User Manual for Android

FAQ

Where can I download the PortSIP VoIP SDK for testing?

You can download the PortSIP VoIP SDK and sample projects from the PortSIP Website.

What Android API version is required?

The PortSIP VoIP SDK requires Android API version 16 or later.

How can I create a new project with the PortSIP VoIP SDK?

1. Download the Sample Project and SDK: Obtain the sample project and trial SDK from the PortSIP Website and extract them to a desired directory.

2. Create a New Project: In Android Studio, create a new Android Application project.

3. Add the SDK Libraries: Copy all files from the libs directory of the extracted SDK to the libs directory of your new application.

4. Import Necessary Classes: Import the required classes from the SDK:

   import com.portsip.OnPortSIPEvent;
   import com.portsip.PortSipSdk;

How can I process callback events?

1. Implement the OnPortSIPEvent Interface: Your class should implement the OnPortSIPEvent interface to handle callback events.

2. Override Callback Methods: Override the necessary callback methods (e.g., onRegistrationState, onCallState) to handle specific events.

How do I initialize the SDK?

1. Create an Instance: Create an instance of the PortSipSdk class.

2. Set the Event Listener: Set the event listener using setOnPortSIPEvent.

3. Create a Call Manager: Create a call manager using CreateCallManager.

4. Initialize the SDK: Call the initialize method with appropriate parameters.

Is the SDK thread-safe?

Yes, the PortSIP SDK is thread-safe. You can call API functions from multiple threads without worrying about synchronization issues. However, there are exceptions: the onAudioRawCallback, onVideoRawCallback, and onRTPPacketCallback callbacks should not be called directly from other threads.

SDK Callback events

Register events

void onRegisterSuccess(String reason, int code,String sipMessage);

When successfully registered to server, this event will be triggered.

Parameters

reason	The status text.
code	The status code.
sipMessage	The SIP message received.

void onRegisterFailure(String reason, int code,String sipMessage);

When failed to register to SIP server, this event will be triggered.

Parameters

reason	The status text.
code	The status code.
sipMessage	The SIP message received.

Call events

void onInviteIncoming(long sessionId,
                      String callerDisplayName,
                      String caller,
                      String calleeDisplayName,
                      String callee,
                      String audioCodecs,
                      String videoCodecs,
                      boolean existsAudio,
                      boolean existsVideo,
                      String sipMessage);

When a call is coming, this event will be triggered.

Parameters

sessionId	The session ID of the call.
callerDisplayNam e	The display name of caller
caller	The caller.
calleeDisplayNam e	The display name of callee.
callee	The callee.
audioCodecs	The matched audio codecs. It's separated by "#" if there are more than one codecs.
videoCodecs	The matched video codecs. It's separated by "#" if there are more than one codecs.
existsAudio	By setting to true, it means that this call include the audio.
existsVideo	By setting to true, it means that this call include the video.
sipMessage	The SIP message received.

void onInviteTrying(long sessionId);

If the outgoing call is being processed, this event will be triggered.

Parameters

sessionId

The session ID of the call.

void onInviteSessionProgress(long sessionId,
                             String audioCodecs,
                             String videoCodecs,
                             boolean existsEarlyMedia,
                             boolean existsAudio,
                             boolean existsVideo,
                             String sipMessage);

Once the caller received the "183 session progress" message, this event would be triggered.

Parameters

sessionId	The session ID of the call.
audioCodecs	The matched audio codecs. It's separated by "#" if there are more than one codecs.
videoCodecs	The matched video codecs. It's separated by "#" if there are more than one codecs.
existsEarlyMedia	By setting to true it means the call has early media.
existsAudio	By setting to true it means this call include the audio.
existsVideo	By setting to true it means this call include the video.
sipMessage	The SIP message received.

void onInviteRinging(long sessionId,
                                String statusText,
                                int statusCode,
                                String sipMessage);

If the outgoing call is ringing, this event will be triggered.

Parameters

sessionId	The session ID of the call.
statusText	The status text.
statusCode	The status code.
sipMessage	The SIP message received.

void onInviteAnswered(long sessionId,
                      String callerDisplayName,
                      String caller,
                      String calleeDisplayName,
                      String callee,
                      String audioCodecs,
                      String videoCodecs,
                      boolean existsAudio,
                      boolean existsVideo,
                      String sipMessage);

If the remote party answered the call, this event would be triggered.

Parameters

sessionId	The session ID of the call.
callerDisplayNam e	The display name of caller
caller	The caller.
calleeDisplayNam e	The display name of callee.
callee	The callee.
audioCodecs	The matched audio codecs. It's separated by "#" if there are more than one codecs.
videoCodecs	The matched video codecs. It's separated by "#" if there are more than one codecs.
existsAudio	By setting to true, this call includes the audio.
existsVideo	By setting to true, this call includes the video.
sipMessage	The SIP message received.

void onInviteFailure(long sessionId, String callerDisplayName,
                                String caller,
                                String calleeDisplayName,
                                String callee,
                                String reason,
                                int code,
                                String sipMessage);

This event will be triggered if the outgoing or incoming call fails.

Parameters

sessionId	The session ID of the call.
callerDisplayNam e	The display name of caller l
caller	The caller. l
calleeDisplayNam e	The display name of callee. l
callee	The callee.
reason	The failure reason.
code	The failure code.
sipMessage	The SIP message received.

public void onInviteUpdated(long sessionId,
                            String audioCodecs,
                            String videoCodecs,
                            String screenCodecs,
                            boolean existsAudio,
                            boolean existsVideo,
                            boolean existsScreen,
                            String sipMessage);

This event will be triggered when remote party updates the call.

Parameters

sessionId	The session ID of the call.
audioCodecs	The matched audio codecs. It's separated by "#" if there are more than one codecs.
videoCodecs	The matched video codecs. It's separated by "#" if there are more than one codecs.
screenCodecs	The matched screen codecs. It's separated by "#" if there are more than one codecs.
existsAudio	By setting to true, this call includes the audio.
existsVideo	By setting to true, this call includes the video.
existsScreen	By setting to true, this call includes the screen shared.
sipMessage	The SIP message received.

void onInviteConnected(long sessionId);

This event will be triggered when UAC sent/UAS received ACK (the call is connected). Some functions (hold, updateCall etc...) can be called only after the call connected, otherwise the functions will return error.

Parameters

sessionId

The session ID of the call.

void onInviteBeginingForward(String forwardTo);

If the enableCallForward method is called and a call is incoming, the call will be forwarded automatically and this event will be triggered.

Parameters

forwardTo

The target SIP URI of the call forwarding.

void onInviteClosed(long sessionId, String sipMessage);

This event is triggered once remote side ends the call.

Parameters

sessionId	The session ID of the call.
sipMessage	The SIP message received.

void onDialogStateUpdated(String BLFMonitoredUri,
                          String BLFDialogState,
                          String BLFDialogId,
                          String BLFDialogDirection);

If a user subscribed and his dialog status monitored, when the monitored user is holding a call or is being rang, this event will be triggered.

Parameters

BLFMonitoredUri	the monitored user's URI
BLFDialogState	- the status of the call
BLFDialogId	- the id of the call
BLFDialogDirecti on	- the direction of the call

void onRemoteHold(long sessionId);

If the remote side places the call on hold, this event will be triggered.

Parameters

sessionId

The session ID of the call.

void onRemoteUnHold(long sessionId,
                               String audioCodecs,
                               String videoCodecs,
                               boolean existsAudio,
                               boolean existsVideo);

If the remote side un-holds the call, this event will be triggered Parameters

sessionId	The session ID of the call.
audioCodecs	The matched audio codecs. It's separated by "#" if there are more than one codec.
videoCodecs	The matched video codecs. It's separated by "#" if there are more than one codec.
existsAudio	By setting to true, this call includes the audio.
existsVideo	By setting to true, this call includes the video.

Refer events

    public void onReceivedRefer(long sessionId,
                         long referId,
                         String to,
                         String from,
                         String referSipMessage);

This event will be triggered once received a REFER message.

Parameters

sessionId	The session ID of the call.
referId	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.

void onReferAccepted(long sessionId);

This callback will be triggered once remote side calls "acceptRefer" to accept the REFER Parameters

sessionId

The session ID of the call.

void onReferRejected(long sessionId, String reason, int code);

This callback will be triggered once remote side calls "rejectRefer" to reject the REFER Parameters

sessionId	The session ID of the call.
reason	Reject reason.
code	Reject code.

void onTransferTrying(long sessionId);

When the refer call is being processed, this event will be triggered.

Parameters

sessionId

The session ID of the call.

void onTransferRinging(long sessionId);

When the refer call is ringing, this event will be triggered.

Parameters

sessionId

The session ID of the call.

void onACTVTransferSuccess(long sessionId);

When the refer call succeeds, this event will be triggered. The ACTV means Active. For example, A establishes the call with B, A transfers B to C, C accepts the refer call, and A will receive this event.

Parameters

sessionId

The session ID of the call.

void onACTVTransferFailure(long sessionId, String reason, int code);

When the refer call fails, this event will be triggered. The ACTV means Active. For example, A establish the call with B, A transfers B to C, C rejects this refer call, and A will receive this event.

Parameters

sessionId	The session ID of the call.
reason	The error reason.
code	The error code.

Signaling events

void onReceivedSignaling(long sessionId, String message);

This event will be triggered when receiving a SIP message. This event is disabled by default. To enable, use enableCallbackSignaling.

Parameters

sessionId	The session ID of the call.
message	The received SIP message.

void onSendingSignaling (long sessionId, String  message)

This event will be triggered when sent a SIP message. This event is disabled by default. To enable, use enableCallbackSignaling.

Parameters

sessionId	The session ID of the call.
message	The sent SIP message.

MWI events

void onWaitingVoiceMessage(String messageAccount,
                           int urgentNewMessageCount,
                           int urgentOldMessageCount,
                           int newMessageCount,
                           int oldMessageCount);

If there is the waiting voice message (MWI), this event will be triggered.

Parameters

messageAccount	Voice message account
urgentNewMessag eCount	Count of new urgent messages.
urgentOldMessage Count	Count of history urgent message.
newMessageCount	Count of new messages.
oldMessageCount	Count of history messages.

void onWaitingFaxMessage(String messageAccount,
                         int urgentNewMessageCount,
                         int urgentOldMessageCount,
                         int newMessageCount,
                         int oldMessageCount);

If there is waiting fax message (MWI), this event will be triggered.

Parameters

messageAccount	Fax message account
urgentNewMessag eCount	Count of new urgent messages.
urgentOldMessage Count	Count of history urgent messages.
newMessageCount	Count of new messages.
oldMessageCount	Count of old messages.

DTMF events

void onRecvDtmfTone(long sessionId, int tone);

This event will be triggered when receiving a DTMF tone from remote side.

Parameters

sessionId	Session ID of the call.
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.
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

void onRecvOptions(String optionsMessage);

This event will be triggered when receiving the OPTIONS message.

Parameters

optionsMessage

The received whole OPTIONS message in text format.

void onRecvInfo(String infoMessage);

This event will be triggered when receiving the INFO message.

Parameters

infoMessage

The whole INFO message received in text format.

void onRecvNotifyOfSubscription(long subscribeId,String notifyMessage,
                                byte[] messageData,
                                int messageDataLength);

This event will be triggered when receiving a NOTIFY message of the subscription.

Parameters

subscribeId	The ID of SUBSCRIBE request.
notifyMessage	The received INFO message in text format.
messageData	The received message body. It's can be either text or binary data.
messageDataLengt h	The length of "messageData".

Presence events

void onPresenceRecvSubscribe(long subscribeId,
                             String fromDisplayName,
                             String from,
                             String subject);

This event will be triggered when receiving the SUBSCRIBE request from a contact.

Parameters

subscribeId	The ID of SUBSCRIBE request.
fromDisplayName	The display name of contact.
from	The contact who sends the SUBSCRIBE request.
subject	The subject of the SUBSCRIBE request.

void onPresenceOnline(String fromDisplayName,
                      String from,
                      String stateText);

When the contact is online or changes presence status, this event will be triggered.

Parameters

fromDisplayName	The display name of contact.
from	The contact who sends the SUBSCRIBE request.
stateText	The presence status text.

void onPresenceOffline(String fromDisplayName,
                       String from);

When the contact is offline, this event will be triggered.

Parameters

fromDisplayName	The display name of contact.
from	The contact who sends the SUBSCRIBE request

void onRecvMessage(long sessionId,
                   String mimeType,
                   String subMimeType,
                   byte[] messageData,
                   int messageDataLength);

This event will be triggered when receiving 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 can be text or binary data. Use the mimeType and subMimeType to differentiate them. For example, if the mimeType is "text" and subMimeType is "plain", "messageData" is text message body. If the mimeType is "application" and subMimeType is "vnd.3gpp.sms", "messageData" is binary message body.
messageDataLengt h	The length of "messageData".

void onRecvOutOfDialogMessage(String fromDisplayName,
                              String from,
                              String toDisplayName,
                              String to,
                              String mimeType,
                              String subMimeType,
                              byte[] messageData,
                              int messageDataLength,
                              String sipMessage);

This event will be triggered when receiving a MESSAGE message out of dialog. For example pager 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 can be text or binary data. Use the mimeType and subMimeType to differentiate them. For example, if the mimeType is "text" and subMimeType is "plain", "messageData" is text message body. If the mimeType is "application" and subMimeType is "vnd.3gpp.sms", "messageData" is binary message body.
messageDataLengt h	The length of "messageData".
sipMessage	The SIP message received.

void onSendMessageSuccess(long sessionId, long messageId,String sipMessage);

If the message is sent successfully in dialog, this event will be triggered.

Parameters

sessionId	The session ID of the call.
messageId	The message ID. It's equal to the return value of sendMessage function.
sipMessage	The SIP message received.

void onSendMessageFailure(long sessionId,
                          long messageId,
                          String reason,
                          int code,
                          String sipMessage);

If the message is failed to be sent out of dialog, this event will be triggered.

Parameters

sessionId

The session ID of the call.

messageId	The message ID. It's equal to the return value of sendMessage function.
reason	The failure reason.
code	Failure code.
sipMessage	The SIP message received.

void onSendOutOfDialogMessageSuccess( long messageId,
                                      String fromDisplayName,
                                      String from,
                                      String toDisplayName,
                                      String to,
                                      String sipMessage);

If the message is sent successfully out of dialog, this event will be triggered.

Parameters

messageId	The message ID. It's equal to 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.
sipMessage	The SIP message received.

void onSendOutOfDialogMessageFailure(long messageId,
                                     String fromDisplayName,
                                     String from,
                                     String toDisplayName,
                                     String to,
                                     String reason,
                                     int code,
                                     String sipMessage);

If the message failed to be sent out of dialog, this event would be triggered.

Parameters

messageId	The message ID. It's equal to 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.
sipMessage	The SIP message received.

void onSubscriptionFailure(long subscribeId, int statusCode);

This event will be triggered on sending SUBSCRIBE failure.

Parameters

subscribeId	The ID of SUBSCRIBE request.
statusCode	The status code.

void onSubscriptionTerminated(long subscribeId);

This event will be triggered when a SUBSCRIPTION is terminated or expired.

Parameters

subscribeId

The ID of SUBSCRIBE request.

audio device changed,Play audio and video file finished events

void onPlayFileFinished(long sessionId, String fileName);

If called startPlayingFileToRemote function with no loop mode, this event will be triggered once the file play finished.

Parameters

sessionId	The session ID of the call.
fileName	The play file name.

void onStatistics(long sessionId,String statistics);

If called getStatistics function, this event will be triggered once the statistics get finished.

Parameters

sessionId	The session ID of the call.
statistics	The session call statistics.

void onAudioDeviceChanged(PortSipEnumDefine.AudioDevice audioDevice, Set<PortSipEnumDefine.AudioDevice> devices);

fired When available audio devices changed or audio device currently in use changed.

Parameters

audioDevice	device currently in use
devices	devices useable. If a wired headset is connected, it should be the only possible option. When no wired headset connected, the devices set may contain speaker, earpiece, Bluetooth devices. can be set by PortSipSdk#setAudioDevice to switch to current device.

void onAudioFocusChange(int focusChange);

fired when the audio focus has been changed.

Parameters

focusChange

the type of focus change, one of AudioManager::AUDIOFOCUS_GAIN, AudioManager::AUDIOFOCUS_LOSS, AudioManager::AUDIOFOCUS_LOSS_TRANSIENT and AudioManager::AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK.

RTP callback events

void onRTPPacketCallback(long sessionId,
                         int mediaType,
                         int enum_direction,
                         byte[] RTPPacket, 
                         int packetSize);

If enableRtpCallback function is called to enable the RTP callback, this event will be triggered once a RTP packet is received or sent.

Parameters

sessionId	The session ID of the call.
mediaType	RTP packet media type, 0 for audio, 1 for video , 2 for screen.
enum_direction	RTP packet direction enum_DIRECTION_SEND, enum_DIRECTION_RECV.
RTPPacket	The received or sent RTP packet.
packetSize	The size of the RTP packet in bytes.

Remarks

Donot call any SDK API functions in this event directly. If you want to call the API functions or other code which is time-consuming, 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

void onAudioRawCallback(long sessionId,
                        int enum_direction,
                        byte[] data, 
                        int dataLength,
                        int samplingFreqHz);

This event will be triggered once receiving the audio packets if called enableAudioStreamCallback function.

Parameters

sessionId	The session ID of the call.
enum_direction	The type passed in enableAudioStreamCallback function. Below types allowed: enum_DIRECTION_SEND, enum_DIRECTION_RECV.
data	The memory of audio stream. It's in PCM format.
dataLength	The data size.
samplingFreqHz	The audio stream sample in HZ. For example, 8000 or 16000.

Remarks

Don't call any SDK API functions in this event directly. If you want to call the API functions or other code which is time-consuming, you should post a message to another thread and execute SDK API functions or other code in another thread.

See also

PortSipSdk::enableAudioStreamCallback

void onVideoRawCallback(long sessionId,
                        int enum_direction, 
                        int width, 
                        int height,
                        byte[] data, 
                        int dataLength);

This event will be triggered once receiving the video packets if enableVideoStreamCallback function is called.

Parameters

sessionId	The session ID of the call.
enum_direction	The type which is passed in enableVideoStreamCallback function. Below types allowed: enum_DIRECTION_SEND, enum_DIRECTION_RECV.
width	The width of video image.
height	The height of video image.
data	The memory of video stream. It's in YUV420 format, YV12.
dataLength	The data size.

See also

PortSipSdk::enableVideoStreamCallback

SDK functions

Initialize and register functions

int initialize(int enum_transport, 
						  String localIP, 
						  int localSIPPort, 
						  int enum_LogLevel,
						  String LogPath, 
						  int maxLines, 
						  String agent,
						  int audioDeviceLayer, 
						  int videoDeviceLayer, 
						  String TLSCertificatesRootPath,
						  String TLSCipherList, 
						  boolean verifyTLSCertificate, 
						  String dnsServers)

Initialize the SDK.

Parameters

enum_transport	Transport for SIP signaling, which can be set as: ENUM_TRANSPORT_UDP, ENUM_TRANSPORT_TCP, ENUM_TRANSPORT_TLS,
localIP	The local PC IP address (for example: 192.168.1.108). It will be used for sending and receiving SIP messages and RTP packets. ` `If the local IP is provided in IPv6 format, the SDK will use IPv6. ` `If you want the SDK to choose correct network interface (IP) automatically, please use "0.0.0.0" for IPv4, or "::" for IPv6.
localSIPPort	The listening port for SIP message transmission, for example 5060.
enum_LogLevel	Set the application log level. The SDK will generate "PortSIP_Log_datatime.log" file if the log is enabled. ENUM_LOG_LEVEL_NONE ENUM_LOG_LEVEL_DEBUG
ENUM_LOG_LEVEL_ERROR ENUM_LOG_LEVEL_WARNING
ENUM_LOG_LEVEL_INFO ENUM_LOG_LEVEL_DEBUG
LogPath	The path for storing log file. The path (folder) specified MUST be existent.
maxLines	Theoretically, unlimited count of lines are supported depending on the device capability. For SIP client, it is recommended to limit it as ranging 1 - 100.
agent	The User-Agent header to be inserted in to SIP messages.
audioDeviceLayer	Specifies the audio device layer that should be using: 0 = Use the OS defaulted device.

	1 = Virtual device, usually use this for the device that has no sound device installed.
videoDeviceLayer	Specifies the video device layer that should be using: 0 = Use the OS defaulted device. 1 = Use Virtual device, usually use this for the device that has no camera installed.
TLSCertificatesRo otPath	Specify the TLS certificate path, from which the SDK will load the certificates automatically. Note: On Windows, this path will be ignored, and SDK will read the certificates from Windows certificates stored area instead.
TLSCipherList	Specify the TLS cipher list. This parameter is usually passed as empty so that the SDK will offer all available ciphers.
verifyTLSCertificate	Indicate if SDK will verify the TLS certificate or not. By setting to false, the SDK will not verify the validity of TLS certificate.
dnsServers	Additional Nameservers DNS servers. Value null indicates system DNS Server. Multiple servers will be split by ";", e.g "8.8.8.8;8.8.4.4"

Returns

If the function succeeds, it returns value 0. If the function fails, it will return a specific error code

void unInitialize() 

Un-initialize the SDK and release resources.

int setInstanceId(String instanceId)

Set the instance Id, the outbound instanceId((RFC5626) ) used in contact headers.

Parameters

instanceId

The SIP instance ID. If this function is not called, the SDK will generate an instance ID automatically. The instance ID MUST be unique on the same device (device ID or IMEI ID is recommended). Recommend to call this function to set the ID on Android devices.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setUser(String userName, 
					   String displayName, 
					   String authName,
					   String password, 
					   String userDomain, 
					   String SIPServer, 
					   int SIPServerPort,
					   String STUNServer, 
					   int STUNServerPort, 
					   String outboundServer,
					   int outboundServerPort)

Set user account info.

Parameters

userName	Account (username) of the SIP, usually provided by an IP-Telephony service provider.
displayName	The name displayed. You can set it as your like, such as "James Kend". It's optional.
authName	Authorization user name (usually equal to the username).
password	User's password. It's optional.
userDomain	User domain; this parameter is optional, which allows to transfer an empty string if you are not using the 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 for NAT traversal. It's optional and can be used to transfer 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. It's optional and allows to transfer an empty string if not using the outbound server.
outboundServerPort	Outbound proxy server port, it will be ignored if the outboundServer is empty.

Returns

If this function succeeds, it will return value 0. If it fails, it will return a specific error code.

void removeUser() 

remove user account info.

int registerServer(int expires, int retryTimes)

Register to SIP proxy server (login to server)

Parameters

expires	Time interval for registration refreshment, in seconds. The maximum of supported value is 3600. It will be inserted into SIP REGISTER message headers.
retryTimes	The maximum of retry attempts if failed to refresh the registration. By setting to <= 0, the attempt will be disabled and onRegisterFailure callback will be triggered when facing retry failure.

Returns

If this function succeeds, it will return value 0. If fails, it will return a specific error code.

If the registration to server succeeds, onRegisterSuccess will be triggered; otherwise onRegisterFailure will be triggered.

int refreshRegistration(int expires) 

Refresh the registration manually after successfully registered.

Parameters

expires

Time interval for registration refreshment, in seconds. The maximum of supported value is 3600. It will be inserted into SIP REGISTER message headers.

Returns

If this function succeeds, it will return value 0. If fails, it will return a specific error code.

If the registration to server succeeds, onRegisterSuccess will be triggered; otherwise onRegisterFailure will be triggered.

int unRegisterServer(int waitMS)

Un-register from the SIP proxy server.

Parameters

waitMS

Wait for the server to reply that the un-registration is successful, waitMS is the longest waiting milliseconds, 0 means not waiting.

Returns

If this function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setDisplayName(String displayName)

Set the display name of user.

Parameters

displayName

That will appear in the From/To Header.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setLicenseKey (String  key)

Set the license key. It must be called before setUser function.

Parameters

key	The SDK license key. Please purchase from PortSIP.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Audio and video codecs functions

int addAudioCodec (int  enum_audiocodec)

Enable an audio codec, and it will be shown in SDP.

Parameters

enum_audiocodec

ENUM_AUDIOCODEC_ISACSWB, ENUM_AUDIOCODEC_OPUS, ENUM_AUDIOCODEC_DTMF.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int addVideoCodec (int  enum_videocodec)

Enable a video codec, and it will be shown in SDP.

Parameters

enum_videocodec

Video codec type. Supported types include enum_VIDEOCODEC_H264, enum_VIDEOCODEC_VP8. enum_VIDEOCODEC_VP9.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

**boolean isAudioCodecEmpty ()

Detect if the audio codecs are enabled.

Returns

If no audio codec enabled, it will return value true; otherwise it returns false.

**boolean isVideoCodecEmpty ()

Detect if the video codecs are enabled.

Returns

If no video codec enabled, it will return value true; otherwise it returns false.

int setAudioCodecPayloadType (int  enum_audiocodec, int  payloadType)

Set the RTP payload type for dynamic audio codec.

Parameters

enum_audiocodec	Audio codec type, which is defined in the PortSIPTypes file.
payloadType	The new RTP payload type that you want to set.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoCodecPayloadType (int  enum_videocodec, int  payloadType)

Set the RTP payload type for dynamic video codec.

Parameters

enum_videocodec	Video codec type. Supported types include: enum_VIDEOCODEC_H264, enum_VIDEOCODEC_VP8. enum_VIDEOCODEC_VP9.
payloadType	The new RTP payload type that you want to set.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void clearAudioCodec ()

Remove all the enabled audio codecs.

void clearVideoCodec ()

Remove all the enabled video codecs.

int setAudioCodecParameter (int  enum_audiocodec, String  sdpParameter)

Set the codec parameter for audio codec.

Parameters

enum_audiocodec	Audio codec type, defined in the PortSIPTypes file.
sdpParameter	The parameter is in string format.
-	-

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

See also

PortSipEnumDefine

Remarks

Example:

setAudioCodecParameter(AUDIOCODEC_AMR, "mode-set=0; octet-align=1; robust-sorting=0"![])

int setVideoCodecParameter (int  enum_videocodec, String  sdpParameter)

Set the codec parameter for video codec.

Parameters

enum_videocodec	Video codec types. Supported types include: enum_VIDEOCODEC_H264, enum_VIDEOCODEC_VP8. enum_VIDEOCODEC_VP9.
sdpParameter	The parameter is in string format.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

Example:

setVideoCodecParameter(PortSipEnumDefine.enum_VIDEOCODEC_H264, profile-level-id=420033; packetization-mode=0");

Additional settings functions

String getVersion ()

Get the version number of the current SDK.

Returns

String with version description

int enableRport (boolean  enable)

Enable/Disable rport(RFC3581).

Parameters

enable

enable Set to true to enable the SDK to support rport. By default it is enabled.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int enableEarlyMedia (boolean  enable)

Enable/disable rport(RFC3581).

Parameters

enable

Set to true to enable the SDK to support rport. By default it is enabled.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code. Enable/Disable Early Media.

Parameters

enable

Set to true to enable the SDK support Early Media. By default the Early Media is disabled.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int enablePriorityIPv6Domain (boolean  enable)

Enable/disable which allows specifying the preferred protocol when a domain supports

both IPV4 and IPV6 simultaneously.

Parameters

enable

Set to true to enable priority IPv6 Domain. with the default priority being IPV4.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setUriUserEncoding (String  character, boolean  enable)

Modifies the default URI user character needs to be escaped.

Parameters

character	The character to be modified, set one character at a time.
enable	Whether escaping is required, true for allowing escaping, false for disabling escaping.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setReliableProvisional (int  mode)

Enable/Disable PRACK.

Parameters

mode	Modes work as follows: 0 - Never, Disable PRACK,By default the PRACK is disabled. 1 - SupportedEssential, Only send reliable provisionals if sending a body and far end supports. 2 - Supported, Always send reliable provisionals if far end supports. 3 - Required Always send reliable provisionals.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int enable3GppTags (boolean  enable)

Enable/disable the 3Gpp tags, including "ims.icsi.mmtel" and "g.3gpp.smsip".

Parameters

enable

Set to true to enable 3Gpp tags for SDK.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void enableCallbackSignaling (boolean   enableSending, boolean  enableReceived)

Enable/disable the callback of the SIP messages.

Parameters

enableSending	Set as true to enable to callback the sent SIP messages, or false to disable. Once enabled, the "onSendingSignaling" event will be triggered when the SDK sends a SIP message.
enableReceived	Set as true to enable to callback the received SIP messages, or false to disable. Once enabled, the "onReceivedSignaling" event will be triggered when the SDK receives a SIP message.

void setSrtpPolicy (int  enum_srtppolicy)

Set the SRTP policy.

Parameters

enum_srtppolicy

The SRTP policy.allow: enum_SRTPPOLICY_NONE, enum_SRTPPOLICY _FORCE, enum_SRTPPOLICY_PREFER.

int setRtpPortRange (int  minimumRtpPort, int  maximumRtpPort)

This function allows to set the RTP port range for audio and video streaming.

Parameters

minimumRtpPort	The minimum RTP port.
maximumRtpPort	The maximum RTP port.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

The port range ((max - min) / maxCallLines) should be greater than 4.

int enableCallForward (boolean  forBusyOnly, String   forwardTo)

Enable call forwarding.

Parameters

forBusyOnly	If this parameter is set to true, the SDK will forward incoming calls when the user is currently busy. If set it to false, SDK will forward all incoming calls.
forwardTo	The target to which the call will be forwarded. It must be in the format of sip:[e-mail].

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int disableCallForward ()

Disable the call forwarding. The SDK will not forward any incoming call when this function is called.

Returns

If the function succeeds, it will not return value 0. If the function fails, it will return a specific error code.

int enableSessionTimer (int  timerSeconds)

This function allows to periodically refresh Session Initiation Protocol (SIP) sessions by

sending repeated INVITE requests.

Parameters

timerSeconds

The value of the refresh interval in seconds. A minimum of 90 seconds required.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

The repeated INVITE requests, or re-INVITEs, are sent during an active call log to allow user agents (UA) or proxies to determine the status of a SIP session. Without this keep-alive mechanism, proxies that remember incoming and outgoing requests (stateful proxies) may continue to retain call state in vain. If a UA fails to send a BYE message at the end of a session, or if the BYE message is lost due to network problems, a stateful proxy will not know that the session has ended. The re-INVITES ensure that active sessions stay active and completed sessions are terminated.

void disableSessionTimer ()

Disable the session timer.

void setDoNotDisturb (boolean  state)

Enable/disable the "Do not disturb" status.

Parameters

state

If it is set to true, the SDK will reject all incoming calls.

void enableAutoCheckMwi (boolean  state)

Enable/disable the "Auto Check MWI" status.

Parameters

state

If it is set to true, the SDK will check Mwi automatically.

int setRtpKeepAlive (boolean  state, int  keepAlivePayloadType, int  deltaTransmitTimeMS)

Enable or disable to send RTP keep-alive packet when the call is ongoing.

Parameters

state	When it's set to true, it's allowed to send the keep-alive packet during the conversation;
keepAlivePayload Type	The payload type of the keep-alive RTP packet. It's usually set to 126.
deltaTransmitTime MS	The interval for sending keep-alive RTP packet, in millisecond. Recommended value ranges 15000 - 300000.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setKeepAliveTime (int  keepAliveTime)

Enable or disable to send SIP keep-alive packet.

Parameters

keepAliveTime

This is the time interval for SIP keep-alive, in seconds. When it is set to 0, the SIP keep-alive will be disabled. Recommended value is 30 or 50.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setAudioSamples (int  ptime, int  maxptime)

Set the audio capture sample, which will be present in the SDP of INVITE and 200 OK message as "ptime and "maxptime" attribute.

Parameters

ptime	It should be a multiple of 10 between 10 - 60 (included 10 and 60).
maxptime	The "maxptime" attribute should be a multiple of 10 between 10 - 60 (included 10 and 60). It can't be less than "ptime".

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int addSupportedMimeType (String  methodName, String  mimeType, String  subMimeType)

Set the SDK to receive SIP messages that include special mime type.

Parameters

methodName	Method name of the SIP message, such as INVITE, OPTION, INFO, MESSAGE, UPDATE, ACK etc. For more details please refer to RFC3261.
mimeType	The mime type of SIP message.
subMimeType	The sub mime type of SIP message.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

In default, PortSIP VoIP SDK supports media types (mime types) included 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 to receive SIP messages that include above mime types. Now if remote side send an INFO SIP message with its "Content-Type" header value "text/plain", SDK will reject this INFO message, because "text/plain" of INFO message is not included in the default type list. How should we enable the SDK to receive SIP INFO messages that include "text/plain" mime type? The answer is addSupportedMimyType:

addSupportedMimeType("INFO", "text", "plain");

If the user wishes to receive the NOTIFY message with "application/media_control+xml", it should be set as below:

addSupportedMimeType("NOTIFY", "application", "media_control+xml");

For more details about the mime type, please visit: http://www.iana.org/assignments/media-types/

Access SIP message header functions

String getSipMessageHeaderValue (String  sipMessage, String  headerName)

Access the SIP header of SIP message.

Parameters

sipMessage	The SIP message.
headerName	The header of which user wishes to access the SIP message.

Returns

String. The SIP header of SIP message.

long addSipMessageHeader(long sessionId, 
									       String methodName, 
									       int msgType, 
									       String headerName,
									       String headerValue)

Add the SIP Message header into the specified outgoing SIP message.

Parameters

sessionId	Add the header to the SIP message with the specified session Id only. By setting to -1, it will be added to all messages.
methodName	Add the header to the SIP message with specified method name only. For example: "INVITE", "REGISTER", "INFO" etc. If "ALL" specified, it will add all SIP messages.
msgType	1 refers to apply to the request message, 2 refers to apply to the response message, 3 refers to apply to both request and response.
headerName	The header name which will appear in SIP message.
headerValue	The custom header value.

Returns

If the function succeeds, it will return the addedSipMessageId , which is greater than 0. If the function fails, it will return a specific error code.

int removeAddedSipMessageHeader (long  addedSipMessageId)

Remove the headers (custom header) added by addSipMessageHeader.

Parameters

addedSipMessageI d

The addedSipMessageId return by addSipMessageHeader.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void clearAddedSipMessageHeaders ()

Clear the added extension headers (custom headers)

Remarks

For example, we have added two custom headers into every outgoing SIP message and want to have them removed.

addSipMessageHeader(-1,"ALL",3,"Blling", "usd100.00");
addSipMessageHeader(-1,"ALL",3,"ServiceId", "8873456"); 
clearAddedSipMessageHeaders(); 

If this function is called, the added extension headers will no longer appear in outgoing SIP message.

long modifySipMessageHeader(long sessionId, 
									          String methodName, 
									          int msgType,
									          String headerName, 
									          String headerValue)

Modify the special SIP header value for every outgoing SIP message.

Parameters

sessionId	The header to the SIP message with the specified session Id. By setting to -1, it will be added to all messages.
methodName	Modify the header to the SIP message with specified method name only. For example: "INVITE", "REGISTER", "INFO" etc. If "ALL" specified, it will add all SIP messages.
msgType	1 refers to apply to the request message, 2 refers to apply to the response message, 3 refers to apply to both request and response.
headerName	The SIP header name of which the value will be modified.
headerValue	The heaver value to be modified.

Returns

If the function succeeds, it will return modifiedSipMessageId, which is greater than 0. If the function fails, it will return a specific error code.

Remarks

Example: modify "Expires" header and "User-Agent" header value for every outgoing SIP message:

modifySipMessageHeader(-1,"ALL",3, "Expires", "1000"); modifySipMessageHeader(-1,"ALL",3, "User-Agent", "MyTest Softphone 1.0");

int removeModifiedSipMessageHeader (long  modifiedSipMessageId)

Remove the headers (custom header) added by modifiedSipMessageId.

Parameters

modifiedSipMessa geId

The modifiedSipMessageId return by modifySipMessageHeader.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void clearModifiedSipMessageHeaders ()

Clear the modify headers value. Once cleared, it will no longer modify every outgoing SIP message header values.

Remarks Example: modify two headers value for every outgoing SIP message and then clear it:

  modifySipMessageHeader(-1,"ALL",3, "Expires", "1000"); 
  modifySipMessageHeader(-1,"ALL",3, "User-Agent", "MyTest Softphone 1.0");   
  cleaModifyHeaders();

Parameters

Audio and video functions

int setVideoDeviceId (int  deviceId)

Set the video device that will be used for video call. Parameters

deviceId

Device ID (index) for video device (camera).

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoOrientation (int  rotation)

Setting the video Device Orientation.

Parameters

rotation

Device Orientation for video device (camera), e.g 0,90,180,270.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int enableVideoHardwareCodec (boolean  enableHWEncoder, boolean  enableHWDecoder)

Set enable/disable video Hardware codec.

enableHWEncoder	If it is set to true, the SDK will use video hardware encoder when available. By default it is true.
enableHWDecoder	If it is set to true, the SDK will use video hardware decoder when available. By default it is true.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoResolution (int  width, int  height)

Set the video capturing resolution.

Parameters

width	Video resolution, width
height	Video resolution, height

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setAudioBitrate (long  sessionId, int  enum_audiocodec, int  bitrateKbps)

Set the audio bitrate. Parameters

sessionId	The session ID of the call.
enum_audiocodec	Audio codec type allowed: enum_AUDIOCODEC_OPUS
bitrateKbps	The Audio bitrate in KBPS.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoBitrate (long  sessionId, int  bitrateKbps)

Set the video bitrate.

Parameters

sessionId	The session ID of the call.
bitrateKbps	The video bitrate in KBPS.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoFrameRate (long  sessionId, int  frameRate)

Set the video frame rate. Usually you do not need to call this function to set the frame rate since the SDK uses default frame rate.

Parameters

sessionId	The session ID of the call.
frameRate	The frame rate value, with its minimum of 5, and maximum value of 30. The greater the value is, the better video quality enabled and more bandwidth required;

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int sendVideo (long  sessionId, boolean  send)

Send the video to remote side.

Parameters

sessionId	The session ID of the call.
send	Set to true to send the video, or false to stop sending.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setRemoteVideoWindow (long  sessionId, PortSIPVideoRenderer  renderer)

Set the window for a session that is used for displaying the received remote video image.

Parameters

sessionId	The session ID of the call.
renderer	SurfaceView a SurfaceView for displaying the received remote video image.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setRemoteScreenWindow (long  sessionId, PortSIPVideoRenderer  renderer)

Set the window for a session that is used for displaying the received remote screen image.

Parameters

sessionId	The session ID of the call.
renderer	SurfaceView a SurfaceView for displaying the received remote screen image.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void displayLocalVideo (boolean   state, boolean  mirror, PortSIPVideoRenderer  renderer)

Start/stop displaying the local video image.

Parameters

state	Set to true to display local video image.
mirror	Set to true to display the mirror image of local video.
renderer	SurfaceView a SurfaceView for displaying local video image from camera.

int setVideoNackStatus (boolean  state)

Enable/disable the NACK feature (rfc6642) which helps to improve the video quality.

Parameters

state

Set to true to enable.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setChannelOutputVolumeScaling (long  sessionId, int  scaling)

Set a volume |scaling| to be applied to the outgoing signal of a specific audio channel.

47 Parameters

sessionId	The session ID of the call.
scaling	Valid scale ranges [0, 1000]. Default is 100.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setChannelInputVolumeScaling (long  sessionId, int  scaling)

Set a volume |scaling| to be applied to the microphone signal of a specific audio channel.

Parameters

sessionId	The session ID of the call.
scaling	Valid scale ranges [0, 1000]. Default is 100.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void enableAudioManager (boolean   state)

enable/disable sdk audio manager,when enable sdk will auto manager audio device input/output. if the state is enabled, the onAudioDeviceChanged event will be triggered when available audio devices changed or audio device currently in use changed .

Parameters

state

@true enable sdk audio manager @false disable audio manager

Set<PortSipEnumDefine.AudioDevice> getAudioDevices ()

Get current set of available/selectable audio devices.

Returns

Current set of available/selectable audio devices.

int setAudioDevice (PortSipEnumDefine.AudioDevice  defaultDevice)

Set the audio device that will used for audio call. For Android and iOS, switch between earphone and Loudspeaker allowed.

Parameters

defaultDevice

Set to true the SDK use loudspeaker for audio call, this just available for mobile platform only.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Call functions

long call (String  callee, boolean  sendSdp, boolean  videoCall)

Make a call

Parameters

callee	The callee. It can be a name only or full SIP URI, for example: user001 or sip:[e-mail] or sip:[e-mail]:5068
sendSdp	If it is set to false, the outgoing call will not include the SDP in INVITE message.
videoCall	If it is set to true and at least one video codec was added, the outgoing call will include the video codec into SDP. Otherwise no video codec will be added into outgoing SDP.

Returns

If the function succeeds, it will return the session ID of the call, which is greater than 0. If the function fails, it will return 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  sessionId, int  code)

rejectCall Reject the incoming call.

Parameters

sessionId	The session ID of the call.
code	Reject code, for example, 486, 480 etc.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int hangUp (long  sessionId)

hangUp Hang up the call.

Parameters

sessionId

Session ID of the call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int answerCall (long  sessionId, boolean  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 does not include video codec answer anyway.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int updateCall (long  sessionId, boolean  enableAudio, boolean  enableVideo)

updateCall 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 updated call, or false to disable audio in updated call.
enableVideo	Set to true to allow the video in update call, or false to disable video in updated call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return specific error code.

Remarks

Example usage:

Example 1: A called B with the audio only, and B answered A, there would be an audio conversation between A and B. Now A want to see B through video, A could use these functions to fulfill it.

clearVideoCodec(); addVideoCodec(VIDEOCODEC_H264); updateCall(sessionId, true, true);

Example 2: Remove video stream from the current conversation. updateCall(sessionId, true, false);

int hold (long  sessionId)

To place a call on hold.

Parameters

sessionId

The session ID of call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int unHold (long  sessionId)

Take off hold.

Parameters

sessionId

The session ID of call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int muteSession(long sessionId, 
						    boolean muteIncomingAudio,
						    boolean muteOutgoingAudio, 
						    boolean muteIncomingVideo,
						    boolean muteOutgoingVideo) 

Mute the specified audio or video session.

Parameters

sessionId	The session ID of the call.
muteIncomingAudi o	Set it to true to mute incoming audio stream. Once set, remote side audio cannot be heard.
muteOutgoingAudi o	Set it to true to mute outgoing audio stream. Once set, the remote side cannot hear the audio.
muteIncomingVide o	Set it to true to mute incoming video stream. Once set, remote side video cannot be seen.
muteOutgoingVide o	Set it to true to mute outgoing video stream, the remote side cannot see the video.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int forwardCall (long  sessionId, String   forwardTo)

Forward call to another one when receiving the incoming call.

Parameters

sessionId	The session ID of the call.
forwardTo	Target of the forward. It can be either "sip:[e-mail]" or "number".

Returns

If the function succeeds, it will return value 0. If the function fails, it will return value a specific error code.

long pickupBLFCall (String  replaceDialogId, boolean  videoCall)

This function will be used for picking up a call based on the BLF (Busy Lamp Field) status.

Parameters

replaceDialogId	The ID of the call which will be pickup. It comes with onDialogStateUpdated callback.
videoCall	Indicates pickup video call or audio call

If the function succeeds, it will return a session ID that is greater than 0 to the new call, otherwise returns a specific error code that is less than 0.

Remarks

The scenario is:

1. User 101 subscribed the user 100's call status: sendSubscription(mSipLib, "100", "dialog");

2. When 100 holds a call or 100 is ringing, onDialogStateUpdated callback will be triggered, and 101 will receive this callback. Now 101 can use pickupBLFCall function to pick the call rather than 100 to talk with caller.

int sendDtmf (long  sessionId, 
              int  enum_dtmfMethod, 
              int  code, 
              int  dtmfDuration, 
              boolean  playDtmfTone)

Send DTMF tone.

Parameters

sessionId	The session ID of the call.
enum_dtmfMethod	DTMF tone could be sent via two methods: DTMF_RFC2833 or DTMF_INFO. The DTMF_RFC2833 is recommend.
code	The DTMF tone. Values include:
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	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. Recommended value 160.
playDtmfTone	Set to true the SDK play local DTMF tone sound during send DTMF.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Refer functions

int refer (long  sessionId, String  referTo)

Transfer the current call to another callee.

Parameters

sessionId	The session ID of the call.
referTo	Target callee of the transfer. It can be either "sip:[e-mail]" or "number".

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

refer(sessionId, "sip:[e-mail]");

You can refer to the video on Youtube at:

https://www.youtube.com/watch?v=_2w9EGgr3FY, which will demonstrate how to complete the transfer.

int attendedRefer (long  sessionId, long  replaceSessionId, String  referTo)

Make an attended refer. Parameters

sessionId	The session ID of the call.
replaceSessionId	Session ID of the replace call.
referTo	Target callee of the refer. It can be either "sip:[e-mail]" or "number".

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

Please read the sample project source code to get more details, or you can refer to the video on YouTube at:

https://www.youtube.com/watch?v=_2w9EGgr3FY

Note: Please use Windows Media Player to play the AVI file, which demonstrates how to complete the transfer.

int attendedRefer2(long sessionId, 
							     long replaceSessionId, 
							     String replaceMethod,
							     String target,
							     String referTo)

Make an attended refer.

Parameters

sessionId	The session ID of the call.
replaceSessionId	The session ID of the session to be replaced.
replaceMethod	The SIP method name to be added in the "Refer-To" header, usually INVITE or BYE.
target	The target to which the REFER message will be sent.
referTo	The URI to be added into the "Refer-To" header.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int outOfDialogRefer (long  replaceSessionId, String  replaceMethod, String  target, String  referTo)

Make an attended refer.

Parameters

replaceSessionId	The session ID of the session which will be replaced.
replaceMethod	The SIP method name which will be added in the "Refer-To" header, usually INVITE or BYE.
target	The target to which the REFER message will be sent.
referTo	The URI which will be added into the "Refer-To" header.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

long acceptRefer (long  referId, String  referSignaling)

By accepting the REFER request, a new call will be made if this function is called. The

function is usually called after onReceivedRefer callback event.

Parameters

referId	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, it will return a session ID greater than 0 to the new call for REFER; otherwise it will return a specific error code less than 0;

int rejectRefer (long  referId)

Reject the REFER request.

Parameters

referId

The ID of REFER request that comes from onReceivedRefer callback event.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Send audio and video stream functions

int enableSendPcmStreamToRemote (long  sessionId, boolean  state, int  streamSamplesPerSec)

Enable the SDK send PCM stream data to remote side from another source instead of microphone. This function MUST be called first to send the PCM stream data to another

side.

Parameters

sessionId	The session ID of call.
state	Set to true to enable the send stream, or false to disable.
streamSamplesPer Sec	The PCM stream data sample, in seconds. For example 8000 or 16000.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int sendPcmStreamToRemote (long  sessionId, byte[]  data, int  dataLength)

Send the audio stream in PCM format from another source instead of audio device

capturing (microphone).

Parameters

sessionId	Session ID of the call conversation.
data	The PCM audio stream data. It must be 16bit, mono.
dataLength	The size of data.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

Usually we should use it like below:

enableSendPcmStreamToRemote(sessionId, true, 16000); 
sendPcmStreamToRemote(sessionId, data, dataSize); 

You can't have too much audio data at one time as we have 100ms audio buffer only. Once you put too much, data will be lost. It is recommended to send 20ms audio data every 20ms.

int enableSendVideoStreamToRemote (long  sessionId, boolean  state)

Enable the SDK to send video stream data to remote side from another source instead of camera.

This function MUST be called first to send the video stream data to another side.

Parameters

sessionId	The session ID of call.
state	Set to true to enable the send stream, or false to disable.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int sendVideoStreamToRemote (long  sessionId, byte[]  data, int  dataLength, int  width, int  height)

Send the video stream in i420 from another source instead of video device capturing (camera).

Before calling this function, you MUST call the enableSendVideoStreamToRemote function.

Parameters

sessionId	Session ID of the call conversation.
data	The video stream data. It must be in i420 format.
dataLength	The size of data.
width	The width of the video image.
height	The height of video image.

Returns

If the function succeeds, it will return value is 0. If the function fails, it will return a specific error code.

RTP packets, Audio stream and video stream callback

long enableRtpCallback (long  sessionId, int  mediaType, int  directionMode)

Set the RTP callbacks to allow access to the sent and received RTP packets.

Parameters

sessionId	The session ID of the call.
mediaType	RTP packet media type, 0 for audio, 1 for video , 2 for screen.
directionMode	RTP packet direction, 0 for sending, 1 for receiving.

Returns

If the function succeeds, it will return value is 0. If the function fails, it will return a specific error code.

void enableAudioStreamCallback (long  sessionId, boolean  enable, int  enum_direction)

Enable/disable the audio stream callback. The onAudioRawCallback event will be triggered if the callback is enabled.

Parameters

sessionId	The session ID of call.
enable	Set to true to enable audio stream callback, or false to stop the callback.
enum_direction	The audio stream callback mode. Supported modes include ENUM_DIRECTION_NONE, ENUM_DIRECTION_SEND,
ENUM_DIRECTION_RECV ENUM_DIRECTION_SEND_RECV.

void enableVideoStreamCallback (long  sessionId, int  enum_direction)

Enable/disable the video stream callback, the onVideoRawCallback event will be triggered if the callback is enabled.

Parameters

sessionId	The session ID of call.
enum_direction	The video stream callback mode. Supported modes include ENUM_DIRECTION_NONE, ENUM_DIRECTION_SEND,
ENUM_DIRECTION_RECV, ENUM_DIRECTION_SEND_RECV.

Record functions

int startRecord(long sessionId, 
						    String recordFilePath,
						    String recordFileName, 
						    boolean appendTimeStamp,
						    int audioChannels,
						    int enum_fileFormat,
						    int enum_audioRecordMode,
						    int enum_videoRecordMode)

Start recording the call.

Parameters

sessionId	The session ID of call conversation.
recordFilePath	The file path to save record file. It must be existent.
recordFileName	The file name of record file. For example audiorecord.wav or videorecord.avi.
appendTimeStamp	Set to true to append the timestamp to the name of the recording file.
audioChannels	Set to record file audio channels, 1 - mono 2 - stereo.
enum_fileFormat	The record file format, allow below values: enum_FILE_FORMAT_WAVE = 1, ///< The record audio file is WAVE format. enum_FILE_FORMAT_AMR =2, ///< The record audio file is in AMR format with all voice data compressed by AMR codec. enum_FILE_FORMAT_MP3 = 3;///< The record audio file is in MP3 format. enum_FILE_FORMAT_MP4 = 4;///< The record video file is in MP4(AAC and H264) format.
enum_audioRecor dMode	The audio record mode, allow below values: enum_RECORD_MODE_NONE = 0, ///< Not Record. enum_RECORD_MODE_RECV = 1, ///< Only record the received data. enum_RECORD_MODE_SEND, ///< Only record send data. enum_RECORD_MODE_BOTH ///< Record both received and sent data.
enum_videoRecord Mode	Allow to set video record mode. Support to record received and/or sent video.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int stopRecord (long  sessionId)

Stop recording.

Parameters

sessionId

The session ID of call conversation.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Play audio and video file and RTMP/RTSP stream functions

int startPlayingFileToRemote (long  sessionId, String  fileUrl, boolean  loop, int  playAudio)

Play an local file or RTSP/RTMP stream to remote party.

Parameters

sessionId	Session ID of the call.
fileUrl	The url or filepath, such as "/mnt/sdcard/test.avi".
loop	Set to false to stop playing video file when it is ended, or true to play it repeatedly.
playAudio	0 - Not play file audio. 1 - Play file audio. 2 - Play file audio mix with Mic.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int stopPlayingFileToRemote (long  sessionId)

Stop play file to remote side.

Parameters

sessionId

Session ID of the call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int startPlayingFileLocally (String  fileUrl, boolean  loop, PortSIPVideoRenderer  renderer)

Play an local file or RTSP/RTMP stream.

Parameters

fileUrl	The url or filepath, such as "/mnt/sdcard/test.avi".
loop	Set to false to stop playing video file when it is ended, or true to play it repeatedly.
renderer	SurfaceView a SurfaceView for displaying the play image.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int stopPlayingFileLocally ()

Stop play file locally.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void audioPlayLoopbackTest (boolean  enable)

Used for testing loopback for the audio device.

Parameters

enable

Set to true to start testing audio loopback test; or set to false to stop.

Conference functions

int createAudioConference ()

Create an audio conference. It will fail if the existing conference is not ended yet. Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int createVideoConference(PortSIPVideoRenderer conferenceVideoWindow,
									 int videoWidth, 
									 int videoHeight, 
									 int layout)

Create a video conference. It will fail if the existing conference is not ended yet.

Parameters

conferenceVideoW indow	SurfaceView The window used for displaying the conference video.
videoWidth	Width of conference video resolution
videoHeight	Height of conference video resolution
layout	Conference Video layout, default is 0 - Adaptive. 0 - Adaptive(1,3,5,6) 1 - Only Local Video 2 - 2 video,PIP 3 - 2 video, Left and right 4 - 2 video, Up and Down 5 - 3 video 6 - 4 split video 7 - 5 video

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

void destroyConference ()

End the exist conference.

int setConferenceVideoWindow (PortSIPVideoRenderer  conferenceVideoWindow)

Set the window for a conference that is used for displaying the received remote video

image.

Parameters

conferenceVideoWindow

SurfaceView The window which is used for displaying the conference vide

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int joinToConference (long  sessionId)

Join a session into existing conference. If the call is in hold, it will be un-hold

automatically.

Parameters

sessionId

Session ID of the call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int removeFromConference (long  sessionId)

Remove a session from an existing conference.

Parameters

sessionId

Session ID of the call.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

RTP and RTCP QOS functions

int setAudioRtcpBandwidth (long  sessionId, int  BitsRR, int  BitsRS, int  KBitsAS)

Set the audio RTCP bandwidth parameters as RFC3556.

Parameters

sessionId	Set the audio RTCP bandwidth parameters as RFC3556.
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, it will return value 0. If the function fails, it will return a specific error code.

int setVideoRtcpBandwidth (long  sessionId, int  BitsRR, int  BitsRS, int  KBitsAS)

Set the video RTCP bandwidth 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, it will return value 0. If the function fails, it will return a specific error code.

int enableAudioQos (boolean  state)

Set the DSCP (differentiated services code point) value of QoS (Quality of Service) for audio channel.

Parameters

state

Set to YES to enable audio QoS and DSCP value will be 46; or NO to disable audio QoS and DSCP value will be 0.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int enableVideoQos (boolean  state)

Set the DSCP(differentiated services code point) value of QoS(Quality of Service) for video channel.

Parameters

state

Set to YES to enable video QoS and DSCP value will be 34; or NO to disable video QoS and DSCP value will be 0.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setVideoMTU (int  mtu)

Set the MTU size for video RTP packet.

Parameters

mtu	Set MTU value. Allow values range 512 - 65507. Default is 14000.

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

int getStatistics (long  sessionId)

Obtain the statistics of channel. the event onStatistics will be triggered.

Parameters

sessionId

The session ID of call conversation.

Returns

If the function succeeds, it will return value is 0. If the function fails, it will return a specific error code.

Audio effect functions

void enableVAD (boolean  state)

Enable/disable Voice Activity Detection(VAD).

Parameters

state

Set to true to enable VAD, or false to disable.

void enableAEC (boolean  state)

Enable/disable AEC (Acoustic Echo Cancellation).

Parameters

state

Set to true to enable AEC, or false to disable.

void enableCNG (boolean  state)

Enable/disable Comfort Noise Generator(CNG).

Parameters

state

Set to true to enable CNG, or false to disable.

void enableAGC (boolean  state)

Enable/disable Automatic Gain Control(AGC).

Parameters

state

Set to true to enable AEC, or false to disable.

void enableANS (boolean  state)

Enable/disable Audio Noise Suppression(ANS).

Parameters

state

Set to true to enable ANS, or false to disable.

Send OPTIONS/INFO/MESSAGE functions

int sendOptions (String  to, String  sdp)

Send OPTIONS message.

Parameters

to	The recipient of OPTIONS message.
sdp	The SDP of OPTIONS message. It's optional if user does not want to send the SDP with OPTIONS message.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return value a specific error code.

int sendInfo (long  sessionId, String  mimeType, String  subMimeType, String  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 will be sent with INFO message.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

long sendMessage(long sessionId, 
							   String mimeType,
							   String subMimeType, 
							   byte[] message, 
							   int messageLength)

Send a MESSAGE message to remote side in dialog.

Parameters

sessionId	The session ID of call.
mimeType	The mime type of MESSAGE message.
subMimeType	The sub mime type of MESSAGE message.
message	The contents that will be sent with MESSAGE message. Binary data allowed.
messageLength	The message size.

Returns

If the function succeeds, it will return a message ID that allows to track the message sending state in onSendMessageSuccess and onSendMessageFailure. If the function fails, it will return a specific error code that is 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 sending.

sendMessage(sessionId, "text", "plain", "hello",6);

Example 2: Send a binary message.

sendMessage(sessionId, "application", "vnd.3gpp.sms", binData, binDataSize);

long sendOutOfDialogMessage (String  to, String  mimeType, String  subMimeType, boolean  isSMS, byte[]  message, int  messageLength)

Send a out of dialog MESSAGE message to remote side.

Parameters

to	The message receiver. Likes sip:[e-mail]
mimeType	The mime type of MESSAGE message.
subMimeType	The sub mime type of MESSAGE message.
isSMS	Set to YES to specify "messagetype=SMS" in the To line, or NO to disable.
message	The contents that will be sent with MESSAGE message. Binary data allowed.
messageLength	The message size.

Returns

If the function succeeds, it will return a message ID that allows to track the message sending state in onSendOutOfMessageSuccess and onSendOutOfMessageFailure. If the function fails, it will return a specific error code that is 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 sending.

sendOutOfDialogMessage("sip:[e-mail]", "text", "plain", "hello", 6);

Example 2: Send a binary message.

sendOutOfDialogMessage("sip:[e-mail]","application", "vnd.3gpp.sms", binData, binDataSize);

long setPresenceMode (int  mode)

Indicate the SDK uses the P2P mode for presence or presence agent mode.

Parameters

mode	0 - P2P mode; 1 - Presence Agent mode. Default is P2P mode.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

Since presence agent mode requires the PBX/Server support the PUBLISH, please ensure you have your server and PortSIP PBX support this feature. For more details please visit: https://www.portsip.com/portsip-pbx

long setDefaultSubscriptionTime (int  secs)

Set the default expiration time to be used when creating a subscription.

Parameters

secs	The default expiration time of subscription.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

long setDefaultPublicationTime (int  secs)

Set the default expiration time to be used when creating a publication.

Parameters

secs	The default expiration time of publication.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

long presenceSubscribe (String  contact, String  subject)

Send a SUBSCRIBE message for presence to a contact.

Parameters

contact	The target contact, it must be in the format of sip:[e-mail].
subject	This subject text will be inserted into the SUBSCRIBE message. For example: "Hello, I'm Jason". The subject maybe is in UTF8 format. You should use UTF8 to decode it.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int presenceTerminateSubscribe (long   subscribeId)

Terminate the given presence subscription.

Parameters

subscribeId

The ID of the subscription.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int presenceAcceptSubscribe (long   subscribeId)

Accept the presence SUBSCRIBE request which received from contact.

Parameters

subscribeId

Subscription ID. When receiving a SUBSCRIBE request from contact, the event onPresenceRecvSubscribe will be triggered. The event includes the subscription ID.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int presenceRejectSubscribe (long   subscribeId)

Reject a presence SUBSCRIBE request received from contact.

Parameters

subscribeId

Subscription ID. When receiving a SUBSCRIBE request from contact, the event onPresenceRecvSubscribe will be triggered. The event inclues the subscription ID.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

int setPresenceStatus (long   subscribeId, String  statusText)

Send a NOTIFY message to contact to notify that presence status is online/offline/changed.

Parameters

subscribeId	Subscription ID. When receiving a SUBSCRIBE request from contact, the event onPresenceRecvSubscribe that includes the Subscription ID will be triggered.
statusText	The state text of presence status. For example: "I'm here", offline must use "offline"

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

long sendSubscription (String  to, String  eventName)

Send a SUBSCRIBE message to subscribe an event.

Parameters

to	The user/extension will be subscribed.
eventName	The event name to be subscribed.

Returns

If the function succeeds, it will return the ID of that SUBSCRIBE which is greater than 0. If the function fails, it will return a specific error code which is less than 0.

Remarks

Example 1, below code indicates that user/extension 101 is subscribed to MWI (Message Waiting notifications) for checking his voicemail: int32 mwiSubId = sendSubscription("sip:[e-mail]", "message-summary");

Example 2, to monitor a user/extension call status, You can use code: sendSubscription(mSipLib, "100", "dialog"); Extension 100 refers to the user/extension to be monitored. Once being monitored, when extension 100 hold a call or is ringing, the onDialogStateUpdated callback will be triggered.

int terminateSubscription (long   subscribeId)

Terminate the given subscription.

Parameters

subscribeId

The ID of the subscription.

Returns

If the function succeeds, it will return value 0. If the function fails, it will return a specific error code.

Remarks

For example, if you want stop check the MWI, use below code:

terminateSubscription(mwiSubId);

Class Documentation

com.portsip.PortSipEnumDefine

com.portsip.PortSipEnumDefine.AudioDevice Enum Reference

Public Attributes

• SPEAKER_PHONE

• WIRED_HEADSET

• EARPIECE

• BLUETOOTH

• NONE

Detailed Description

AudioDevice list possible audio devices that we currently support.

Static Public Attributes

• static final int enum_AUDIOCODEC_G729 = 18

• static final int enum_AUDIOCODEC_PCMA = 8

• static final int enum_AUDIOCODEC_PCMU = 0

• static final int enum_AUDIOCODEC_GSM = 3

• static final int enum_AUDIOCODEC_G722 = 9

• static final int enum_AUDIOCODEC_ILBC = 97

• static final int enum_AUDIOCODEC_AMR = 98

• static final int enum_AUDIOCODEC_AMRWB = 99

• static final int enum_AUDIOCODEC_SPEEX = 100

• static final int enum_AUDIOCODEC_SPEEXWB =102

• static final int enum_AUDIOCODEC_ISACWB = 103

• static final int enum_AUDIOCODEC_ISACSWB =104

• static final int enum_AUDIOCODEC_OPUS =105

• static final int enum_AUDIOCODEC_DTMF = 101

• static final int enum_VIDEOCODEC_NONE = -1

• static final int enum_VIDEOCODEC_I420 = 133

• static final int enum_VIDEOCODEC_H264 = 125

• static final int enum_VIDEOCODEC_VP8 = 120

• static final int enum_VIDEOCODEC_VP9 = 122

• static final int enum_SRTPPOLICY_NONE = 0

• static final int enum_SRTPPOLICY_FORCE = 1

• static final int enum_SRTPPOLICY_PREFER = 2

• static final int enum_TRANSPORT_UDP = 0

• static final int enum_TRANSPORT_TLS = 1

• static final int enum_TRANSPORT_TCP = 2

• static final int enum_LOG_LEVEL_NONE = -1

• static final int enum_LOG_LEVEL_ERROR = 1

• static final int enum_LOG_LEVEL_WARNING = 2

• static final int enum_LOG_LEVEL_INFO = 3

• static final int enum_LOG_LEVEL_DEBUG = 4

• static final int enum_DTMF_MOTHOD_RFC2833 = 0

• static final int enum_DTMF_MOTHOD_INFO = 1

• static final int enum_DIRECTION_NONE = 0

• static final int enum_DIRECTION_SEND_RECV = 1

• static final int enum_DIRECTION_SEND = 2

• static final int enum_DIRECTION_RECV = 3

• static final int enum_RECORD_MODE_NONE = 0

• static final int enum_RECORD_MODE_RECV = 1

• static final int enum_RECORD_MODE_SEND = 2

• static final int enum_RECORD_MODE_BOTH = 3

• static final int enum_FILE_FORMAT_WAVE = 1

  The record audio file is in WAVE format.

• static final int enum_FILE_FORMAT_AMR = 2

  The record audio file is in AMR format - all voice data are compressed by AMR codec. Mono.

• static final int enum_FILE_FORMAT_MP3 = 3 The record audio file is in MP3 format.

• static final int enum_FILE_FORMAT_MP4 = 4

  The record video file is in MP4(AAC and H264) format. ![ref6]

Member Data Documentation

final int com.portsip.PortSipEnumDefine.enum_VIDEOCODEC_NONE = -1[static]

Used in startRecord only

final int com.portsip.PortSipEnumDefine.enum_VIDEOCODEC_I420 = 133[static]

Used in startRecord only

final int com.portsip.PortSipEnumDefine.enum_DIRECTION_SEND_RECV = 1[static]

both received and sent

final int com.portsip.PortSipEnumDefine.enum_DIRECTION_SEND = 2[static]

Only the sent

final int com.portsip.PortSipEnumDefine.enum_DIRECTION_RECV = 3[static]

Only the received

final int com.portsip.PortSipEnumDefine.enum_RECORD_MODE_NONE = 0[static]

Not Recorded.

final int com.portsip.PortSipEnumDefine.enum_RECORD_MODE_RECV = 1[static]

Only record the received data.

final int com.portsip.PortSipEnumDefine.enum_RECORD_MODE_SEND = 2[static]

Only record the sent data.

final int com.portsip.PortSipEnumDefine.enum_RECORD_MODE_BOTH = 3[static]

Static Public Attributes

• static final int ECoreErrorNone = 0

• static final int INVALID_SESSION_ID = -1

• static final int ECoreAlreadyInitialized = -60000

• static final int ECoreNotInitialized = -60001

• static final int ECoreSDKObjectNull = -60002

• static final int ECoreArgumentNull = -60003

• static final int ECoreInitializeWinsockFailure = -60004

• static final int ECoreUserNameAuthNameEmpty = -60005

• static final int ECoreInitiazeStackFailure = -60006

• static final int ECorePortOutOfRange = -60007

• static final int ECoreAddTcpTransportFailure = -60008

• static final int ECoreAddTlsTransportFailure = -60009

• static final int ECoreAddUdpTransportFailure = -60010

• static final int ECoreNotSupportMediaType = -60011

• static final int ECoreNotSupportDTMFValue = -60012

• static final int ECoreAlreadyRegistered = -60021

• static final int ECoreSIPServerEmpty = -60022

• static final int ECoreExpiresValueTooSmall = -60023

• static final int ECoreCallIdNotFound = -60024

• static final int ECoreNotRegistered = -60025

• static final int ECoreCalleeEmpty = -60026

• static final int ECoreInvalidUri = -60027

• static final int ECoreAudioVideoCodecEmpty = -60028

• static final int ECoreNoFreeDialogSession = -60029

• static final int ECoreCreateAudioChannelFailed = -60030

• static final int ECoreSessionTimerValueTooSmall = -60040

• static final int ECoreAudioHandleNull = -60041

• static final int ECoreVideoHandleNull = -60042

• static final int ECoreCallIsClosed = -60043

• static final int ECoreCallAlreadyHold = -60044

• static final int ECoreCallNotEstablished = -60045

• static final int ECoreCallNotHold = -60050

• static final int ECoreSipMessaegEmpty = -60051

• static final int ECoreSipHeaderNotExist = -60052

• static final int ECoreSipHeaderValueEmpty = -60053

• static final int ECoreSipHeaderBadFormed = -60054

• static final int ECoreBufferTooSmall = -60055

• static final int ECoreSipHeaderValueListEmpty = -60056

• static final int ECoreSipHeaderParserEmpty = -60057

• static final int ECoreSipHeaderValueListNull = -60058

• static final int ECoreSipHeaderNameEmpty = -60059

• static final int ECoreAudioSampleNotmultiple = -60060

• static final int ECoreAudioSampleOutOfRange = -60061

• static final int ECoreInviteSessionNotFound = -60062

• static final int ECoreStackException = -60063

• static final int ECoreMimeTypeUnknown = -60064

• static final int ECoreDataSizeTooLarge = -60065

• static final int ECoreSessionNumsOutOfRange = -60066

• static final int ECoreNotSupportCallbackMode = -60067

• static final int ECoreNotFoundSubscribeId = -60068

• static final int ECoreCodecNotSupport = -60069

• static final int ECoreCodecParameterNotSupport = -60070

• static final int ECorePayloadOutofRange = -60071

• static final int ECorePayloadHasExist = -60072

• static final int ECoreFixPayloadCantChange = -60073

• static final int ECoreCodecTypeInvalid = -60074

• static final int ECoreCodecWasExist = -60075

• static final int ECorePayloadTypeInvalid = -60076

• static final int ECoreArgumentTooLong = -60077

• static final int ECoreMiniRtpPortMustIsEvenNum = -60078

• static final int ECoreCallInHold = -60079

• static final int ECoreNotIncomingCall = -60080

• static final int ECoreCreateMediaEngineFailure = -60081

• static final int ECoreAudioCodecEmptyButAudioEnabled = -60082

• static final int ECoreVideoCodecEmptyButVideoEnabled = -60083

• static final int ECoreNetworkInterfaceUnavailable = -60084

• static final int ECoreWrongDTMFTone = -60085

• static final int ECoreWrongLicenseKey = -60086

• static final int ECoreTrialVersionLicenseKey = -60087

• static final int ECoreOutgoingAudioMuted = -60088

• static final int ECoreOutgoingVideoMuted = -60089

• static final int ECoreFailedCreateSdp = -60090

• static final int ECoreTrialVersionExpired = -60091

• static final int ECoreStackFailure = -60092

• static final int ECoreTransportExists = -60093

• static final int ECoreUnsupportTransport = -60094

• static final int ECoreAllowOnlyOneUser = -60095

• static final int ECoreUserNotFound = -60096

• static final int ECoreTransportsIncorrect = -60097

• static final int ECoreCreateTransportFailure = -60098

• static final int ECoreTransportNotSet = -60099

• static final int ECoreECreateSignalingFailure = -60100

• static final int ECoreArgumentIncorrect = -60101

• static final int ECoreIVRObjectNull = -61001

• static final int ECoreIVRIndexOutOfRange = -61002

• static final int ECoreIVRReferFailure = -61003

• static final int ECoreIVRWaitingTimeOut = -61004

• static final int EAudioFileNameEmpty = -70000

• static final int EAudioChannelNotFound = -70001

• static final int EAudioStartRecordFailure = -70002

• static final int EAudioRegisterRecodingFailure = -70003

• static final int EAudioRegisterPlaybackFailure = -70004

• static final int EAudioGetStatisticsFailure = -70005

• static final int EAudioPlayFileAlreadyEnable = -70006

• static final int EAudioPlayObjectNotExist = -70007

• static final int EAudioPlaySteamNotEnabled = -70008

• static final int EAudioRegisterCallbackFailure = -70009

• static final int EAudioCreateAudioConferenceFailure = -70010

• static final int EAudioOpenPlayFileFailure = -70011

• static final int EAudioPlayFileModeNotSupport = -70012

• static final int EAudioPlayFileFormatNotSupport = -70013

• static final int EAudioPlaySteamAlreadyEnabled = -70014

• static final int EAudioCreateRecordFileFailure = -70015

• static final int EAudioCodecNotSupport = -70016

• static final int EAudioPlayFileNotEnabled = -70017

• static final int EAudioPlayFileUnknowSeekOrigin = -70018

• static final int EAudioCantSetDeviceIdDuringCall =-70019

• static final int EAudioVolumeOutOfRange =-70020

• static final int EVideoFileNameEmpty = -80000

• static final int EVideoGetDeviceNameFailure = -80001

• static final int EVideoGetDeviceIdFailure = -80002

• static final int EVideoStartCaptureFailure = -80003

• static final int EVideoChannelNotFound = -80004

• static final int EVideoStartSendFailure = -80005

• static final int EVideoGetStatisticsFailure = -80006

• static final int EVideoStartPlayAviFailure = -80007

• static final int EVideoSendAviFileFailure = -80008

• static final int EVideoRecordUnknowCodec = -80009

• static final int EVideoCantSetDeviceIdDuringCall = -80010

• static final int EVideoUnsupportCaptureRotate = -80011

• static final int VideoUnsupportCaptureResolution = -80012

• static final int ECameraSwitchTooOften = -80013

• static final int EMTUOutOfRange = -80014

• static final int EDeviceGetDeviceNameFailure = -90001