it will be 10 using for send and receive SIP message and RTP packet If pass this IP as the IPv6 format then the SDK using IPv6 If you want the SDK choose correct network interface IP automatically please pass the 0 0 0 0 for IPv4 or for IPv6 localSIP Port The SIP message transport listener port for example 5060 userDomain User domain this parameter is optional that allow pass a empty string if you are not use domain sipServer SIP proxy server IP or domain for example XX XXX XX X Or Sip XXX com sipServerPort Port of the SIP proxy server for example 5060 stunServer Stun server use for NAT traversal it s optional and can be pass empty string to disable STUN stunServerPort STUN server port it will be ignored if the outboundServer is empty outboundServer Outbound proxy server for example sip domain com i
40. Without this keepalive mechanism proxies that remember incoming and outgoing requests stateful proxies may continue to retain call state needlessly If a UA fails to send a BYE message at the end of a session or if the BYE message is lost because of network problems a stateful proxy does not know that the session has 19 ended The re INVITES ensure that active sessions stay active and completed sessions are terminated int disableSessionTimer Disable the session timer Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code void setDoNotDisturb BOOL state Enable the Do not disturb to enable disable Parameters state If set to true the SDK reject all incoming calls anyway int detectMwi Use to obtain the MWI status Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int enableCheckMwi BOOL state Allows enable disable the check MWI Message Waiting Indication Parameters state If set as true will check MWI automatically once successfully registered to a SIP proxy server Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setRtpKeepAlive BOOL state keepAlivePayloadType int keepAlivePayloadType deltaTransmitTimeMS int deltaTransmitTimeMS Ena
41. _LOG_LEVEL logLevel logPath NSString logFilePath maxLine int maxCallLines agent NSString sipAgent audioDeviceLayer int audioDeviceLayer videoDeviceLayer int videoDeviceLayer Initialize the SDK Parameters transport Transport for SIP signaling TRANSPORT_PERS is the PortSIP private transport for anti the SIP blocking it must using with the PERS logLevel Set the application log level the SDK generate the PortSIP_Log_datatime log file if the log enabled logFilePath The log file path the path folder MUST is exists maxCallLines In theory support unlimited lines just depends on the device capability for SIP client recommend less than 1 100 sipAgent The User Agent header to insert in SIP messages audioDeviceLayer 0 Use OS default device 1 Set to 1 to use the virtual audio device if the no sound device installed videoDeviceLayer 0 Use OS default device 1 Set to 1 to use the virtual video device if no camera installed Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setUser NSString userName displayName NSString displayName authName NSString authName password NSString password locallP NSString lIocallP localSIPPort int localS PPort userDomain NSString userDomain SIPServer NSString sipServer SiIPServerPort int sipServerPort STUNServer NSString stun
42. a S 50 Device Manage functions sss zr etes sorsien i iiser te iier nee oar S EEr a a E EE EE EE EEEE Er ar 52 SDK Callback events r a E R E E r E E EE E 56 Register events niire a ar e a a bees are a ES E EE EE EE E REES 56 UR EE 57 E EE 60 e 61 MWe Vents ee ee nd ne ieee a ne tee ae 62 EN KAA i ss 00 ant e Ate ee Aree ee ee eee A eee 63 INFO OPTIONS message eVemts n esere roeiers roert erae EEERSE ashes se Seve svendsceasdetbpescesscehsetotsnasogessess 64 PHESENGCE EVENTS ice fi A E E Re si I ee cis 64 MESSAGE message event e ssccsscerssisdes cbegseeseksdeeen shoes Ere omr E TE rE EEE E stay E EEr rE 65 Play audio and video file finished events 67 RAR EAI DET EAE THT 68 Audio and video stream callback events sse see eee 69 Hierarchical Index Class Hierarchy This inheritance list is sorted roughly but not completely alphabetically lt NSObject gt lt PortSIPEventDelegate gt n tieien e A parea R S E e E E O S AARSE eari PortSIPSDRK ene ee T A E T eee EE set pee NSView Port SP Vide Rend S TTT Class Index Class List Here are the classes structs unions and interfaces with brief descriptions lt PortSIPEventDelegate gt PortSIP SDK Callback events Delegate PortSIPSDK PortSIP VoIP SDK functions class PortSIPVideoRender View PortSIP VoIP SDK Video Render View class Module Documentation SDK functions Modules Initialize and register functions NIC and local IP functions Audio and video code
43. a specific error code NIC and local IP functions Functions int PortSIPSDK getNICNums Get the Network Interface Card numbers NSString PortSIPSDK getLocallpAddress Get the local IP address by Network Interface Card index Detailed Description Function Documentation int getNICNums Get the Network Interface Card numbers Returns If the function succeeds the return value is NIC numbers gt 0 If the function fails the return value is a specific error code NSString getLocallpAddress int index Get the local IP address by Network Interface Card index Parameters The IP address index for example the PC has two NICs we want to obtain the second NIC IP then set this parameter 1 The first NIC IP index is 0 Returns The buffer that to receives the IP Audio and video codecs functions Functions int PortSIPSDK addAudioCodec 12 Enable an audio codec it will be appears in SDP int PortSIPSDK addVideoCodec Enable a video codec it will be appears in SDP 6 BOOL PortSIPSDK isAudioCodecEmpty Detect enabled audio codecs is empty or not BOOL PortSIPSDK is VideoCodecEmpty Detect enabled video codecs is empty or not int PortSIPSDK setAudioCodecPayloadType payloadType Set the RTP payload type for dynamic audio codec 6 int PortSIPSDK setVideoCodecPayloadT ype payloadType Set the RTP payload type for dynamic Video codec
44. ble or disable send RTP keep alive packet during the call is established Parameters state Set to true allow send the keep alive packet during the conversation keepAlivePayload The payload type of the keep alive RTP packet usually set to 126 20 Type MS 15000 300000 deltaTransmitTime The keep alive RTP packet send interval in millisecond usually recommend Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setKeepAliveTime int keepAliveTime Enable or disable send SIP keep alive packet Parameters keepAliveTime This is the SIP keep alive time interval in seconds set to 0 to disable the SIP keep alive it s in seconds recommend 30 or 50 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setAudioSamples int ptime maxPtime int maxPtime Set the audio capture sample Parameters ptime It s should be a multiple of 10 and between 10 60 included 10 and 60 maxPtime For the maxptime attribute should be a multiple of 10 and between 10 60 included 10 and 60 Can t less than ptime Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks which will be appears in the SDP of INVITE and 200 OK message as
45. cardedPackets int discardedPackets Obtain the RTP statisics of audio channel Parameters sessionld The session ID of call conversation 45 averageJitterMs Short time average jitter in milliseconds maxJitterMs Maximum short time jitter in milliseconds discardedPackets The number of discarded packets on a channel during the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getAudioRtcpStatistics long sessionld bytesSent int bytesSent packetsSent int packetsSent bytesReceived int bytesReceived packetsReceived int packetsReceived sendFractionLost int sendFractionLost sendCumulativeLost int sendCumulativeLost recvFractionLost int recvFractionLost recvCumulativeLost int recvCumulativeLost Obtain the RTCP statisics of audio channel Parameters sessionId The session ID of call conversation bytesSent The number of sent bytes packetsSent The number of sent packets bytesReceived The number of received bytes packetsReceived The number of received packets sendFractionLost Fraction of sent lost in percent sendCumulativeLo The number of sent cumulative lost packet st recvFractionLost Fraction of received lost in percent recvCumulativeLo The number of received cumulative lost packets st Returns If the function succeeds the return
46. cs functions Additional setting functions Access SIP message header functions Audio and video functions Call functions Refer functions Send audio and video stream functions RTP packets Audio stream and video stream callback functions Record functions Play audio and video file to remoe functions Conference functions RTP and RTCP QOS functions RTP statistics functions Audio effect functions Send OPTIONS INFO MESSAGE functions Presence functions Device Manage functions Detailed Description SDK functions Initialize and register functions Functions int PortSIPSDK initialize loglevel logPath maxLine agent audioDeviceLayer videoDeviceLayer Initialize the SDK void PortSIPSDK unInitialize Un initialize the SDK and release resources int PortSIPSDK setUser displayName authName password localIP localSIPPort userDomain SIPServer SIPServerPort STUNServer STUNServerPort outboundServer outboundServerPort Set user account info int PortSIPSDK registerServer retryTimes Register to SIP proxy server login to server int PortSIPSDK unRegisterServer Un register from the SIP proxy server int PortSIPSDK setLicenseKey Set the license key must called before setUser function Detailed Description Initialize and register functions Function Documentation int initialize TRANSPORT_TYPE transport loglevel PORTSIP
47. dMessageCou nt newMessageCount oldMessageCount void lt PortSIPEventDelegate gt onWaitingFax Message urgentNewMessageCount urgentOldMessageCount newMessageCount oldMessageCount Detailed Description Function Documentation void onWaitingVoiceMessage char messageAccount urgentNewMessageCount int urgentNewMessageCount urgentOldMessageCount int urgentOldMessageCount newMessageCount int newMessageCount oldMessageCount int oldMessageCount If has the waiting voice message MWI then this event will be triggered Parameters messageAccount Voice message account urgentNewMessag Urgent new message count 62 void onWaitingFaxMessage char messageAccount urgentNewMessageCount int urgentNewMessageCount urgentOldMessageCount int urgentOldMessageCount newMessageCount int newMessageCount oldMessageCount int oldMessageCount eCount urgentOldMessage Urgent old message count Count newMessageCount New message count oldMessageCount Old message count If has the waiting fax message MWD then this event will be triggered Parameters messageAccount Fax message account urgentNewMessag Urgent new message count eCount urgentOldMessage Urgent old message count Count newMessageCount New message count oldMessageCount Old message count DTMF events Functions void lt PortSIPEventDelegate gt onRecvDtm
48. deo file to remote side int playAudioFileToRemote filename fileSamplesPerSec loop Play an wave file to remote party int stopPlayAudioFileToRemote Stop play wave file to remote side int playAudioFileToRemoteAsBackground filename fileSamplesPerSec Play an wave file to remote party as conversation background sound int stopPlayAudioFileToRemoteAsBackground Stop play an wave file to remote party as conversation background sound int createConference videoResolution displayLocal Video 76 Create a conference It s failures if the exists conference isn t destroy yet void destroyConference Destroy the exist conference int setConference Video Window Set the window for a conference that using to display the received remote video image int joinToConference Join a session into exist conference if the call is in hold please un hold first int removeFromConference Remove a session from an exist conference int setAudioRtcpBandwidth BitsRR BitsRS KBitsAS Set the audio RTCP bandwidth parameters as the RFC3556 int setVideoRtcpBandwidth BitsRR BitsRS KBitsAS Set the video RTCP bandwidth parameters as the RFC3556 int setAudioQos DSCPValue priority Set the DSCP differentiated services code point value of QoS Quality of Service for audio channel int setVideoQos DSCP Value Set the DSCP differentiated services code point value of
49. e int sendInfo long sessionid mimeType NSString mimeType subMimeType NSString subMimeType infoContents NSString infoContents Send a INFO message to remote side in dialog Parameters sessionId The session ID of call mimeType The mime type of INFO message subMimeType The sub mime type of INFO message infoContents The contents that send with INFO message Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code long sendMessage long sessionid mimeType NSString mimeType subMimeType NSString subMimeType message NSData message messageLength int messageLength Send a MESSAGE message to remote side in dialog Parameters sessionId The session ID of the call mimeType The mime type of MESSAGE message subMimeType The sub mime type of MESSAGE message message The contents which send with MESSAGE message allow binary data messageLength The message size Returns If the function succeeds the return value is a message ID allows track the message send state in onSendMessageSuccess and onSendMessageFailure If the function fails the return value is a specific error code less than 0 Remarks Example 1 send a plain text message Note to send other languages text please use the UTF8 to encode the message before send myVoIPsdk sendMessage sessionId mimeType text subMimeTyp
50. e MESSAGE message events 65 onSendMessageSuccess messageld MESSAGE message events 65 onSendOutOfDialogMessageFailure fromDispla yName from toDisplayName to reason code MESSAGE message events 66 onSendOutOfDialogMessageSuccess fromDispla yName from toDisplayName to MESSAGE message events 66 onTransferRinging Refer events 60 onTransferTrying Refer events 60 onVideoRawCallback videoCallbackMode width sheight data dataLength Audio and video stream callback events 68 onWaitingFaxMessage urgentNewMessageCoun t urgentOldMessageCount newMessageCount ol dMessageCount MWI events 62 onWaiting VoiceMessage urgentNewMessageCo unt urgentOldMessageCount newMessageCount oldMessageCount MWI events 61 Play audio and video file finished events 66 onPlayAudioFileFinished fileName 66 onPlayVideoFileFinished 67 Play audio and video file to remoe functions 38 playAudioFileToRemote filename fileSamp lesPerSec loop 39 playAudioFileToRemoteAsBackground file name fileSamplesPerSec 40 playVideoFileToRemote aviFile loop playA udio 38 stopPlayAudioFileToRemote 39 stopPlayAudioFileToRemoteAsBackground 40 stopPlay VideoFileToRemote 39 playAudioFileToRemote filename fileSamplesPe rSec loop Play audio and video file to remoe functions 39 play AudioFileToRemoteAsBackground filename fileSamplesPerSec Play audio and video file to remoe functions 40 85 play VideoFileToRemote aviFile loop play
51. e plain message data messageLength dataLen Example 2 send a binary message myVoIPsdk sendMessage sessionId mimeType application subMimeType vnd 3gpp sms message data messageLength dataLen long sendOutOfDialogMessage NSSiring to mimeType NSSiring mimeType subMimeType NSSiring subMimeType message NSData message messageLength int messageLength Send a out of dialog MESSAGE message to remote side 49 Parameters to The message receiver Likes sip receiver portsip com mimeType The mime type of MESSAGE message subMimeType The sub mime type of MESSAGE message message The contents which send with MESSAGE message allow binary data messageLength The message size Returns If the function succeeds the return value is a message ID allows track the message send state in onSendOutOfMessageSuccess and onSendOutOfMessageFailure If the function fails the return value is a specific error code less than 0 Remarks Example 1 send a plain text message Note to send other languages text please use the UTF8 to encode the message before send myVoIPsdk sendOutOfDialogMessage sip userl sip portsip com mimeType text subMimeType plain message data messageLength dataLen Example 2 send a binary message myVoIPsdk sendOutOfDialogMessage sip userl sip portsip com mimeType application subMimeType vnd 3gpp sms message data messageLength dataLen Presenc
52. e functions Functions int PortSIPSDK presenceSubscribeContact subject Send a SUBSCRIBE message for presence to a contact int PortSIPSDK presenceAcceptSubscribe Accept the presence SUBSCRIBE request which received from contact int PortSIPSDK presenceRejectSubscribe Reject a presence SUBSCRIBE request which received from contact int PortSIPSDK presenceOnline statusText Send a NOTIFY message to contact to notify that presence status is online changed int PortSIPSDK presenceOffline Send a NOTIFY message to contact to notify that presence status is offline Detailed Description Function Documentation int presenceSubscribeContact NSString contact subject NSString subject Send a SUBSCRIBE message for presence to a contact 50 Parameters contact The target contact it must likes sip contact001 sip portsip com subject This subject text will be insert into the SUBSCRIBE message For example Hello I m Jason The subject maybe is UTF8 format you should use UTF8 to decode it Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int presenceAcceptSubscribe long subscribeld Accept the presence SUBSCRIBE request which received from contact Parameters subscribeld Subscribe id when received a SUBSCRIBE request from contact the event onPresenceRecvSub
53. ed Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoQos BOOL enable DSCPValue int DSCPValue Set the DSCP differentiated services code point value of QoS Quality of Service for video channel Parameters enable Set as true to enable QoS false to disable DSCPValue The six bit DSCP value Valid range is 0 63 As defined in RFC 2472 the DSCP value is the high order 6 bits of the IP version 4 IPv4 TOS field and the IP version 6 IPv6 Traffic Class field Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code RTP statistics functions Functions int PortSIPSDK getNetworkStatistics currentB ufferSize preferredBufferSize currentPacketLossRate curr entDiscardRate currentExpandRate currentPreemptiveRate currentAccelerateRate Get the in call statistics The statistics are reset after the query int PortSIPSDK getAudioRtpStatistics averageJitterMs maxJitterMs discardedPackets Obtain the RTP statisics of audio channel 44 int PortSIPSDK getAudioRtcpStatistics bytesSent packetsSent bytesReceived packetsReceived sendFract ionLost sendCumulativeLost recvFractionLost recvCumulativeLost Obtain the RTCP statisics of audio channel int PortSIPSDK getVideoRtpStatistics bytesSent packetsSent bytesReceived pac
54. ed once remote side close the call Parameters sessionld The session ID of the call 59 void onRemoteHold long sessionld optional If the remote side has placed the call on hold this event triggered Parameters sessionld The session ID of the call void onRemoteUnHold long sessionid audioCodecs char audioCodecs videoCodecs char videoCodecs existsAudio BOOL existsAudio existsVideo BOOL existsVideo optional If the remote side was un hold the call this event triggered Parameters sessionId The session ID of the call audioCodecs The matched audio codecs it s separated by if have more than one codec videoCodecs The matched video codecs it s separated by if have more than one codec existsAudio If it s true means this call include the audio exists Video If it s true means this call include the video Refer events Functions void lt PortSIPEventDelegate gt onReceivedRefer referld to from referSipMessage void lt PortSIPEventDelegate gt onReferAccepted void lt PortSIPEventDelegate gt onReferRejected reason code void lt PortSIPEventDelegate gt onTransferTrying void lt PortSIPEventDelegate gt onTransferRinging void lt PortSIPEventDelegate gt onACTVTransferSuccess void lt PortSIPEventDelegate gt onACTVTransferFailure reason code Detailed De
55. eeaseecececseacesseedececelsdecceveseeceaessescsedasGecsds 50 Device Manage Tune Dm si EE RAI ES AG RO ERT AS altos 52 KIRICH TEE 56 Resister events c2s0 5 Asari E Meal ie ella Glin eee A aes EE te alas 56 Calle EE 57 Refer EVENS H SEENEN Sane aS aaa EE 60 Signaling EE 61 MW Gvents Z Eaa aT cae ests Saeed A aT Sas cheek te Rape enone See Pe ee A 2 62 RR IS EE 63 INFO OPTIONS message event 64 REENEN See Eed ee EE ta ohn ln cee atte S E e azg 64 MESSAGE message events ee deer EN 65 Play audio and video file finished events 67 ARTE 68 Audio and video stream Callback events 69 Class DOCUMENtAt oh T 71 lt PortSIPEVentDele cater a c si sc cos cobs casctess csseti sees conseeeh E celevsvsash suis ob la See Eed E sees 71 POTMSIPRS E Aa E E A E T EEE AE EE E S A S 73 PortSIPVi deo R 010 21 gh E V ee a a a a 80 rte EE 82 Welcome to the PortSIP VoIP SDK Create your SIP based application for multiple platforms iOS Android Windows Mac OS Linux base on our SDK The award winning PortSIP VoIP SDK is a powerful and highly versatile set of tools to dramatically accelerate SIP application development It includes a suite of stacks SDKs Sample projects Each one enables developers to combine all the necessary components to create an ideal development environment for every application s specific needs The PortSIP VoIP SDK complies with IETF and 3GPP standards and is IMS compliant 3GPP 3GPP2 TISPAN and PacketCable 2 0 These high
56. enableCNG 46 enable VAD 46 audioPlayLoopbackTest Device Manage functions 55 Call events 56 onInviteAnswered callerDisplayName calle r calleeDisplayName callee audioCodecs vi deoCodecs exists Audio exists Video 57 onInviteBeginingForward 58 onInviteClosed 58 onInviteConnected 58 onInviteFailure reason code 58 onInviteIncoming callerDisplayName caller calleeDisplayName callee audioCodecs vid eoCodecs existsAudio exists Video 56 onInviteRinging statusText statusCode 57 onInviteSessionProgress audioCodecs video Codecs existsEarly Media existsAudio exist sVideo 57 82 onInviteTrying 57 onInviteUpdated audioCodecs videoCodecs existsAudio exists Video 58 onRemoteHold 59 onRemoteUnHold audioCodecs videoCode cs existsAudio exists Video 59 Call functions 27 answerCall videoCall 29 call sendSdp videoCall 28 forwardCall forwardTo 30 hangUp 28 hold 29 muteSession muteIncomingAudio muteOut going Audio muteIncoming Video muteOutg oingVideo 30 rejectCall code 28 sendDtmf dtmfMethod code dtmfDration pl ayDtmfTone 31 unHold 30 updateCall enableAudio enableVideo 29 call sendSdp videoCall Call functions 28 clearAddExtensionHeaders Access SIP message header functions 22 clearModifyHeaders Access SIP message header functions 23 Conference functions 40 createConference videoResolution displayL ocalVideo 41 joinToConference 41 removeFromConference 41 setConferenceVideoWindow 4
57. er message Parameters fromDisplayName_ The display name of sender from The message sender toDisplayName The display name of receiver to The receiver mimeType The message mime type subMimeType The message sub mime type messageData The received message body it s can be text or binary data messageDataLengt The length of messageData h void onSendMessageSuccess long sessionid messageld long messageld If the message was sent succeeded in dialog this event will be triggered Parameters sessionId The session ID of the call messageld The message ID it s equals the return value of sendMessage function void onSendMessageFailure long sessionid messageld long messageld reason char reason code int code If the message was sent failure out of dialog this event will be triggered Parameters sessionId The session ID of the call 66 messageld The message ID it s equals the return value of sendMessage function reason The failure reason code Failure code void onSendOutOfDialogMessageSuccess long messageld fromDisplayName char fromDisplayName from char from toDisplayName char toDisplayName to char to If the message was sent succeeded out of dialog this event will be triggered Parameters messageld The message ID it s equals the return value of SendOutOfDialogMessage func
58. er value use getExtensionHeaderValue to do it NSString headerValue myVoIPSdk getExtensionHeaderValue message headerName name int addExtensionHeader NSSiring headerName headerValue NSSiring headerValue Add the extension header custom header into every outgoing SIP message Parameters headerName The custom header name which will be appears in every outgoing SIP message headerValue The custom header value Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int clearAddExtensionHeaders Clear the added extension headers custom headers Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Example we have added two custom headers into every outgoing SIP message and want remove them myVoIPSdk addExtensionHeader Blling headerValue usd100 00 myVoIPSdk addExtensionHeader ServiceId headerValue 8873456 myVoIPSdk clearAddextensionHeaders int modifyHeaderValue NSString headerName headerValue NSString header Value Modify the special SIP header value for every outgoing SIP message Parameters headerName The SIP header name which will be modify it s value 23 headerValue The heaver value want to modify Returns If the function succeeds the return value is 0 If t
59. ers codecType Audio codec type defined in the PortSIPTypes file payloadType The new RTP payload type that you want to set Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoCodecPayloadType VIDEOCODEC_TYPE codecType payloadType int payloadType Set the RTP payload type for dynamic Video codec Parameters codecType Video codec type defined in the PortSIPTypes file payloadType The new RTP payload type that you want to set Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setAudioCodecParameter AUDIOCODEC_TYPE codecType parameter NSString parameter 14 Set the codec parameter for audio codec Parameters codecType Audio codec type defined in the PortSIPTypes file parameter The parameter in string format Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Example myVoIPsdk setAudioCodecParameter AUDIOCODEC_AMR parameter mode set 0 octet align 1 robust sorting 0 int setVideoCodecParameter VIDEOCODEC_TYPE codecType parameter NSString parameter Set the codec parameter for video codec Parameters codecType Video codec type defined in the PortSIPTypes file parameter The parameter in s
60. etely in the OS BOOL PortSIPSDK getSystemInputMute Gets the mute state of the input device in the operating system void PortSIPSDK audioPlayLoopbackTest Use to do the audio device loop back test 52 Detailed Description Function Documentation int getNumOfVideoCaptureDevices Gets the number of available capture devices Returns The return value is number of video capture devices if fails the return value is a specific error code less than 0 int getVideoCaptureDeviceName int index uniqueld NSString uniqueldUTF8 deviceName NSSiring deviceNameUTF8 Gets the name of a specific video capture device given by an index Parameters index Device index 0 1 2 N 1 where N is given by getNumOfVideoCaptureDevices Also 1 is a valid value and will return the name of the default capture device uniqueldUTF8 Unique identifier of the capture device deviceNameUTF8 A character buffer to which the device name will be copied as a null terminated string in UTF8 format Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getNumOfRecordingDevices Gets the number of audio devices available for audio recording Returns The return value is number of recording devices If the function fails the return value is a specific error code less than 0 int getNumOfPlayoutDevices
61. events 57 onInviteUpdated audioCodecs videoCodecs exist sAudio exists Video Call events 58 onPlayAudioFileFinished fileName Play audio and video file finished events 66 onPlay VideoFileFinished Play audio and video file finished events 67 onPresenceOffline from Presence events 64 onPresenceOnline from stateText Presence events 64 onPresenceRecvSubscribe fromDisplayName fro m subject Presence events 64 onReceivedRefer referld to from referSipMessag e Refer events 59 onReceivedRTPPacket isAudio RTPPacket pack etSize RTP callback events 67 onReceivedSignaling message Signaling events 61 onRecvDtmfTone tone DTMF events 62 onRecvInfo INFO OPTIONS message events 63 onRecvMessage mimeType subMimeT ype mess ageData messageDataLength MESSAGE message events 65 onRecvOptions INFO OPTIONS message events 63 onRecvOutOfDialogMessage from toDisplayNa me to mimeType subMimeT Ype message Data m essageDataLength MESSAGE message events 65 onReferAccepted Refer events 60 onReferRejected reason code Refer events 60 onRegisterFailure statusCode Register events 56 onRegisterSuccess statusCode Register events 56 onRemoteHold Call events 59 onRemote UnHold audioCodecs videoCodecs exi stsAudio exists Video Call events 59 onSendingRTPPacket isAudio RTPPacket packe tSize RTP callback events 67 onSendingSignaling message Signaling events 61 onSendMessageFailure messageld reason cod
62. fTone tone Detailed Description Function Documentation void onRecvDtmfTone long sessionid tone int tone This event will be triggered when received a DTMF tone from remote side Parameters sessionld The session ID of the call tone Dtmf tone code Description 0 The DTMF tone 0 1 The DTMF tone 1 2 The DTMF tone 2 3 The DTMF tone 3 4 The DTMF tone 4 5 The DTMF tone 5 6 The DTMF tone 6 7 The DTMF tone 7 8 The DTMF tone 8 63 9 The DTMF tone 9 10 The DTMF tone 11 The DTMF tone 12 The DTMF tone A 13 The DTMF tone B 14 The DTMF tone C 15 The DTMF tone D 16 The DTMF tone FLASH INFO OPTIONS message events Functions void lt PortSIPEventDelegate gt onRecvOptions void lt PortSIPEventDelegate gt onRecvInfo Detailed Description Function Documentation void onRecvOptions char optionsMessage This event will be triggered when received the OPTIONS message Parameters optionsMessage The received whole OPTIONS message in text format void onRecvinfo char infoMessage This event will be triggered when received the INFO message Parameters infoMessage The received whole INFO message in text format Presence events Functions void lt PortSIPEventDelegate gt onPresenceRecvSubscribe fromDisplayName from subject void
63. ferld Reject the REFER request Parameters referld The ID of REFER request that comes from onReceivedRefer callback event Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Send audio and video stream functions Functions int PortSIPSDK enableSendPcmStreamToRemote state streamSamplesPerSec Enable the SDK send PCM stream data to remote side from another source to instread of microphone int PortSIPSDK sendPcmStreamToRemote data Send the audio stream in PCM format from another source to instead of audio device capture microphone int PortSIPSDK enableSend VideoStreamToRemote state Enable the SDK send video stream data to remote side from another source to instead of camera int PortSIPSDK sendVideoStreamToRemote data width height Send the video stream to remote 34 Detailed Description Function Documentation int enableSendPcmStreamToRemote long sessionld state BOOL state streamSamplesPerSec int streamSamplesPerSec Enable the SDK send PCM stream data to remote side from another source to instread of microphone Parameters sessionld The session ID of call state Set to true to enable the send stream false to disable streamSamplesPer The PCM stream data sample in seconds for example 8000 or 16000 Sec Returns If the function succeeds the
64. from DisplayName from toDisplayName to 66 modifyHeaderValue headerValue Access SIP message header functions 22 muteMicrophone Audio and video functions 26 muteSession muteIncoming Audio muteOutgoing Audio muteIncoming Video muteOutgoing Video Call functions 30 muteSpeaker Audio and video functions 26 MWI events 61 onWaitingFaxMessage urgentNewMessage Count urgentOldMessageCount newMessag eCount oldMessageCount 62 onWaiting VoiceMessage urgentNewMessa geCount urgentOldMessageCount newMess ageCount oldMessageCount 61 NIC and local IP functions 11 getLocallpAddress 11 getNICNums 11 onACTVTransferFailure reason code Refer events 60 onACTVTransferSuccess Refer events 60 onAudioRawCallback audioCallbackMode data dataLength samplingFreqHz Audio and video stream callback events 68 onInviteAnswered callerDisplayName caller call eeDisplayName callee audioCodecs videoCodec s exists Audio exists Video Call events 57 onInviteBeginingForward Call events 58 onInviteClosed Call events 58 onInviteConnected Call events 58 onInviteFailure reason code Call events 58 84 onInviteIncoming callerDisplayName caller call eeDisplayName callee audioCodecs videoCodec s exists Audio exists Video Call events 56 onInviteRinging statusText statusCode Call events 57 onInviteSessionProgress audioCodecs videoCod ecs existsEarly Media existsAudio exists Video Call events 57 onInviteTrying Call
65. g call is fails this event triggered Parameters sessionId The session ID of the call reason The failure reason code The failure code void onInviteUpdated long sessionid audioCodecs char audioCodecs videoCodecs char videoCodecs existsAudio BOOL existsAudio existsVideo BOOL existsVideo optional This event will be triggered when remote party updated this call Parameters sessionId The session ID of the call audioCodecs The matched audio codecs it s separated by if have more than one codec videoCodecs The matched video codecs it s separated by if have more than one codec existsAudio If it s true means this call include the audio exists Video If it s true means this call include the video void onInviteConnected long sessionid optional This event will be triggered when UAC sent UAS received ACK the call is connected Some functions hold updateCall etc can called only after the call connected otherwise the functions will return error Parameters sessionld The session ID of the call void onInviteBeginingForward char forward To optional If the enableCallForward method is called and a call is incoming the call will be forwarded automatically and trigger this event Parameters forwardTo The forward target SIP URI void onInviteClosed long sessionid optional This event is trigger
66. he function fails the return value is a specific error code int clearModifyHeaders Clear the modify headers value no longer modify every outgoing SIP message header values Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Example modified two headers value for every outging SIP message and then clear it myVoIPSdk modifyHeaderValue Expires headerValue 1000 myVoIPSdk modifyHeaderValue User Agent headerValue MyTest Softphone 1 0 myVoIPSdk cleaModifyHeaders Audio and video functions Functions int PortSIPSDK setVideoDeviceld Set the video device that will use for video call int PortSIPSDK setVideoResolution Set the video capture resolution int PortSIPSDK setVideoBitrate Set the video bit rate int PortSIPSDK setVideoFrameRate Set the video frame rate int PortSIPSDK sendVideo sendState Send the video to remote side int PortSIPSDK set VideoOrientation Changing the orientation of the video void PortSIPSDK setLocalVideoWindow Set the the window that using to display the local video image int PortSIPSDK setRemote VideoWindow remote VideoWindow Set the window for a session that using to display the received remote video image int PortSIPSDK displayLocalVideo Start stop to display the local video image int
67. indtransfer rar use the Windows Media Player to play the AVI file after extracted it will shows how to do the transfer int attendedRefer long sessionid replaceSessionld long replaceSessionld referTo NSString referTo Make an attended refer Parameters sessionId The session ID of the call replaceSessionId Session ID of the replace call referTo Target of the refer it can be sip number sipserver com or number only Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code 33 Remarks Please read the sample project source code to got more details Or download the demo AVI at http www portsip com downloads video blindtransfer rar use the Windows Media Player to play the AVI file after extracted it will shows how to do the transfer long acceptRefer long referld referSignaling NSString referSignaling Accept the REFER request a new call will be make if called this function usuall called after onReceivedRefer callback event Parameters referld The ID of REFER request that comes from onReceivedRefer callback event referSignaling The SIP message of REFER request that comes from onReceivedRefer callback event Returns If the function succeeds the return value is a session ID greater than 0 to the new call for REFER otherwise is a specific error code less than 0 int rejectRefer long re
68. ion fails the return value is a specific error code int joinToConference long sessionld Join a session into exist conference if the call is in hold please un hold first Parameters sessionld Session ID of the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int removeFromConference long sessionid Remove a session from an exist conference Parameters sessionld Session ID of the call 42 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code RTP and RTCP QOS functions Functions int PortSIPSDK setAudioRtcpBandwidth BitsRR BitsRS KBitsAS Set the audio RTCP bandwidth parameters as the RFC3556 6 int PortSIPSDK setVideoRtcpBandwidth BitsRR BitsRS KBitsAS Set the video RTCP bandwidth parameters as the RFC3556 int PortSIPSDK setAudioQos DSCPValue priority Set the DSCP differentiated services code point value of QoS Quality of Service for audio channel int PortSIPSDK setVideoQos DSCP Value Set the DSCP differentiated services code point value of QoS Quality of Service for video channel Detailed Description Function Documentation int setAudioRtcpBandwidth long sessionid BitsRR int BitsRR BitsRS int BitsRS KBitsAS int KBitsAS Set the audio RTCP bandwidth
69. ist conference int PortSIPSDK setConference VideoWindow Set the window for a conference that using to display the received remote video image int PortSIPSDK joinToConference Join a session into exist conference if the call is in hold please un hold first int PortSIPSDK removeFromConference Remove a session from an exist conference Detailed Description 41 Function Documentation int createConference PortSIPVideoRenderView conferenceVideoWindow videoResolution VIDEO_RESOLUTION videoResolution displayLocalVideo BOOL displayLocalVideoInConference Create a conference It s failures if the exists conference isn t destroy yet Parameters conferenceVideoW The PortSIPVideoRenderView which using to display the conference video indow videoResolution The conference video resolution displayLocalVideo Display the local video on video window or not InConference Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setConferenceVideoWindow PortSIPVideoRenderView conferenceVideoWindow Set the window for a conference that using to display the received remote video image Parameters conferenceVideoW The PortSIPVideoRenderView which using to display the conference video indow Returns If the function succeeds the return value is 0 If the funct
70. ketsReceived Obtain the RTP statisics of video Detailed Description Function Documentation int getNetworkStatistics long sessionid currentBufferSize int currentBufferSize preferredBufferSize int preferredBufferSize currentPacketLossRate int currentPacketLossRate currentDiscardRate int currentDiscardRate currentExpandRate int currentExpandRate currentPreemptiveRate int currentPreemptiveRate currentAccelerateRate int currentAccelerateRate Get the in call statistics The statistics are reset after the query Parameters sessionId The session ID of call conversation currentBufferSize Preferred optimal buffer size in ms preferredBufferSiz Preferred optimal buffer size in ms e currentPacketLoss_ Loss rate network late in percent Rate currentDiscardRat Fraction of synthesized speech inserted through pre emptive expansion e currentExpandRat Fraction of synthesized speech inserted through pre emptive expansion e currentPreemptive Fraction of synthesized speech inserted through pre emptive expansion Rate currentAccelerate Fraction of data removed through acceleration Rate Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getAudioRtpStatistics long sessionid averageJitterMs int averageJitterMs maxJitterMs int maxJitterMs dis
71. l any SDK API functions in this event directly If you want to call the API functions or other code which will spend long time you should post a message to another thread and execute SDK API functions or other code in another thread void onVideoRawCallback long sessionld videoCallbackMode int videoCallbackMode width int width height int height data unsigned char data dataLength int dataLength This event will be triggered once received the video packets if called enable VideoStreamCallback function Parameters sessionld The session ID of the call 69 videoCallbackMod The type which pasdded in enable VideoStreamCallback function e width The width of video image height The height of video image data The memory of video stream it s YUV420 format YV12 dataLength The data size Note Don t call any SDK API functions in this event directly If you want to call the API functions or other code which will spend long time you should post a message to another thread and execute SDK API functions or other code in another thread 70 Class Documentation lt PortSiIPEventDelegate gt Protocol Reference PortSIP SDK Callback events Delegate import lt PortSIPEventDelegate h gt Inherits lt NSObject gt Instance Methods void onRegisterSuccess statusCode void onRegisterFailure statusCode void onInviteIncoming callerDisplayName caller ca
72. l logPath maxLine agent audioDeviceLayer videoDeviceLayer Initialize the SDK void unInitialize Un initialize the SDK and release resources int setUser displayName authName password localIP localSIPPort userDomain SIPServer SIPServerPort STUNServer STUNServerPort outboundServer outboundServerPort Set user account info int registerServer retryTimes Register to SIP proxy server login to server int unRegisterServer Un register from the SIP proxy server int setLicenseKey Set the license key must called before setUser function int getNICNums Get the Network Interface Card numbers NSString getLocalIpAddress Get the local IP address by Network Interface Card index int addAudioCodec Enable an audio codec it will be appears in SDP int add VideoCodec Enable a video codec it will be appears in SDP BOOL isAudioCodecEmpty Detect enabled audio codecs is empty or not BOOL isVideoCodecEmpty Detect enabled video codecs is empty or not int setAudioCodecPayloadType payloadType Set the RTP payload type for dynamic audio codec int setVideoCodecPayloadType payloadType Set the RTP payload type for dynamic Video codec void clearAudioCodec Remove all enabled audio codecs void clear VideoCodec Remove all enabled video codecs int setAudioCodecParameter parameter Set the codec parameter for audio codec int
73. livePayloadType deltaTransmitTimeMS Enable or disable send RTP keep alive packet during the call is established int setKeepAliveTime Enable or disable send SIP keep alive packet int setAudioSamples maxPtime Set the audio capture sample int addSupportedMimeType mimeType subMimeT ype Set the SDK receive the SIP message that include special mime type NSString getExtensionHeaderValue headerName Access the SIP header of SIP message int addExtensionHeader headerValue Add the extension header custom header into every outgoing SIP message int clearAddExtensionHeaders Clear the added extension headers custom headers int modifyHeaderValue headerValue Modify the special SIP header value for every outgoing SIP message int clearModifyHeaders 74 Clear the modify headers value no longer modify every outgoing SIP message header values int setVideoDeviceld Set the video device that will use for video call int setVideoResolution Set the video capture resolution int setVideoBitrate Set the video bit rate int setVideoFrameRate Set the video frame rate int send Video sendState Send the video to remote side int setVideoOrientation Changing the orientation of the video void setLocalVideoWindow Set the the window that using to display the local video image int setRemote Video Window remote VideoWindo
74. lleeDisplayName callee audioCodecs videoCodecs exis tsAudio exists Video void onInviteTrying void onInviteSessionProgress audioCodecs videoCodecs existsEarlyMedia existsAudio exists Video void onInviteRinging statusText statusCode void onInviteAnswered callerDisplayName caller calleeDisplayName callee audioCodecs videoCodecs exis tsAudio exists Video void onInviteFailure reason code void onInviteUpdated audioCodecs videoCodecs existsAudio exists Video void onInviteConnected void onInviteBeginingForward void onInviteClosed void onRemoteHold void onRemoteUnHold audioCodecs videoCodecs exists Audio exists Video void onReceivedRefer referld to from referSipMessage void onReferAccepted void onReferRejected reason code void onTransferTrying void onTransferRinging void onACTVTransferSuccess void onnACTVTransferFailure reason code void onReceivedSignaling message void onSendingSignaling message void onWaiting VoiceMessage urgentNewMessageCount urgentOldMessageCount newMessageCount oldM essageCount void onWaitingFaxMessage urgentNewMessageCount urgentOldMessageCount newMessageCount oldMes sageCount void onRecvDtmfTone tone void onRecvOptions void onRecvInfo void onPresenceRecvSubscribe fromDisplayName from subject void onPresenceOnline from stateText void onPresenceOffline from
75. meters state AEC type default is EC_NONE void enableCNG BOOL state Enable disable Comfort Noise Generator CNG Parameters state state Set to true to enable CNG false to disable 47 void enableAGC AGC_MODES siate Enable disable Automatic Gain Control AGC Parameters state AGC type default is AGC_NONE void enableANS NS_MODES state Enable disable Audio Noise Suppression ANS Parameters state NS type default is NS_NONE Send OPTIONS INFO MESSAGE functions Functions int PortSIPSDK sendOptions sdp Send OPTIONS message int PortSIPSDK sendInfo mimeType subMimeT ype infoContents Send a INFO message to remote side in dialog long PortSIPSDK sendMessage mimeType subMimeType message messageLength Send a MESSAGE message to remote side in dialog long PortSIPSDK sendOutOfDialogMessage mimeT ype subMimeType message messageLength Send a out of dialog MESSAGE message to remote side Detailed Description Function Documentation int sendOptions NSString to sdp NSSiring sdp Send OPTIONS message Parameters to The receiver of OPTIONS message sdp The SDP of OPTIONS message it s optional if don t want send the SDP with OPTIONS message 48 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error cod
76. nction with no loop mode this event will be triggered once the file play finished Parameters sessionId The session ID of the call RTP callback events Functions void lt PortSIPEventDelegate gt onReceivedRTPPacket isAudio RTPPacket packetSize void lt PortSIPEventDelegate gt onSendingRTPPacket isAudio RTPPacket packetSize Detailed Description Function Documentation void onReceivedRTPPacket long sessionid isAudio BOOL isAudio RTPPacket unsigned char RTPPacket packetSize int packetSize If called setRTPCallback function to enabled the RTP callback this event will be triggered once received a RTP packet Parameters sessionId The session ID of the call isAudio If the received RTP packet is of audio this parameter is true otherwise false RTPPacket The memory of whole RTP packet packetSize The size of received RTP Packet Note Don t call any SDK API functions in this event directly If you want to call the API functions or other code which will spend long time you should post a message to another thread and execute SDK API functions or other code in another thread void onSendingRTPPacket long sessionid isAudio BOOL isAudio RTPPacket unsigned char RTPPacket packetSize int packetSize If called setRTPCallback function to enabled the RTP callback this event will be triggered once sending a RTP packet Parameter
77. of the call code Reject code for example 486 480 etc Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int hangUp long sessionId hangUp Hang up the call 29 Parameters sessionld Session ID of the call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int answerCall long sessionid videoCall BOOL videoCall answerCall Answer the incoming call Parameters sessionId The session ID of call videoCall If the incoming call is a video call and the video codec is matched set to true to answer the video call If set to false the answer call doesn t include video codec answer anyway Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int updateCall long sessionld enableAudio BOOL enableAudio enableVideo BOOL enableVideo Use the re INVITE to update the established call Parameters sessionId The session ID of call enableAudio Set to true to allow the audio in update call false for disable audio in update call enable Video Set to true to allow the video in update call false for disable video in update call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific er
78. ons void lt PortSIPEventDelegate gt onInviteIncoming callerDisplayName caller calleeDisplayName callee audi oCodecs videoCodecs existsAudio exists Video void lt PortSIPEventDelegate gt onInviteTrying void lt PortSIPEventDelegate gt onInviteSessionProgress audioCodecs videoCodecs existsEarlyMedia exists Audio exists Video void lt PortSIPEventDelegate gt onInviteRinging statusText statusCode void lt PortSIPEventDelegate gt onInviteAnswered callerDisplayName caller calleeDisplayName callee audi oCodecs videoCodecs existsAudio exists Video void lt PortSIPEventDelegate gt onInviteFailure reason code void lt PortSIPEventDelegate gt onInviteUpdated audioCodecs videoCodecs exists Audio exists Video void lt PortSIPEventDelegate gt onInviteConnected void lt PortSIPEventDelegate gt onInviteBeginingForward void lt PortSIPEventDelegate gt onInviteClosed void lt PortSIPEventDelegate gt onRemoteHold void lt PortSIPEventDelegate gt onRemoteUnHold audioCodecs videoCodecs exists Audio exists Video Detailed Description Function Documentation void onInvitelncoming long sessionid callerDisplayName char callerDisplayName caller char caller calleeDisplayName char calleeDisplayName callee char callee 57 audioCodecs char audioCodecs videoCodecs char videoCodecs existsAudio
79. or version number Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int enableReliableProvisional BOOL enable Enable disable PRACK Parameters enable enable Set to true to enable the SDK support PRACK default the PRACK is disabled Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int enable3GppTags BOOL enable Enable disable the 3Gpp tags include ims icsi mmtel and g 3gpp smsip Parameters enable enable Set to true to enable the SDK support 3Gpp tags Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code void enableCallbackSendingSignaling BOOL enable Enable disable callback Parameters the sending SIP messages enable enable Set as true to enable callback the sent SIP messages false to disable Once enabled the onSendingSignaling event will be fired once the SDK sending a SIP message 17 int setSrtpPolicy SRTP_POLICY srtpPolicy Set the SRTP policy Parameters srtpPolicy The SRTP policy Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setRtpPortRange int minimumRitpAudioPort maximumRtpA
80. ptime and maxptime attribute int addSupportedMimeType NSString methodName mimeType NSString mimeType subMimeType NSString subMimeType Set the SDK receive the SIP message that include special mime type Parameters methodName Method name of the SIP message likes INVITE OPTION INFO MESSAGE UPDATE ACK etc More details please read the RFC3261 mimeType The mime type of SIP message subMimeType The sub mime type of SIP message 21 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Default the PortSIP VoIP SDK support these media types mime types that in the below incoming SIP messages message sipfrag in NOTIFY message application simple message summary in NOTIFY message text plain in MESSAGE message application dtmf relay in INFO message application media_control xml in INFO message The SDK allows received SIP message that included above mime types Now if remote side send a INFO SIP message this message Content Type header value is text plain the SDK will reject this INFO message because text plain of INFO message does not included in the default support list Then how to let the SDK receive the SIP INFO message that included text plain mime type We should use addSupportedMimyType to do it myVoIPSdk addSupportedMimeType INFO mimeType text subMimeType plain If want to receive
81. putMute Device Manage functions 54 getVersion minorVersion Additional setting functions 16 getVideoCaptureDeviceName uniqueld deviceN ame Device Manage functions 52 getVideoRtpStatistics bytesSent packetsSent byt esReceived packetsReceived RTP statistics functions 45 hangUp Call functions 28 hold Call functions 29 INFO OPTIONS message events 63 onRecvInfo 63 onRecvOptions 63 Initialize and register functions 8 initialize loglevel logPath maxLine agent a udioDeviceLayer videoDeviceLayer 9 registerServer retryTimes 10 setLicenseKey 10 setUser displayName authName password 1 ocalIP localSIPPort userDomain SIPServer SIPServerPort STUNServer STUNServerPo rt outboundServer outboundServerPort 9 unRegisterServer 10 initialize loglevel logPath maxLine agent audio DeviceLayer videoDeviceLayer Initialize and register functions 9 isAudioCodecEmpty Audio and video codecs functions 13 isVideoCodecEmpty Audio and video codecs functions 13 joinToConference Conference functions 41 MESSAGE message events 64 onRecvMessage mimeT ype subMimeType messageData messageDataLength 65 onRecvOutOfDialogMessage from toDispla yName to mimeType subMimeType messa geData messageDataLength 65 onSendMessageFailure messageld reason c ode 65 onSendMessageSuccess messageld 65 onSendOutOfDialogMessageFailure fromD isplayName from toDisplayName to reason code 66 onSendOutOfDialogMessageSuccess
82. rements Development using the PortSIP VoIP IMS SDK for Mac requires an Intel based Macintosh running Snow Leopard OS X 10 8 or higher Xcode 5 0 and above Frequently Asked Questions 1 Where can I download the PortSIP VoIP SDK for test All sample projects of the PortSIP VoIP SDK can be download to test at http www PortSIP com downloads html http www PortSIP com voipsdk html 2 How to compile the sample project Download the sample projects from PortSIP website Extract the zip file Open the project by your xcode 4 Compile the sample project directly the trial version SDK allows 2 3 minutes conversation w I Ei 3 Create a new project base on PortSIP VoIP SDK 1 Download the Sample project and evaluation SDK and extract it to a directory 2 Run the Xcode and create a new OS X Cocoa Applicatiion Project 3 Drag and drop PortSIPSDK framework from Finder to XCode gt Frameworks 4 Copying Frameworks Files While Building a Product Click Build Phases at the top of the project editor Choose Editor gt Add Build Phase gt Add Copy Files Build Phase Destination specify Frameworks Click the Add button to select PortSIPSDK framework to copy and click Add 5 Add the code in h file to import the SDK example import lt PortSIPSDK PortSIPSDK h gt 6 Inherit the interface PortSIPEventDelegate to process the callback events 7 Initialize sdk Example MEOCESUESDKe Eege salllioc Manat mPortSIPSDK
83. ror code Remarks Example usage Example 1 A called B with the audio only B answered A there has an audio conversation between A B Now A want to see B video A use these functions to do it myVoIPSdk clearVideoCodec myVoIPSdk addVideoCodec VIDEOCODEC_H264 myVoIPSdk updateCall sessionId enableAudio true enableVideo true Example 2 Remove video stream from currently conversation myVoIPSdk updateCall sessionId enableAudio true enableVideo false int hold long sessionld To place a call on hold 30 Parameters sessionld The session ID of call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int unHold long sessionld Take off hold Parameters sessionld The session ID of call Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int muteSession long sessionid mutelncomingAudio BOOL mutelncomingAudio muteOutgoingAudio BOOL muteOutgoingAudio mutelncomingVideo BOOL mutelncoming Video muteOutgoingVideo BOOL muteOutgoing Video Mute the specified session audio or video Parameters sessionId The session ID of the call muteIncomingAudi Set it to true to mute incoming audio stredam can t hearing remote side audio 0 muteOutgoingAudi Set it to true to mute o
84. rward BOOL forBusyOnly forwardTo NSString forwardTo Enable call forward Parameters forBusyOnly If set this parameter as true the SDK will forward all incoming calls when currently it s busy If set this as false the SDK forward all inconing calls anyway forwardTo The call forward target it s must likes sip xxxx sip portsip com Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int disableCallForward Disable the call forward the SDK is not forward any incoming call after this function is called Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int enableSessionTimer int timerSeconds refreshMode SESSION_REFRESH_MODE refreshMode Allows to periodically refresh Session Initiation Protocol SIP sessions by sending repeated INVITE requests Parameters timerSeconds The value of the refresh interval in seconds Minimum requires 90 seconds refreshMode Allow set the session refresh by UAC or UAS SESSION_REFERESH_UAC or SESSION_REFERESH_UAS Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks The repeated INVITE requests or re INVITEs are sent during an active call leg to allow user agents UA or proxies to determine the status of a SIP session
85. s sessionId The session ID of the call isAudio If the received RTP packet is of audio this parameter is true otherwise false RTPPacket The memory of whole RTP packet 68 packetSize The size of received RTP Packet Note Don t call any SDK API functions in this event directly If you want to call the API functions or other code which will spend long time you should post a message to another thread and execute SDK API functions or other code in another thread Audio and video stream callback events Functions void lt PortSIPEventDelegate gt onAudioRawCallback audioCallbackMode data dataLength samplingFreqH Zz void lt PortSIPEventDelegate gt 0n VideoRawCallback videoCallbackMode width height data dataLength Detailed Description Function Documentation void onAudioRawCallback long sessionid audioCallbackMode int audioCallbackMode data unsigned char data dataLength int dataLength samplingFreqHz int samplingFreqHz This event will be triggered once received the audio packets if called enableAudioStreamCallback function Parameters sessionId The session ID of the call audioCallbackMod TThe type which pasdded in enableAudioStreamCallback function e data The memory of audio stream it s PCM format dataLength The data size sampling Fred H The audio stream sample in HZ for example it s 8000 or 16000 Note Don t cal
86. s to periodically refresh Session Initiation Protocol SIP sessions by sending repeated INVITE requests int PortSIPSDK disableSessionTimer Disable the session timer void PortSIPSDK setDoNotDisturb Enable the Do not disturb to enable disable int PortSIPSDK detectMwi Use to obtain the MWI status int PortSIPSDK enableCheckMwi Allows enable disable the check MWI Message Waiting Indication 6 int PortSIPSDK setRtpKeepAlive keepAlivePayloadType deltaTransmitTimeMS Enable or disable send RTP keep alive packet during the call is established int PortSIPSDK setKeepAliveTime Enable or disable send SIP keep alive packet int PortSIPSDK setAudioSamples maxPtime Set the audio capture sample int PortSIPSDK addSupportedMimeType mimeT ype subMimeType Set the SDK receive the SIP message that include special mime type Detailed Description Function Documentation int setDisplayName NSString displayName Set user display name Parameters displayName The display name 16 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int getVersion int majorVersion minorVersion int minorVersion Get the current version number of the SDK Parameters majorVersion Return the major version number minorVersion Return the min
87. scribe will be triggered the event inclues the subscribe id Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int presenceRejectSubscribe long subscribeld Reject a presence SUBSCRIBE request which received from contact Parameters subscribeld Subscribe id when received a SUBSCRIBE request from contact the event onPresenceRecvSubscribe will be triggered the event inclues the subscribe id Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int presenceOnline long subscribeld statusText NSString statusText Send a NOTIFY message to contact to notify that presence status is online changed Parameters subscribeld Subscribe id when received a SUBSCRIBE request from contact the event onPresenceRecvSubscribe will be triggered the event inclues the subscribe id statusText The state text of presende online for example I m here Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int presenceOffline long subscribeld 51 Send a NOTIFY message to contact to notify that presence status is offline Parameters subscribeld Subscribe id when received a SUBSCRIBE request from contact the event onPresenceRecvSubscribe will be triggered the event inclues the
88. scription Function Documentation void onReceivedRefer long sessionid referld long referld to char to from char from referSipMessage char referSipMessage This event will be triggered once received a REFER message Parameters sessionId The session ID of the call referld The ID of the REFER message pass it to acceptRefer or rejectRefer to The refer target from The sender of REFER message referSipMessage The SIP message of REFER pass it to acceptRefer function 60 void onReferAccepted long sessionld This callback will be triggered once remote side called acceptRefer to accept the REFER Parameters sessionld The session ID of the call void onReferRejected long sessionid reason char reason code int code This callback will be triggered once remote side called rejectRefer to reject the REFER Parameters sessionId The session ID of the call reason Reject reason code Reject code void onTransferTrying long sessionld When the refer call is processing this event trigged Parameters sessionld The session ID of the call void onTransferRinging long sessionld When the refer call is ringing this event trigged Parameters sessionId The session ID of the call void onACTVTransferSuccess long sessionld When the refer call is succeeds this event will be triggered
89. set VideoCodecParameter parameter Set the codec parameter for video codec int setDisplayName 73 Set user display name int get Version minorVersion Get the current version number of the SDK int enableReliableProvisional Enable disable PRACK int enable3GppTags Enable disable the 3Gpp tags include ims icsi mmtel and g 3gpp smsip void enableCallbackSendingSignaling Enable disable callback the sending SIP messages int setSrtpPolicy Set the SRTP policy int setRtpPortRange maximumRtpAudioPort minimumRtp VideoPort maximumkRtp VideoPort Set the RTP ports range for audio and video streaming int setRtcpPortRange maximumRtcpAudioPort minimumRtcp VideoPort maximumRtcp VideoPort Set the RTCP ports range for audio and video streaming int enableCallForward forwardTo Enable call forward int disableCallForward Disable the call forward the SDK is not forward any incoming call after this function is called int enableSessionTimer refreshMode Allows to periodically refresh Session Initiation Protocol SIP sessions by sending repeated INVITE requests int disableSessionTimer Disable the session timer void setDoNotDisturb Enable the Do not disturb to enable disable int detectMwi Use to obtain the MWI status int enableCheckMwi Allows enable disable the check MWI Message Waiting Indication int setRtpKeepAlive keepA
90. subscribe id Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Device Manage functions Functions 6 int PortSIPSDK getNumOfVideoCaptureDevices Gets the number of available capture devices int PortSIPSDK getVideoCaptureDeviceName uniqueld deviceName Gets the name of a specific video capture device given by an index int PortSIPSDK getNumOfRecordingDevices Gets the number of audio devices available for audio recording int PortSIPSDK getNumOfPlayoutDevices Gets the number of audio devices available for audio playout NSString PortSIPSDK getRecordingDeviceName Gets the name of a specific recording device given by an index NSString PortSIPSDK getPlayoutDeviceName Gets the name of a specific playout device given by an index int PortSIPSDK setSpeakerVolume Set the speaker volume level int PortSIPSDK getSpeaker Volume Gets the speaker volume level int PortSIPSDK setSystemOutputMute Mutes the speaker device completely in the OS BOOL PortSIPSDK getSystemOutputMute Retrieves the output device mute state in the operating system int PortSIPSDK setMic Volume Sets the microphone volume level int PortSIPSDK getMicVolume Retrieves the current microphone volume int PortSIPSDK setSystemInputMute Mute the microphone input device compl
91. t s optional that allow pass a empty string if not use outbound server outboundServerPo Outbound proxy server port it will be ignored if the outboundServer is empty rt Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int registerServer int expires retryTimes int retryTimes Register to SIP proxy server login to server Parameters expires Registration refresh Interval in seconds maximum is 3600 it will be inserted into SIP REGISTER message headers retryTimes The retry times if failed to refresh the registration set to lt 0 the retry will be disabled and onRegisterFailure callback triggered when retry failure Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code if register to server succeeded then onRegisterSuccess will be triggered otherwise onRegisterFailure triggered int unRegisterServer Un register from the SIP proxy server Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setLicenseKey NSSitring key Set the license key must called before setUser function Parameters key The SDK license key please purchase from PortSIP 11 Returns If the function succeeds the return value is 0 If the function fails the return value is
92. tOfDialogMessage from toDisplayName to mimeT ype subMime Type messageData messageDataLength void lt PortSIPEventDelegate gt onSendMessageSuccess messageld void lt PortSIPEventDelegate gt onSendMessageFailure messageld reason code void lt PortSIPEventDelegate gt onSendOutOfDialogMessageSuccess fromDisplayName from toDisplayNa me to void lt PortSIPEventDelegate gt onSendOutOfDialogMessageFailure fromDisplayName from toDisplayNam e to reason code 65 Detailed Description Function Documentation void onRecvMessage long sessionid mimeType char mimeType subMimeType char subMimeType messageData unsigned char messageData messageDataLength int messageDataLength This event will be triggered when received a MESSAGE message in dialog Parameters sessionId The session ID of the call mimeType The message mime type subMimeType The message sub mime type messageData The received message body it s can be text or binary data messageDataLengt The length of messageData h void onRecvOutOfDialogMessage char fromDisplayName from char from toDisplayName char toDisplayName to char to mimeType char mimeType subMimeType char subMimeType messageData unsigned char messageData messageDataLength int messageDataLength This event will be triggered when received a MESSAGE message out of dialog for example pag
93. the NOTIFY message with application media_control xml then myVoIPSdk addSupportedMimeType NOTIFY mimeType application subMimeType media_control xml About the mime type details please visit this website http www iana org assignments media types Access SIP message header functions Functions 6 NSString PortSIPSDK getExtensionHeaderValue headerName Access the SIP header of SIP message int PortSIPSDK addExtensionHeader headerValue Add the extension header custom header into every outgoing SIP message int PortSIPSDK clearAddExtensionHeaders Clear the added extension headers custom headers int PortSIPSDK modifyHeaderValue headerValue Modify the special SIP header value for every outgoing SIP message int PortSIPSDK clearModifyHeaders Clear the modify headers value no longer modify every outgoing SIP message header values Detailed Description Function Documentation NSString getExtensionHeaderValue NSSiring sipMessage headerName NSString headerName Access the SIP header of SIP message 22 Parameters sipMessage The SIP message headerName Which header want to access of the SIP message Returns If the function succeeds the return value is headerValue If the function fails the return value is nil Remarks When got a SIP message in the onReceivedSignaling callback event and want to get SIP message head
94. the operating system int setMic Volume Sets the microphone volume level int getMicVolume Retrieves the current microphone volume int setSystemInputMute Mute the microphone input device completely in the OS BOOL getSystemInputMute Gets the mute state of the input device in the operating system void audioPlayLoopbackTest Use to do the audio device loop back test Properties id lt PortSIPEventDelegate gt delegate Detailed Description PortSIP VoIP SDK functions class 78 Author Copyright c 2006 2015 PortSIP Solutions Inc All rights reserved Version 11 2 2 See also http www PortSIP com PortSIP SDK functions class description The documentation for this class was generated from the following file PortSIPSDK h 79 PortSIPVideoRenderView Class Reference PortSIP VoIP SDK Video Render View class import lt PortSIPVideoRenderView h gt Inherits NS View Instance Methods void initVideoRender Initialize the Video Render view shoud Initialize render before use void release VideoRender release the Video Render void getVideoRenderView Don t use this Just call by SDK void updateVideoRenderFrame change the Video Render size Detailed Description PortSIP VoIP SDK Video Render View class Author Copyright c 2006 2015 PortSIP Solutions Inc All rights reserved Version 11 2 2 See also
95. tion fromDisplayName_ The display name of message sender from The message sender toDisplayName The display name of message receiver to The message receiver void onSendOutOfDialogMessageFailure long messageld fromDisplayName char fromDisplayName from char from toDisplayName char toDisplayName to char to reason char reason code int code If the message was sent failure out of dialog this event will be triggered Parameters messageld The message ID it s equals the return value of SendOutOfDialogMessage function fromDisplayName_ The display name of message sender from The message sender toDisplayName The display name of message receiver to The message receiver reason The failure reason code The failure code Play audio and video file finished events Functions void lt PortSIPEventDelegate gt onPlayAudioFileFinished fileName void lt PortSIPEventDelegate gt onPlayVideoFileFinished Detailed Description Function Documentation void onPlayAudioFileFinished long sessionid fileName char fileName If called playAudioFileToRemote function with no loop mode this event will be triggered once the file play finished 67 Parameters sessionld The session ID of the call fileName The play file name void onPlayVideoFileFinished long sessionld If called playVideoFileToRemote fu
96. tring format Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Example myVoIPsdk setVideoCodecParameter VIDEOCODEC_H264 parameter profile level id 420033 packetization mode 0 Additional setting functions Functions int PortSIPSDK setDisplayName Set user display name int PortSIPSDK getVersion minorVersion Get the current version number of the SDK int PortSIPSDK enableReliableProvisional Enable disable PRACK int PortSIPSDK enable3GppTags Enable disable the 3Gpp tags include ims icsi mmtel and g 3gpp smsip void PortSIPSDK enableCallbackSendingSignaling Enable disable callback the sending SIP messages int PortSIPSDK setSrtpPolicy 15 Set the SRTP policy int PortSIPSDK setRtpPortRange maximumRtpAudioPort minimumRtp VideoPort maximumRtp VideoPo rt Set the RTP ports range for audio and video streaming int PortSIPSDK setRtcpPortRange maximumRtcpAudioPort minimumRtcp VideoPort maximumRtcp Vid eoPort Set the RTCP ports range for audio and video streaming int PortSIPSDK enableCallForward forwardTo Enable call forward int PortSIPSDK disableCallForward Disable the call forward the SDK is not forward any incoming call after this function is called int PortSIPSDK enableSessionTimer refreshMode Allow
97. ucceeds the return value is 0 If the function fails the return value is a specific error code 25 int setVideoFrameRate int frameRate Set the video frame rate Parameters frameRate The frame rate value minimum is 5 maximum is 30 The bigger value will give you better video quality but require more bandwidth Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks Usually you do not need to call this function set the frame rate the SDK using default frame rate int sendVideo long sessionid sendState BOOL sendState Send the video to remote side Parameters sessionId The session ID of the call sendState Set to true to send the video false to stop send Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int setVideoOrientation int rotation Changing the orientation of the video Parameters rotation The video rotation that you want to set 0 90 180 270 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code void setLocalVideoWindow PortSIPVideoRenderView local VideoWindow Set the the window that using to display the local video image Parameters localVideoWindow The PortSIPVideoRenderView to display local video
98. udioPort int maximumRipAudioPort minimumRtpVideoPort int minimumRtpVideoPort maximumRipVideoPort int maximumRtpVideoPort Set the RTP ports range for audio and video streaming Parameters minimumRtpAudio The minimum RTP port for audio stream Port maximumRtpAudio The maximum RTP port for audio stream Port minimumRtpVideo The minimum RTP port for video stream Port maximumRtpVideo The maximum RTP port for video stream Port Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks The port range max min maxCallLines should more than 4 int setRtcpPortRange int minmumRtcpAudioPort maximumRtcpAudioPort int maximumRicpAudioPort minimumRtcpVideoPort int minimumRtcpVideoPort maximumRicpVideoPort int maximumRtcpVideoPort Set the RTCP ports range for audio and video streaming Parameters minimumRtcpAudi The minimum RTCP port for audio stream oPort maximumRtcpAudi The maximum RTCP port for audio stream oPort minimumRtcpVide The minimum RTCP port for video stream oPort maximumRtcpVide The maximum RTCP port for video stream oPort 18 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code Remarks The port range max min maxCallLines should more than 4 int enableCallFo
99. ue is set to true the speaker is muted set to false to un mute it 27 int setAudioDeviceld int inputDeviceld outputDeviceld int outputDeviceld Set the audio device that will use for audio call Parameters inputDeviceld Device ID index for audio record Microphone outputDeviceld Device ID index for audio playback Speaker Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code void getDynamicVolumeLevel int soeakerVolume microphoneVolume int microphone Volume Obtain the dynamic microphone volume level from current call Parameters speakerVolume Return the dynamic speaker volume by this parameter the range is 0 9 microphoneVolum Return the dynamic microphone volume by this parameter the range is 0 9 e Remarks Usually set a timer to call this function to refresh the volume level indicator Call functions long PortSIPSDK call sendSdp videoCall int PortSIPSDK rejectCall code rejectCall Reject the incoming call int PortSIPSDK answerCall videoCall answerCall Answer the incoming call int PortSIPSDK updateCall enableAudio enable Video Use the re INVITE to update the established call Functions L Make a call L int PortSIPSDK hangUp hangUp Hang up the call L L int PortSIPSDK hold To place a call on hold
100. utgoing audio stredam the remote side can t hearing 0 audio muteIncomingVide Set it to true to mute incoming video stredam can t see remote side video 0 muteOutgoingVide Set it to true to mute outgoing video stredam the remote side can t see video 0 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int forwardCall long sessionid forwardTo NSString forwardTo Forward call to another one when received the incoming call Parameters sessionId The session ID of the call forwardTo Target of the forward it can be sip number sipserver com or number only 31 Returns If the function succeeds the return value is 0 If the function fails the return value is a specific error code int sendDtmf long sessionid dtmfMethod DTMF_METHOD dtmfMethod code int code dtmfDration int dtmfDuration playDtmfTone BOOL playDimfTone Send DTMF tone Parameters sessionld The session ID of the call dtmfMethod Support send DTMF tone with two methods DIMF_RFC2833 and DTMF_INFO The DTMF_RFC2833 is recommend code The DTMF tone 0 16 code Description 0 The DTMF tone 0 1 The DTMF tone 1 2 The DTMF tone 2 3 The DTMF tone 3 4 The DTMF tone 4 5 The DTMF tone 5 6 The DTMF tone 6 7 The DTMF tone 7 8 The DTMF tone 8 9 The DTMF tone 9 10
101. w Set the window for a session that using to display the received remote video image int displayLocal Video Start stop to display the local video image int setVideoNackStatus Enable disable the NACK feature rfc6642 which help to improve the video quatliy void muteMicrophone Mute the device microphone it s unavailable for Android and iOS void muteSpeaker Mute the device speaker it s unavailable for Android and iOS int setAudioDeviceld outputDeviceld Set the audio device that will use for audio call void getDynamicVolumeLevel microphoneVolume Obtain the dynamic microphone volume level from current call long call sendSdp videoCall Make a call int rejectCall code rejectCall Reject the incoming call int hangUp hangUp Hang up the call int answerCall videoCall answerCall Answer the incoming call int updateCall enableAudio enable Video Use the re INVITE to update the established call ant hold To place a call on hold int unHold Take off hold int muteSession muteIncomingAudio muteOutgoing Audio muteIncoming Video muteOutgoing Video Mute the specified session audio or video int forwardCall forwardTo 75 Forward call to another one when received the incoming call int sendDtmf dtmfMethod code dtmfDration playDtmfTone Send DTMF tone int refer referTo Refer the currently call to another one

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