PortSIP VoIP SDK Manual for Android

Version 19.4

5/21/24

Table of Contents

Welcome to PortSIP VoIP SDK For Android .......................................................................................... 2

Getting Started .......................................................................................................................................... 2

Contents .................................................................................................................................................... 2

SDK User Manual ..................................................................................................................................... 2

Website ..................................................................................................................................................... 2

Support ...................................................................................................................................................... 2

Installation Prerequisites ........................................................................................................................... 2

Frequently Asked Questions ..................................................................................................................... 3

1. Where can I download the PortSIP VoIP SDK for test? .............................................................. 3

2. How can I compile the sample project? ........................................................................................ 3

3. How can I create a new project with PortSIP VoIP SDK? ........................................................... 3

4. How can I test the P2P call (without SIP server)? ........................................................................ 3

5. Is the SDK thread safe? ................................................................................................................. 4

Module Index ............................................................................................................................................ 5

Hierarchical Index ..................................................................................................................................... 6

Class Index ................................................................................................................................................ 7

Module Documentation ............................................................................................................................ 8

SDK Callback events ........................................................................................................................ 8

Register events .................................................................................................................................. 9

Call events ....................................................................................................................................... 10

Refer events .................................................................................................................................... 14

Signaling events .............................................................................................................................. 16

MWI events ..................................................................................................................................... 17

DTMF events .................................................................................................................................. 18

INFO/OPTIONS message events ................................................................................................... 19

Presence events ............................................................................................................................... 20

audio device changed,Play audio and video file finished events .................................................... 23

RTP callback events ........................................................................................................................ 25

SDK functions ................................................................................................................................. 27

Initialize and register functions ....................................................................................................... 28

Audio and video codecs functions .................................................................................................. 32

Additional settings functions .......................................................................................................... 35

Access SIP message header functions ............................................................................................ 41

Audio and video functions .............................................................................................................. 44

Call functions .................................................................................................................................. 48

Refer functions ................................................................................................................................ 52

Send audio and video stream functions .......................................................................................... 55

RTP packets, Audio stream and video stream callback .................................................................. 57

Record functions ............................................................................................................................. 58

Play audio and video file and RTMP/RTSP stream functions ........................................................ 60

Conference functions ...................................................................................................................... 62

RTP and RTCP QOS functions ...................................................................................................... 64

RTP statistics functions .................................................................................................................. 66

Audio effect functions ..................................................................................................................... 67

Send OPTIONS/INFO/MESSAGE functions ................................................................................. 68

Class Documentation .............................................................................................................................. 73

com.portsip.PortSipEnumDefine.AUDIOCODEC ......................................................................... 73

com.portsip.PortSipEnumDefine.AudioDevice .............................................................................. 74

com.portsip.OnPortSIPEvent .......................................................................................................... 75

com.portsip.PortSIPCameraCapturer .............................................................................................. 77

com.portsip.PortSipEnumDefine .................................................................................................... 78

com.portsip.PortSipErrorcode ......................................................................................................... 80

com.portsip.PortSipSdk .................................................................................................................. 83

com.portsip.PortSIPVideoRenderer ................................................................................................ 87

Index ....................................................................................................................................................... 88

Welcome to PortSIP VoIP SDK For Android

Create your SIP-based application for multiple platforms (iOS, Android, Windows, Mac OS

and Linux) with our SDK.

The rewarding PortSIP VoIP SDK is a powerful and versatile set of tools that dramatically

accelerate SIP application development. It includes a suite of stacks, SDKs, and some Sample

projects, with each of them enables developers to combine all the necessary components to

create an ideal development environment for every application's specific needs.

The PortSIP VoIP SDK complies with IETF and 3GPP standards, and is IMS-compliant

(3GPP/3GPP2, TISPAN and PacketCable 2.0). These high performance SDKs provide

unified API layers for full user control and flexibility.

Getting Started

You can download PortSIP VoIP SDK Sample projects at our Website. Samples include

demos for VC++, C#, VB.NET, Delphi XE, Xcode (for iOS and Mac OS), Android Studio

with the sample project source code provided (with SDK source code exclusive). The sample

projects demonstrate how to create a powerful SIP application with our SDK easily and

quickly.

Contents

The sample package for downloading contains almost all of materials for PortSIP VoIP SDK:

documentation, Dynamic/Static libraries, sources, headers, datasheet, and everything else a

SDK user might need!

SDK User Manual

To be started with, it is recommended to read the documentation of PortSIP VoIP SDK, SDK

User Manual page, which gives a brief description of each API function.

Website

Some general interest or often changing PortSIP SDK information will be posted on the

PortSIP website in real time. The release contains links to the site, so while browsing you

may see occasional broken links if you are not connected to the Internet. To be sure

everything needed for using the PortSIP VoIP SDK has been contained within the release.

Support

Please send email to our Support team if you need any help.

Installation Prerequisites

To use PortSIP VoIP/IMS SDK for Android for development, SDK version with later than

API-16 is required.

Frequently Asked Questions

1. Where can I download the PortSIP VoIP SDK for test?

All sample projects of the PortSIP VoIP SDK can be found and downloaded at:

https://www.portsip.com/download-portsip-voip-sdk/

https://www.PortSIP.com/portsip-voip-sdk

2. How can I compile the sample project?

1. Download the sample project from PortSIP website.

2. Extract the .zip file.

3. Open the project by your Android studio:

4. Compile the sample project directly.

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

1. Download the sample project and evaluation SDK and extract it to a specified directory

2. Run Android Studio and create a new Android Application Project

3. Copy all files form libs directory under extracted directory to the libs directory of your new

application.

4. Import the dependent class form the SDK. For example: import com.portsip.OnPortSIPEvent;

import com.portsip.PortSipSdk;

5. Inherit the interface OnPortSIPEvent to process the callback events.

6. Initialize SDK. For example: mPortSIPSDK = new PortSipSdk();

mPortSIPSDK.setOnPortSIPEvent(instanceofOnPortSIPEvent);

mPortSIPSDK.CreateCallManager(context); mPortSIPSDK.initialize(...); For more details please

refer to the Sample project source code.

4. How can I test the P2P call (without SIP server)?

1. Download and extract the SDK sample project ZIP file into local. Compile and run the

"P2PSample" project.

2. Run the P2Psample on two devices. For example, run it on device A and device B, and IP address

for A is 192.168.1.10, IP address for B is 192.168.1.11.

3. Enter a user name and password on A. For example, enter user name 111, and password aaa (you

can enter anything for the password as the SDK will ignore it). Enter a user name and password on

B. For example, enter user name 222, and password aaa.

4. Click the "Initialize" button on A and B. If the default port 5060 is already in use, the P2PSample

will prompt "Initialize failure". In case of this, please click the "Uninitialize" button and change

the local port, and click the "Initialize" button again.

5. The log box will appear "Initialized" if the SDK is successfully initialized.

6. To make call from A to B, enter "sip:[email protected].11" and click "Dial" button; while to make

call from B to A, enter "sip:[email protected]".

Note: If the local sip port is changed to other port, for example, A is using local port 5080,

and B is using local port 6021, to make call from A to B, please enter

"sip:[email protected]:6021" and dial; while to make call from B to A, enter

"sip:[email protected]:5080".

5. Is the SDK thread safe?

Yes, the SDK is thread safe. You can call any of the API functions without the need to

consider the multiple threads.

Note: the SDK allows to call API functions in callback events directly - except for the

"onAudioRawCallback", "onVideoRawCallback", "onRTPPacketCallback" callbacks.

Module Index

Modules

Here is a list of all modules:

SDK Callback events ................................................................................................................................ 8

Register events ................................................................................................................................... 9

Call events ....................................................................................................................................... 10

Refer events ..................................................................................................................................... 14

Signaling events ............................................................................................................................... 16

MWI events ..................................................................................................................................... 17

DTMF events ................................................................................................................................... 18

INFO/OPTIONS message events .................................................................................................... 19

Presence events ................................................................................................................................ 20

audio device changed,Play audio and video file finished events .................................................... 23

RTP callback events ........................................................................................................................ 25

SDK functions ........................................................................................................................................ 27

Initialize and register functions ....................................................................................................... 28

Audio and video codecs functions ................................................................................................... 32

Additional settings functions ........................................................................................................... 35

Access SIP message header functions ............................................................................................. 41

Audio and video functions ............................................................................................................... 44

Call functions ................................................................................................................................... 48

Refer functions ................................................................................................................................ 52

Send audio and video stream functions ........................................................................................... 55

RTP packets, Audio stream and video stream callback .................................................................. 57

Record functions .............................................................................................................................. 58

Play audio and video file and RTMP/RTSP stream functions ........................................................ 60

Conference functions ....................................................................................................................... 62

RTP and RTCP QOS functions ....................................................................................................... 64

RTP statistics functions ................................................................................................................... 66

Audio effect functions ..................................................................................................................... 67

Send OPTIONS/INFO/MESSAGE functions ................................................................................. 68

Hierarchical Index

Class Hierarchy

This inheritance list is sorted roughly, but not completely, alphabetically:

com.portsip.PortSipEnumDefine.AUDIOCODEC ................................................................................ 73

com.portsip.PortSipEnumDefine.AudioDevice ...................................................................................... 74

AppRTCAudioManager.AudioManagerEvents

com.portsip.PortSipSdk ................................................................................................................... 83

com.portsip.OnPortSIPEvent .................................................................................................................. 75

com.portsip.PortSipEnumDefine ............................................................................................................ 78

com.portsip.PortSipErrorcode ................................................................................................................ 80

CapturerObserver

com.portsip.PortSIPCameraCapturer .............................................................................................. 77

SurfaceViewRenderer

com.portsip.PortSIPVideoRenderer ................................................................................................ 87

Class Index

Class List

Here are the classes, structs, unions and interfaces with brief descriptions:

com.portsip.PortSipEnumDefine.AUDIOCODEC ........................................................................ 73

com.portsip.PortSipEnumDefine.AudioDevice .............................................................................. 74

com.portsip.OnPortSIPEvent .......................................................................................................... 75

com.portsip.PortSIPCameraCapturer ........................................................................................... 77

com.portsip.PortSipEnumDefine .................................................................................................... 78

com.portsip.PortSipErrorcode ........................................................................................................ 80

com.portsip.PortSipSdk ................................................................................................................... 83

com.portsip.PortSIPVideoRenderer ............................................................................................... 87

Module Documentation

SDK Callback events

Modules

l Register events

l Call events

l Refer events

l Signaling events

l MWI events

l DTMF events

l INFO/OPTIONS message events

l Presence events

l audio device changed,Play audio and video file finished events

l RTP callback events

Detailed Description

SDK Callback events

Functions

l void com.portsip.OnPortSIPEvent.onRegisterSuccess (String reason, int code, String sipMessage)

l void com.portsip.OnPortSIPEvent.onRegisterFailure (String reason, int code, String sipMessage)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.onRegisterFailure (String reason, int code, String

sipMessage)

If 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

Functions

l void com.portsip.OnPortSIPEvent.onInviteIncoming (long sessionId, String callerDisplayName,

String caller, String calleeDisplayName, String callee, String audioCodecs, String videoCodecs,

boolean existsAudio, boolean existsVideo, String sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteTrying (long sessionId)

l void com.portsip.OnPortSIPEvent.onInviteSessionProgress (long sessionId, String audioCodecs,

String videoCodecs, boolean existsEarlyMedia, boolean existsAudio, boolean existsVideo, String

sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteRinging (long sessionId, String statusText, int

statusCode, String sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteAnswered (long sessionId, String callerDisplayName,

String caller, String calleeDisplayName, String callee, String audioCodecs, String videoCodecs,

boolean existsAudio, boolean existsVideo, String sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteFailure (long sessionId, String callerDisplayName,

String caller, String calleeDisplayName, String callee, String reason, int code, String sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteUpdated (long sessionId, String audioCodecs, String

videoCodecs, String screenCodecs, boolean existsAudio, boolean existsVideo, boolean

existsScreen, String sipMessage)

l void com.portsip.OnPortSIPEvent.onInviteConnected (long sessionId)

l void com.portsip.OnPortSIPEvent.onInviteBeginingForward (String forwardTo)

l void com.portsip.OnPortSIPEvent.onInviteClosed (long sessionId, String sipMessage)

l void com.portsip.OnPortSIPEvent.onDialogStateUpdated (String BLFMonitoredUri, String

BLFDialogState, String BLFDialogId, String BLFDialogDirection)

l void com.portsip.OnPortSIPEvent.onRemoteHold (long sessionId)

l void com.portsip.OnPortSIPEvent.onRemoteUnHold (long sessionId, String audioCodecs, String

videoCodecs, boolean existsAudio, boolean existsVideo)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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

The display name of caller

caller

The caller.

calleeDisplayNam

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 com.portsip.OnPortSIPEvent.onInviteTrying (long sessionId)

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

Parameters

sessionId

The session ID of the call.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

The display name of caller

caller

The caller.

calleeDisplayNam

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 com.portsip.OnPortSIPEvent.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

The display name of caller

caller

The caller.

calleeDisplayNam

The display name of callee.

callee

The callee.

reason

The failure reason.

code

The failure code.

sipMessage

The SIP message received.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

- the direction of the call

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

Functions

l void com.portsip.OnPortSIPEvent.onReceivedRefer (long sessionId, long referId, String to, String

from, String referSipMessage)

l void com.portsip.OnPortSIPEvent.onReferAccepted (long sessionId)

l void com.portsip.OnPortSIPEvent.onReferRejected (long sessionId, String reason, int code)

l void com.portsip.OnPortSIPEvent.onTransferTrying (long sessionId)

l void com.portsip.OnPortSIPEvent.onTransferRinging (long sessionId)

l void com.portsip.OnPortSIPEvent.onACTVTransferSuccess (long sessionId)

l void com.portsip.OnPortSIPEvent.onACTVTransferFailure (long sessionId, String reason, int

code)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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

The refer target.

from

The sender of REFER message.

referSipMessage

The SIP message of "REFER". Pass it to "acceptRefer" function.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.onTransferTrying (long sessionId)

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

Parameters

sessionId

The session ID of the call.

void com.portsip.OnPortSIPEvent.onTransferRinging (long sessionId)

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

Parameters

sessionId

The session ID of the call.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

Functions

l void com.portsip.OnPortSIPEvent.onReceivedSignaling (long sessionId, String message)

l void com.portsip.OnPortSIPEvent.onSendingSignaling (long sessionId, String message)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

Functions

l void com.portsip.OnPortSIPEvent.onWaitingVoiceMessage (String messageAccount, int

urgentNewMessageCount, int urgentOldMessageCount, int newMessageCount, int

oldMessageCount)

l void com.portsip.OnPortSIPEvent.onWaitingFaxMessage (String messageAccount, int

urgentNewMessageCount, int urgentOldMessageCount, int newMessageCount, int

oldMessageCount)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

Functions

l void com.portsip.OnPortSIPEvent.onRecvDtmfTone (long sessionId, int tone)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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

The DTMF tone 0.

The DTMF tone 1.

The DTMF tone 2.

The DTMF tone 3.

The DTMF tone 4.

The DTMF tone 5.

The DTMF tone 6.

The DTMF tone 7.

The DTMF tone 8.

The DTMF tone 9.

The DTMF tone *.

The DTMF tone #.

The DTMF tone A.

The DTMF tone B.

The DTMF tone C.

The DTMF tone D.

The DTMF tone FLASH.

INFO/OPTIONS message events

Functions

l void com.portsip.OnPortSIPEvent.onRecvOptions (String optionsMessage)

l void com.portsip.OnPortSIPEvent.onRecvInfo (String infoMessage)

l void com.portsip.OnPortSIPEvent.onRecvNotifyOfSubscription (long subscribeId, String

notifyMessage, byte[] messageData, int messageDataLength)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.onRecvOptions (String optionsMessage)

This event will be triggered when receiving the OPTIONS message.

Parameters

optionsMessage

The received whole OPTIONS message in text format.

void com.portsip.OnPortSIPEvent.onRecvInfo (String infoMessage)

This event will be triggered when receiving the INFO message.

Parameters

infoMessage

The whole INFO message received in text format.

void com.portsip.OnPortSIPEvent.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

The length of "messageData".

Presence events

Functions

l void com.portsip.OnPortSIPEvent.onPresenceRecvSubscribe (long subscribeId, String

fromDisplayName, String from, String subject)

l void com.portsip.OnPortSIPEvent.onPresenceOnline (String fromDisplayName, String from,

String stateText)

l void com.portsip.OnPortSIPEvent.onPresenceOffline (String fromDisplayName, String from)

l void com.portsip.OnPortSIPEvent.onRecvMessage (long sessionId, String mimeType, String

subMimeType, byte[] messageData, int messageDataLength)

l void com.portsip.OnPortSIPEvent.onRecvOutOfDialogMessage (String fromDisplayName, String

from, String toDisplayName, String to, String mimeType, String subMimeType, byte[]

messageData, int messageDataLength, String sipMessage)

l void com.portsip.OnPortSIPEvent.onSendMessageSuccess (long sessionId, long messageId, String

sipMessage)

l void com.portsip.OnPortSIPEvent.onSendMessageFailure (long sessionId, long messageId, String

reason, int code, String sipMessage)

l void com.portsip.OnPortSIPEvent.onSendOutOfDialogMessageSuccess (long messageId, String

fromDisplayName, String from, String toDisplayName, String to, String sipMessage)

l void com.portsip.OnPortSIPEvent.onSendOutOfDialogMessageFailure (long messageId, String

fromDisplayName, String from, String toDisplayName, String to, String reason, int code, String

sipMessage)

l void com.portsip.OnPortSIPEvent.onSubscriptionFailure (long subscribeId, int statusCode)

l void com.portsip.OnPortSIPEvent.onSubscriptionTerminated (long subscribeId)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

The length of "messageData".

void com.portsip.OnPortSIPEvent.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.

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

The length of "messageData".

sipMessage

The SIP message received.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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.

The message receiver.

sipMessage

The SIP message received.

void com.portsip.OnPortSIPEvent.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.

The message receiver.

reason

The failure reason.

code

The failure code.

sipMessage

The SIP message received.

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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

Functions

l void com.portsip.OnPortSIPEvent.onPlayFileFinished (long sessionId, String fileName)

l void com.portsip.OnPortSIPEvent.onStatistics (long sessionId, String statistics)

l void com.portsip.OnPortSIPEvent.onAudioDeviceChanged (PortSipEnumDefine.AudioDevice

audioDevice, Set< PortSipEnumDefine.AudioDevice > devices)

l void com.portsip.OnPortSIPEvent.onAudioFocusChange (int focusChange)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.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 com.portsip.OnPortSIPEvent.onAudioFocusChange (int focusC h a n ge)

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

Functions

l void com.portsip.OnPortSIPEvent.onRTPPacketCallback (long sessionId, int mediaType, int

enum_direction, byte[] RTPPacket, int packetSize)

l void com.portsip.OnPortSIPEvent.onAudioRawCallback (long sessionId, int enum_direction,

byte[] data, int dataLength, int samplingFreqHz)

l void com.portsip.OnPortSIPEvent.onVideoRawCallback (long sessionId, int enum_direction, int

width, int height, byte[] data, int dataLength)

Detailed Description

Function Documentation

void com.portsip.OnPortSIPEvent.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.

void com.portsip.OnPortSIPEvent.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

PortSipEnumDefine

Remarks

Example:

setAudioCodecParameter(AUDIOCODEC_AMR, "mode-set=0; octet-align=1;

robust-sorting=0")

int com.portsip.PortSipSdk.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

Functions

l String com.portsip.PortSipSdk.getVersion ()

l int com.portsip.PortSipSdk.enableRport (boolean enable)

l int com.portsip.PortSipSdk.enableEarlyMedia (boolean enable)

Enable/disable rport(RFC3581).

l int com.portsip.PortSipSdk.enablePriorityIPv6Domain (boolean enable)

l int com.portsip.PortSipSdk.setUriUserEncoding (String character, boolean enable)

l int com.portsip.PortSipSdk.setReliableProvisional (int mode)

l int com.portsip.PortSipSdk.enable3GppTags (boolean enable)

l void com.portsip.PortSipSdk.enableCallbackSignaling (boolean enableSending, boolean

enableReceived)

l void com.portsip.PortSipSdk.setSrtpPolicy (int enum_srtppolicy)

l int com.portsip.PortSipSdk.setRtpPortRange (int minimumRtpPort, int maximumRtpPort)

l int com.portsip.PortSipSdk.enableCallForward (boolean forBusyOnly, String forwardTo)

l int com.portsip.PortSipSdk.disableCallForward ()

l int com.portsip.PortSipSdk.enableSessionTimer (int timerSeconds)

l void com.portsip.PortSipSdk.disableSessionTimer ()

l void com.portsip.PortSipSdk.setDoNotDisturb (boolean state)

l void com.portsip.PortSipSdk.enableAutoCheckMwi (boolean state)

l int com.portsip.PortSipSdk.setRtpKeepAlive (boolean state, int keepAlivePayloadType, int

deltaTransmitTimeMS)

l int com.portsip.PortSipSdk.setKeepAliveTime (int keepAliveTime)

l int com.portsip.PortSipSdk.setAudioSamples (int ptime, int maxptime)

l int com.portsip.PortSipSdk.addSupportedMimeType (String methodName, String mimeType,

String subMimeType)

Detailed Description

Function Documentation

String com.portsip.PortSipSdk.getVersion ()

Get the version number of the current SDK.

Returns

String with version description

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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:[email protected].

Returns

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

error code.

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.disableSessionTimer ()

Disable the session timer.

void com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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. <br> "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

Functions

l String com.portsip.PortSipSdk.getSipMessageHeaderValue (String sipMessage, String

headerName)

l long com.portsip.PortSipSdk.addSipMessageHeader (long sessionId, String methodName, int

msgType, String headerName, String headerValue)

l int com.portsip.PortSipSdk.removeAddedSipMessageHeader (long addedSipMessageId)

l void com.portsip.PortSipSdk.clearAddedSipMessageHeaders ()

l long com.portsip.PortSipSdk.modifySipMessageHeader (long sessionId, String methodName, int

msgType, String headerName, String headerValue)

l int com.portsip.PortSipSdk.removeModifiedSipMessageHeader (long modifiedSipMessageId)

l void com.portsip.PortSipSdk.clearModifiedSipMessageHeaders ()

Detailed Description

Function Documentation

String com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.removeAddedSipMessageHeader (long

addedSipMessageId)

Remove the headers (custom header) added by addSipMessageHeader.

Parameters

addedSipMessageI

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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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();

Audio and video functions

Functions

l int com.portsip.PortSipSdk.setVideoDeviceId (int deviceId)

l int com.portsip.PortSipSdk.setVideoOrientation (int rotation)

l int com.portsip.PortSipSdk.enableVideoHardwareCodec (boolean enableHWEncoder, boolean

enableHWDecoder)

l int com.portsip.PortSipSdk.setVideoResolution (int width, int height)

l int com.portsip.PortSipSdk.setAudioBitrate (long sessionId, int enum_audiocodec, int bitrateKbps)

l int com.portsip.PortSipSdk.setVideoBitrate (long sessionId, int bitrateKbps)

l int com.portsip.PortSipSdk.setVideoFrameRate (long sessionId, int frameRate)

l int com.portsip.PortSipSdk.sendVideo (long sessionId, boolean send)

l int com.portsip.PortSipSdk.setRemoteVideoWindow (long sessionId, PortSIPVideoRenderer

renderer)

l int com.portsip.PortSipSdk.setRemoteScreenWindow (long sessionId, PortSIPVideoRenderer

renderer)

l void com.portsip.PortSipSdk.displayLocalVideo (boolean state, boolean mirror,

PortSIPVideoRenderer renderer)

l int com.portsip.PortSipSdk.setVideoNackStatus (boolean state)

l int com.portsip.PortSipSdk.setChannelOutputVolumeScaling (long sessionId, int scaling)

l int com.portsip.PortSipSdk.setChannelInputVolumeScaling (long sessionId, int scaling)

l void com.portsip.PortSipSdk.enableAudioManager (boolean state)

l Set< PortSipEnumDefine.AudioDevice > com.portsip.PortSipSdk.getAudioDevices ()

l int com.portsip.PortSipSdk.setAudioDevice (PortSipEnumDefine.AudioDevice defaultDevice)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.enableVideoHardwareCodec (boolean enableHWEncoder,

boolean enableHWDecoder)

Set enable/disable video Hardware codec.

Parameters

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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.setChannelOutputVolumeScaling (long sessionId, int

scaling)

Set a volume |scaling| to be applied to the outgoing 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.

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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> com.portsip.PortSipSdk.getAudioDevices ()

Get current set of available/selectable audio devices.

Returns

Current set of available/selectable audio devices.

int com.portsip.PortSipSdk.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

Functions

l long com.portsip.PortSipSdk.call (String callee, boolean sendSdp, boolean videoCall)

l int com.portsip.PortSipSdk.rejectCall (long sessionId, int code)

l int com.portsip.PortSipSdk.hangUp (long sessionId)

l int com.portsip.PortSipSdk.answerCall (long sessionId, boolean videoCall)

l int com.portsip.PortSipSdk.updateCall (long sessionId, boolean enableAudio, boolean

enableVideo)

l int com.portsip.PortSipSdk.hold (long sessionId)

l int com.portsip.PortSipSdk.unHold (long sessionId)

l int com.portsip.PortSipSdk.muteSession (long sessionId, boolean muteIncomingAudio, boolean

muteOutgoingAudio, boolean muteIncomingVideo, boolean muteOutgoingVideo)

l int com.portsip.PortSipSdk.forwardCall (long sessionId, String forwardTo)

l long com.portsip.PortSipSdk.pickupBLFCall (String replaceDialogId, boolean videoCall)

l int com.portsip.PortSipSdk.sendDtmf (long sessionId, int enum_dtmfMethod, int code, int

dtmfDuration, boolean playDtmfTone)

Detailed Description

Function Documentation

long com.portsip.PortSipSdk.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:[email protected] or sip:user002@sip.yourdomain.com: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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Set it to true to mute incoming audio stream. Once set, remote side audio

cannot be heard.

muteOutgoingAudi

Set it to true to mute outgoing audio stream. Once set, the remote side cannot

hear the audio.

muteIncomingVide

Set it to true to mute incoming video stream. Once set, remote side video

cannot be seen.

muteOutgoingVide

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 com.portsip.PortSipSdk.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:[email protected]" 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 com.portsip.PortSipSdk.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

Returns

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 com.portsip.PortSipSdk.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

The DTMF tone 0.

The DTMF tone 1.

The DTMF tone 2.

The DTMF tone 3.

The DTMF tone 4.

The DTMF tone 5.

The DTMF tone 6.

The DTMF tone 7.

The DTMF tone 8.

The DTMF tone 9.

The DTMF tone *.

The DTMF tone #.

The DTMF tone A.

The DTMF tone B.

The DTMF tone C.

The DTMF tone D.

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

Functions

l int com.portsip.PortSipSdk.refer (long sessionId, String referTo)

l int com.portsip.PortSipSdk.attendedRefer (long sessionId, long replaceSessionId, String referTo)

l int com.portsip.PortSipSdk.attendedRefer2 (long sessionId, long replaceSessionId, String

replaceMethod, String target, String referTo)

l int com.portsip.PortSipSdk.outOfDialogRefer (long replaceSessionId, String replaceMethod,

String target, String referTo)

l long com.portsip.PortSipSdk.acceptRefer (long referId, String referSignaling)

l int com.portsip.PortSipSdk.rejectRefer (long referId)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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:number@sipserver.com" 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:[email protected]");

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 com.portsip.PortSipSdk.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:number@sipserver.com" 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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.enableSendPcmStreamToRemote (long sessionId, boolean state, int

streamSamplesPerSec)

l int com.portsip.PortSipSdk.sendPcmStreamToRemote (long sessionId, byte[] data, int dataLength)

l int com.portsip.PortSipSdk.enableSendVideoStreamToRemote (long sessionId, boolean state)

l int com.portsip.PortSipSdk.sendVideoStreamToRemote (long sessionId, byte[] data, int

dataLength, int width, int height)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l long com.portsip.PortSipSdk.enableRtpCallback (long sessionId, int mediaType, int

directionMode)

l void com.portsip.PortSipSdk.enableAudioStreamCallback (long sessionId, boolean enable, int

enum_direction)

l void com.portsip.PortSipSdk.enableVideoStreamCallback (long sessionId, int enum_direction)

Detailed Description

functions

Function Documentation

long com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.startRecord (long sessionId, String recordFilePath, String

recordFileName, boolean appendTimeStamp, int audioChannels, int enum_fileFormat, int

enum_audioRecordMode, int enum_videoRecordMode)

l int com.portsip.PortSipSdk.stopRecord (long sessionId)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.startPlayingFileToRemote (long sessionId, String fileUrl, boolean loop,

int playAudio)

l int com.portsip.PortSipSdk.stopPlayingFileToRemote (long sessionId)

l int com.portsip.PortSipSdk.startPlayingFileLocally (String fileUrl, boolean loop,

PortSIPVideoRenderer renderer)

l int com.portsip.PortSipSdk.stopPlayingFileLocally ()

l void com.portsip.PortSipSdk.audioPlayLoopbackTest (boolean enable)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.createAudioConference ()

l int com.portsip.PortSipSdk.createVideoConference (PortSIPVideoRenderer

conferenceVideoWindow, int videoWidth, int videoHeight, int layout)

l void com.portsip.PortSipSdk.destroyConference ()

l int com.portsip.PortSipSdk.setConferenceVideoWindow (PortSIPVideoRenderer

conferenceVideoWindow)

l int com.portsip.PortSipSdk.joinToConference (long sessionId)

l int com.portsip.PortSipSdk.removeFromConference (long sessionId)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.destroyConference ()

End the exist conference.

int com.portsip.PortSipSdk.setConferenceVideoWindow (PortSIPVideoRenderer

conferenceVideoWindow)

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

image.

Parameters

conferenceVideoW

SurfaceView The window which is used for displaying the conference video

indow

Returns

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

error code.

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.setAudioRtcpBandwidth (long sessionId, int BitsRR, int BitsRS, int

KBitsAS)

l int com.portsip.PortSipSdk.setVideoRtcpBandwidth (long sessionId, int BitsRR, int BitsRS, int

KBitsAS)

l int com.portsip.PortSipSdk.enableAudioQos (boolean state)

l int com.portsip.PortSipSdk.enableVideoQos (boolean state)

l int com.portsip.PortSipSdk.setVideoMTU (int mtu)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.getStatistics (long sessionId)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.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

Functions

l void com.portsip.PortSipSdk.enableVAD (boolean state)

l void com.portsip.PortSipSdk.enableAEC (boolean state)

l void com.portsip.PortSipSdk.enableCNG (boolean state)

l void com.portsip.PortSipSdk.enableAGC (boolean state)

l void com.portsip.PortSipSdk.enableANS (boolean state)

Detailed Description

Function Documentation

void com.portsip.PortSipSdk.enableVAD (boolean state)

Enable/disable Voice Activity Detection(VAD).

Parameters

state

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

void com.portsip.PortSipSdk.enableAEC (boolean state)

Enable/disable AEC (Acoustic Echo Cancellation).

Parameters

state

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

void com.portsip.PortSipSdk.enableCNG (boolean state)

Enable/disable Comfort Noise Generator(CNG).

Parameters

state

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

void com.portsip.PortSipSdk.enableAGC (boolean state)

Enable/disable Automatic Gain Control(AGC).

Parameters

state

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

void com.portsip.PortSipSdk.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

Functions

l int com.portsip.PortSipSdk.sendOptions (String to, String sdp)

l int com.portsip.PortSipSdk.sendInfo (long sessionId, String mimeType, String subMimeType,

String infoContents)

l long com.portsip.PortSipSdk.sendMessage (long sessionId, String mimeType, String

subMimeType, byte[] message, int messageLength)

l long com.portsip.PortSipSdk.sendOutOfDialogMessage (String to, String mimeType, String

subMimeType, boolean isSMS, byte[] message, int messageLength)

l long com.portsip.PortSipSdk.setPresenceMode (int mode)

l long com.portsip.PortSipSdk.setDefaultSubscriptionTime (int secs)

l long com.portsip.PortSipSdk.setDefaultPublicationTime (int secs)

l long com.portsip.PortSipSdk.presenceSubscribe (String contact, String subject)

l int com.portsip.PortSipSdk.presenceTerminateSubscribe (long subscribeId)

l int com.portsip.PortSipSdk.presenceAcceptSubscribe (long subscribeId)

l int com.portsip.PortSipSdk.presenceRejectSubscribe (long subscribeId)

l int com.portsip.PortSipSdk.setPresenceStatus (long subscribeId, String statusText)

l long com.portsip.PortSipSdk.sendSubscription (String to, String eventName)

Send a SUBSCRIBE message to subscribe an event.

l int com.portsip.PortSipSdk.terminateSubscription (long subscribeId)

Detailed Description

Function Documentation

int com.portsip.PortSipSdk.sendOptions (String to, String sdp)

Send OPTIONS message.

Parameters

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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.sendOutOfDialogMessage (String to, String mimeType,

String subMimeType, boolean isSMS, byte[] message, int messageLength)

Send a out of dialog MESSAGE message to remote side.

Parameters

The message receiver. Likes sip:[email protected]

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:[email protected]", "text", "plain", "hello", 6);

Example 2: Send a binary message.

sendOutOfDialogMessage("sip:[email protected]","application",

"vnd.3gpp.sms", binData, binDataSize);

long com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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:[email protected].

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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.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 com.portsip.PortSipSdk.sendSubscription (String to, String eventName)

Send a SUBSCRIBE message to subscribe an event.

Parameters

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:[email protected]", "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 com.portsip.PortSipSdk.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.AUDIOCODEC Interface

Reference

The documentation for this interface was generated from the following file:

l PortSipEnumDefine.java

com.portsip.PortSipEnumDefine.AudioDevice Enum

Reference

Public Attributes

l SPEAKER_PHONE

l WIRED_HEADSET

l EARPIECE

l BLUETOOTH

l NONE

Detailed Description

AudioDevice list possible audio devices that we currently support.

The documentation for this enum was generated from the following file:

l PortSipEnumDefine.java

com.portsip.OnPortSIPEvent Interface Reference

Public Member Functions

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

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

l void onInviteIncoming (long sessionId, String callerDisplayName, String caller, String

calleeDisplayName, String callee, String audioCodecs, String videoCodecs, boolean existsAudio,

boolean existsVideo, String sipMessage)

l void onInviteTrying (long sessionId)

l void onInviteSessionProgress (long sessionId, String audioCodecs, String videoCodecs, boolean

existsEarlyMedia, boolean existsAudio, boolean existsVideo, String sipMessage)

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

l void onInviteAnswered (long sessionId, String callerDisplayName, String caller, String

calleeDisplayName, String callee, String audioCodecs, String videoCodecs, boolean existsAudio,

boolean existsVideo, String sipMessage)

l void onInviteFailure (long sessionId, String callerDisplayName, String caller, String

calleeDisplayName, String callee, String reason, int code, String sipMessage)

l void onInviteUpdated (long sessionId, String audioCodecs, String videoCodecs, String

screenCodecs, boolean existsAudio, boolean existsVideo, boolean existsScreen, String

sipMessage)

l void onInviteConnected (long sessionId)

l void onInviteBeginingForward (String forwardTo)

l void onInviteClosed (long sessionId, String sipMessage)

l void onDialogStateUpdated (String BLFMonitoredUri, String BLFDialogState, String

BLFDialogId, String BLFDialogDirection)

l void onRemoteHold (long sessionId)

l void onRemoteUnHold (long sessionId, String audioCodecs, String videoCodecs, boolean

existsAudio, boolean existsVideo)

l void onReceivedRefer (long sessionId, long referId, String to, String from, String

referSipMessage)

l void onReferAccepted (long sessionId)

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

l void onTransferTrying (long sessionId)

l void onTransferRinging (long sessionId)

l void onACTVTransferSuccess (long sessionId)

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

l void onReceivedSignaling (long sessionId, String message)

l void onSendingSignaling (long sessionId, String message)

l void onWaitingVoiceMessage (String messageAccount, int urgentNewMessageCount, int

urgentOldMessageCount, int newMessageCount, int oldMessageCount)

l void onWaitingFaxMessage (String messageAccount, int urgentNewMessageCount, int

urgentOldMessageCount, int newMessageCount, int oldMessageCount)

l void onRecvDtmfTone (long sessionId, int tone)

l void onRecvOptions (String optionsMessage)

l void onRecvInfo (String infoMessage)

l void onRecvNotifyOfSubscription (long subscribeId, String notifyMessage, byte[] messageData,

int messageDataLength)

l void onPresenceRecvSubscribe (long subscribeId, String fromDisplayName, String from, String

subject)

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

l void onPresenceOffline (String fromDisplayName, String from)

l void onRecvMessage (long sessionId, String mimeType, String subMimeType, byte[]

messageData, int messageDataLength)

l void onRecvOutOfDialogMessage (String fromDisplayName, String from, String toDisplayName,

String to, String mimeType, String subMimeType, byte[] messageData, int messageDataLength,

String sipMessage)

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

l void onSendMessageFailure (long sessionId, long messageId, String reason, int code, String

sipMessage)

l void onSendOutOfDialogMessageSuccess (long messageId, String fromDisplayName, String from,

String toDisplayName, String to, String sipMessage)

l void onSendOutOfDialogMessageFailure (long messageId, String fromDisplayName, String from,

String toDisplayName, String to, String reason, int code, String sipMessage)

l void onSubscriptionFailure (long subscribeId, int statusCode)

l void onSubscriptionTerminated (long subscribeId)

l void onPlayFileFinished (long sessionId, String fileName)

l void onStatistics (long sessionId, String statistics)

l void onAudioDeviceChanged (PortSipEnumDefine.AudioDevice audioDevice, Set<

PortSipEnumDefine.AudioDevice > devices)

l void onAudioFocusChange (int focusChange)

l void onRTPPacketCallback (long sessionId, int mediaType, int enum_direction, byte[] RTPPacket,

int packetSize)

l void onAudioRawCallback (long sessionId, int enum_direction, byte[] data, int dataLength, int

samplingFreqHz)

l void onVideoRawCallback (long sessionId, int enum_direction, int width, int height, byte[] data,

int dataLength)

The documentation for this interface was generated from the following file:

l OnPortSIPEvent.java

com.portsip.PortSIPCameraCapturer Class Reference

Inherits CapturerObserver.

Public Member Functions

l void InitializeVideoCapturer (Context applicationContext)

l void UnInitializeVideoCapturer ()

l void switchCamera (int newDeviceId)

l void onCapturerStarted (boolean success)

l void onCapturerStopped ()

l void onFrameCaptured (VideoFrame frame)

Public Attributes

l CameraVideoCapturer capturer

Member Function Documentation

void com.portsip.PortSIPCameraCapturer.onCapturerStarted (boolean success)

Notify if the capturer have been started successfully or not.

void com.portsip.PortSIPCameraCapturer.onCapturerStopped ()

Notify that the capturer has been stopped.

void com.portsip.PortSIPCameraCapturer.onFrameCaptured (VideoFrame frame)

Delivers a captured frame.

The documentation for this class was generated from the following file:

l PortSIPCameraCapturer.java

com.portsip.PortSipEnumDefine Class Reference

Classes

l interface AUDIOCODEC

l enum AudioDevice

Static Public Attributes

l static final int ENUM_AUDIOCODEC_G729 = 18

l static final int ENUM_AUDIOCODEC_PCMA = 8

l static final int ENUM_AUDIOCODEC_PCMU = 0

l static final int ENUM_AUDIOCODEC_GSM = 3

l static final int ENUM_AUDIOCODEC_G722 = 9

l static final int ENUM_AUDIOCODEC_ILBC = 97

l static final int ENUM_AUDIOCODEC_AMR = 98

l static final int ENUM_AUDIOCODEC_AMRWB = 99

l static final int ENUM_AUDIOCODEC_SPEEX = 100

l static final int ENUM_AUDIOCODEC_SPEEXWB =102

l static final int ENUM_AUDIOCODEC_ISACWB = 103

l static final int ENUM_AUDIOCODEC_ISACSWB =104

l static final int ENUM_AUDIOCODEC_OPUS =105

l static final int ENUM_AUDIOCODEC_DTMF = 101

l static final int ENUM_VIDEOCODEC_NONE = -1

l static final int ENUM_VIDEOCODEC_I420 = 133

l static final int ENUM_VIDEOCODEC_H264 = 125

l static final int ENUM_VIDEOCODEC_VP8 = 120

l static final int ENUM_VIDEOCODEC_VP9 = 122

l static final int ENUM_SRTPPOLICY_NONE = 0

l static final int ENUM_SRTPPOLICY_FORCE = 1

l static final int ENUM_SRTPPOLICY_PREFER = 2

l static final int ENUM_TRANSPORT_UDP = 0

l static final int ENUM_TRANSPORT_TLS = 1

l static final int ENUM_TRANSPORT_TCP = 2

l static final int ENUM_LOG_LEVEL_NONE = -1

l static final int ENUM_LOG_LEVEL_ERROR = 1

l static final int ENUM_LOG_LEVEL_WARNING = 2

l static final int ENUM_LOG_LEVEL_INFO = 3

l static final int ENUM_LOG_LEVEL_DEBUG = 4

l static final int ENUM_DTMF_MOTHOD_RFC2833 = 0

l static final int ENUM_DTMF_MOTHOD_INFO = 1

l static final int ENUM_DIRECTION_NONE = 0

l static final int ENUM_DIRECTION_SEND_RECV = 1

l static final int ENUM_DIRECTION_SEND = 2

l static final int ENUM_DIRECTION_RECV = 3

l static final int ENUM_RECORD_MODE_NONE = 0

l static final int ENUM_RECORD_MODE_RECV = 1

l static final int ENUM_RECORD_MODE_SEND = 2

l static final int ENUM_RECORD_MODE_BOTH = 3

l static final int ENUM_FILE_FORMAT_WAVE = 1

The record audio file is in WAVE format.

l 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.

l static final int ENUM_FILE_FORMAT_MP3 = 3

The record audio file is in MP3 format.

l static final int ENUM_FILE_FORMAT_MP4 = 4

The record video file is in MP4(AAC and H264) format.

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]

Record both received and sent data.

The documentation for this class was generated from the following file:

l PortSipEnumDefine.java

com.portsip.PortSipErrorcode Class Reference

Static Public Attributes

l static final int ECoreErrorNone = 0

l static final int INVALID_SESSION_ID = -1

l static final int ECoreAlreadyInitialized = -60000

l static final int ECoreNotInitialized = -60001

l static final int ECoreSDKObjectNull = -60002

l static final int ECoreArgumentNull = -60003

l static final int ECoreInitializeWinsockFailure = -60004

l static final int ECoreUserNameAuthNameEmpty = -60005

l static final int ECoreInitiazeStackFailure = -60006

l static final int ECorePortOutOfRange = -60007

l static final int ECoreAddTcpTransportFailure = -60008

l static final int ECoreAddTlsTransportFailure = -60009

l static final int ECoreAddUdpTransportFailure = -60010

l static final int ECoreNotSupportMediaType = -60011

l static final int ECoreNotSupportDTMFValue = -60012

l static final int ECoreAlreadyRegistered = -60021

l static final int ECoreSIPServerEmpty = -60022

l static final int ECoreExpiresValueTooSmall = -60023

l static final int ECoreCallIdNotFound = -60024

l static final int ECoreNotRegistered = -60025

l static final int ECoreCalleeEmpty = -60026

l static final int ECoreInvalidUri = -60027

l static final int ECoreAudioVideoCodecEmpty = -60028

l static final int ECoreNoFreeDialogSession = -60029

l static final int ECoreCreateAudioChannelFailed = -60030

l static final int ECoreSessionTimerValueTooSmall = -60040

l static final int ECoreAudioHandleNull = -60041

l static final int ECoreVideoHandleNull = -60042

l static final int ECoreCallIsClosed = -60043

l static final int ECoreCallAlreadyHold = -60044

l static final int ECoreCallNotEstablished = -60045

l static final int ECoreCallNotHold = -60050

l static final int ECoreSipMessaegEmpty = -60051

l static final int ECoreSipHeaderNotExist = -60052

l static final int ECoreSipHeaderValueEmpty = -60053

l static final int ECoreSipHeaderBadFormed = -60054

l static final int ECoreBufferTooSmall = -60055

l static final int ECoreSipHeaderValueListEmpty = -60056

l static final int ECoreSipHeaderParserEmpty = -60057

l static final int ECoreSipHeaderValueListNull = -60058

l static final int ECoreSipHeaderNameEmpty = -60059

l static final int ECoreAudioSampleNotmultiple = -60060

l static final int ECoreAudioSampleOutOfRange = -60061

l static final int ECoreInviteSessionNotFound = -60062

l static final int ECoreStackException = -60063

l static final int ECoreMimeTypeUnknown = -60064

l static final int ECoreDataSizeTooLarge = -60065

l static final int ECoreSessionNumsOutOfRange = -60066

l static final int ECoreNotSupportCallbackMode = -60067

l static final int ECoreNotFoundSubscribeId = -60068

l static final int ECoreCodecNotSupport = -60069

l static final int ECoreCodecParameterNotSupport = -60070

l static final int ECorePayloadOutofRange = -60071

l static final int ECorePayloadHasExist = -60072

l static final int ECoreFixPayloadCantChange = -60073

l static final int ECoreCodecTypeInvalid = -60074

l static final int ECoreCodecWasExist = -60075

l static final int ECorePayloadTypeInvalid = -60076

l static final int ECoreArgumentTooLong = -60077

l static final int ECoreMiniRtpPortMustIsEvenNum = -60078

l static final int ECoreCallInHold = -60079

l static final int ECoreNotIncomingCall = -60080

l static final int ECoreCreateMediaEngineFailure = -60081

l static final int ECoreAudioCodecEmptyButAudioEnabled = -60082

l static final int ECoreVideoCodecEmptyButVideoEnabled = -60083

l static final int ECoreNetworkInterfaceUnavailable = -60084

l static final int ECoreWrongDTMFTone = -60085

l static final int ECoreWrongLicenseKey = -60086

l static final int ECoreTrialVersionLicenseKey = -60087

l static final int ECoreOutgoingAudioMuted = -60088

l static final int ECoreOutgoingVideoMuted = -60089

l static final int ECoreFailedCreateSdp = -60090

l static final int ECoreTrialVersionExpired = -60091

l static final int ECoreStackFailure = -60092

l static final int ECoreTransportExists = -60093

l static final int ECoreUnsupportTransport = -60094

l static final int ECoreAllowOnlyOneUser = -60095

l static final int ECoreUserNotFound = -60096

l static final int ECoreTransportsIncorrect = -60097

l static final int ECoreCreateTransportFailure = -60098

l static final int ECoreTransportNotSet = -60099

l static final int ECoreECreateSignalingFailure = -60100

l static final int ECoreArgumentIncorrect = -60101

l static final int ECoreIVRObjectNull = -61001

l static final int ECoreIVRIndexOutOfRange = -61002

l static final int ECoreIVRReferFailure = -61003

l static final int ECoreIVRWaitingTimeOut = -61004

l static final int EAudioFileNameEmpty = -70000

l static final int EAudioChannelNotFound = -70001

l static final int EAudioStartRecordFailure = -70002

l static final int EAudioRegisterRecodingFailure = -70003

l static final int EAudioRegisterPlaybackFailure = -70004

l static final int EAudioGetStatisticsFailure = -70005

l static final int EAudioPlayFileAlreadyEnable = -70006

l static final int EAudioPlayObjectNotExist = -70007

l static final int EAudioPlaySteamNotEnabled = -70008

l static final int EAudioRegisterCallbackFailure = -70009

l static final int EAudioCreateAudioConferenceFailure = -70010

l static final int EAudioOpenPlayFileFailure = -70011

l static final int EAudioPlayFileModeNotSupport = -70012

l static final int EAudioPlayFileFormatNotSupport = -70013

l static final int EAudioPlaySteamAlreadyEnabled = -70014

l static final int EAudioCreateRecordFileFailure = -70015

l static final int EAudioCodecNotSupport = -70016

l static final int EAudioPlayFileNotEnabled = -70017

l static final int EAudioPlayFileUnknowSeekOrigin = -70018

l static final int EAudioCantSetDeviceIdDuringCall =-70019

l static final int EAudioVolumeOutOfRange =-70020

l static final int EVideoFileNameEmpty = -80000

l static final int EVideoGetDeviceNameFailure = -80001

l static final int EVideoGetDeviceIdFailure = -80002

l static final int EVideoStartCaptureFailure = -80003

l static final int EVideoChannelNotFound = -80004

l static final int EVideoStartSendFailure = -80005

l static final int EVideoGetStatisticsFailure = -80006

l static final int EVideoStartPlayAviFailure = -80007

l static final int EVideoSendAviFileFailure = -80008

l static final int EVideoRecordUnknowCodec = -80009

l static final int EVideoCantSetDeviceIdDuringCall = -80010

l static final int EVideoUnsupportCaptureRotate = -80011

l static final int VideoUnsupportCaptureResolution = -80012

l static final int ECameraSwitchTooOften = -80013

l static final int EMTUOutOfRange = -80014

l static final int EDeviceGetDeviceNameFailure = -90001

The documentation for this class was generated from the following file:

l PortSipErrorcode.java

com.portsip.PortSipSdk Class Reference

Inherits AppRTCAudioManager.AudioManagerEvents.

Classes

l class MainHandler

Public Member Functions

l void onAudioDeviceChanged (AppRTCAudioManager.AudioDevice innerAdioDevice, Set<

AppRTCAudioManager.AudioDevice > innerSet)

l void onAudioFocusChange (int focusChange)

l PortSipSdk (Context context)

l 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)

l void unInitialize ()

l int setInstanceId (String instanceId)

l int setUser (String userName, String displayName, String authName, String password, String

userDomain, String SIPServer, int SIPServerPort, String STUNServer, int STUNServerPort,

String outboundServer, int outboundServerPort)

l void removeUser ()

remove user account info.

l int registerServer (int expires, int retryTimes)

l int refreshRegistration (int expires)

l int unRegisterServer (int waitMS)

l int setDisplayName (String displayName)

l int setLicenseKey (String key)

l int addAudioCodec (int enum_audiocodec)

l int addVideoCodec (int enum_videocodec)

l boolean isAudioCodecEmpty ()

l boolean isVideoCodecEmpty ()

l int setAudioCodecPayloadType (int enum_audiocodec, int payloadType)

l int setVideoCodecPayloadType (int enum_videocodec, int payloadType)

l void clearAudioCodec ()

l void clearVideoCodec ()

l int setAudioCodecParameter (int enum_audiocodec, String sdpParameter)

l int setVideoCodecParameter (int enum_videocodec, String sdpParameter)

l String getVersion ()

l int enableRport (boolean enable)

l int enableEarlyMedia (boolean enable)

Enable/disable rport(RFC3581).

l int enablePriorityIPv6Domain (boolean enable)

l int setUriUserEncoding (String character, boolean enable)

l int setReliableProvisional (int mode)

l int enable3GppTags (boolean enable)

l void enableCallbackSignaling (boolean enableSending, boolean enableReceived)

l void setSrtpPolicy (int enum_srtppolicy)

l int setRtpPortRange (int minimumRtpPort, int maximumRtpPort)

l int enableCallForward (boolean forBusyOnly, String forwardTo)

l int disableCallForward ()

l int enableSessionTimer (int timerSeconds)

l void disableSessionTimer ()

l void setDoNotDisturb (boolean state)

l void enableAutoCheckMwi (boolean state)

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

l int setKeepAliveTime (int keepAliveTime)

l int setAudioSamples (int ptime, int maxptime)

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

l String getSipMessageHeaderValue (String sipMessage, String headerName)

l long addSipMessageHeader (long sessionId, String methodName, int msgType, String

headerName, String headerValue)

l int removeAddedSipMessageHeader (long addedSipMessageId)

l void clearAddedSipMessageHeaders ()

l long modifySipMessageHeader (long sessionId, String methodName, int msgType, String

headerName, String headerValue)

l int removeModifiedSipMessageHeader (long modifiedSipMessageId)

l void clearModifiedSipMessageHeaders ()

l int setVideoDeviceId (int deviceId)

l int setVideoOrientation (int rotation)

l int enableVideoHardwareCodec (boolean enableHWEncoder, boolean enableHWDecoder)

l int setVideoResolution (int width, int height)

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

l int setVideoBitrate (long sessionId, int bitrateKbps)

l int setVideoFrameRate (long sessionId, int frameRate)

l int sendVideo (long sessionId, boolean send)

l int setRemoteVideoWindow (long sessionId, PortSIPVideoRenderer renderer)

l int setRemoteScreenWindow (long sessionId, PortSIPVideoRenderer renderer)

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

l int setVideoNackStatus (boolean state)

l int setChannelOutputVolumeScaling (long sessionId, int scaling)

l int setChannelInputVolumeScaling (long sessionId, int scaling)

l void enableAudioManager (boolean state)

l Set< PortSipEnumDefine.AudioDevice > getAudioDevices ()

l int setAudioDevice (PortSipEnumDefine.AudioDevice defaultDevice)

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

l int rejectCall (long sessionId, int code)

l int hangUp (long sessionId)

l int answerCall (long sessionId, boolean videoCall)

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

l int hold (long sessionId)

l int unHold (long sessionId)

l int muteSession (long sessionId, boolean muteIncomingAudio, boolean muteOutgoingAudio,

boolean muteIncomingVideo, boolean muteOutgoingVideo)

l int forwardCall (long sessionId, String forwardTo)

l long pickupBLFCall (String replaceDialogId, boolean videoCall)

l int sendDtmf (long sessionId, int enum_dtmfMethod, int code, int dtmfDuration, boolean

playDtmfTone)

l int refer (long sessionId, String referTo)

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

l int attendedRefer2 (long sessionId, long replaceSessionId, String replaceMethod, String target,

String referTo)

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

l long acceptRefer (long referId, String referSignaling)

l int rejectRefer (long referId)

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

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

l int enableSendVideoStreamToRemote (long sessionId, boolean state)

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

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

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

l void enableVideoStreamCallback (long sessionId, int enum_direction)

l int startRecord (long sessionId, String recordFilePath, String recordFileName, boolean

appendTimeStamp, int audioChannels, int enum_fileFormat, int enum_audioRecordMode, int

enum_videoRecordMode)

l int stopRecord (long sessionId)

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

l int stopPlayingFileToRemote (long sessionId)

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

l int stopPlayingFileLocally ()

l void audioPlayLoopbackTest (boolean enable)

l int createAudioConference ()

l int createVideoConference (PortSIPVideoRenderer conferenceVideoWindow, int videoWidth, int

videoHeight, int layout)

l void destroyConference ()

l int setConferenceVideoWindow (PortSIPVideoRenderer conferenceVideoWindow)

l int joinToConference (long sessionId)

l int removeFromConference (long sessionId)

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

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

l int enableAudioQos (boolean state)

l int enableVideoQos (boolean state)

l int setVideoMTU (int mtu)

l int getStatistics (long sessionId)

l void enableVAD (boolean state)

l void enableAEC (boolean state)

l void enableCNG (boolean state)

l void enableAGC (boolean state)

l void enableANS (boolean state)

l int sendOptions (String to, String sdp)

l int sendInfo (long sessionId, String mimeType, String subMimeType, String infoContents)

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

messageLength)

l long sendOutOfDialogMessage (String to, String mimeType, String subMimeType, boolean

isSMS, byte[] message, int messageLength)

l long setPresenceMode (int mode)

l long setDefaultSubscriptionTime (int secs)

l long setDefaultPublicationTime (int secs)

l long presenceSubscribe (String contact, String subject)

l int presenceTerminateSubscribe (long subscribeId)

l int presenceAcceptSubscribe (long subscribeId)

l int presenceRejectSubscribe (long subscribeId)

l int setPresenceStatus (long subscribeId, String statusText)

l long sendSubscription (String to, String eventName)

Send a SUBSCRIBE message to subscribe an event.

l int terminateSubscription (long subscribeId)

l void receiveSIPEvent (long sipCommand)

l void rtpPacketCallback (long sessionId, int mediaType, int directionMode, byte[] RTPPacket, int

packetSize)

l void audioRawCallback (long sessionId, int enum_audioCallbackMode, byte[] data, int

dataLength, int samplingFreqHz)

l void videoRawCallback (long sessionId, int enum_videoCallbackMode, int width, int height,

byte[] data, int dataLength)

l void setOnPortSIPEvent (OnPortSIPEvent l)

Protected Member Functions

l void finalize ()

Detailed Description

Author

Version

The documentation for this class was generated from the following file:

l PortSipSdk.java

com.portsip.PortSIPVideoRenderer Class Reference

Inherits SurfaceViewRenderer.

Classes

l enum ScalingType

Public Member Functions

l PortSIPVideoRenderer (Context context)

l PortSIPVideoRenderer (Context context, AttributeSet attrs)

l void setScalingType (ScalingType scalingType)

l void setVideoRotation (int rotation)

l void release ()

l void surfaceDestroyed (SurfaceHolder holder)

Detailed Description

Display the video stream on a SurfaceView.

Constructor & Destructor Documentation

com.portsip.PortSIPVideoRenderer.PortSIPVideoRenderer (Context context)

Standard View constructor. In order to render something, you must first call init().

com.portsip.PortSIPVideoRenderer.PortSIPVideoRenderer (Context context,

AttributeSet attrs)

Standard View constructor. In order to render something, you must first call init().

Member Function Documentation

void com.portsip.PortSIPVideoRenderer.setVideoRotation (int rotation)

Standard View constructor. In order to render something, you must first call init().

void com.portsip.PortSIPVideoRenderer.release ()

Block until any pending frame is returned and all GL resources released, even if an

interrupt occurs. If an interrupt occurs during release(), the interrupt flag will be set. This

function should be called before the Activity is destroyed and the EGLContext is still

valid. If you don't call this function, the GL resources might leak.

The documentation for this class was generated from the following file:

l PortSIPVideoRenderer.java

Index

acceptRefer

Refer functions, 53

Access SIP message header functions, 41

addSipMessageHeader, 41

clearAddedSipMessageHeaders, 42

clearModifiedSipMessageHeaders, 43

getSipMessageHeaderValue, 41

modifySipMessageHeader, 42

removeAddedSipMessageHeader, 41

removeModifiedSipMessageHeader, 42

addAudioCodec

Audio and video codecs functions, 32

Additional settings functions, 35

addSupportedMimeType, 39

disableCallForward, 37

disableSessionTimer, 38

enable3GppTags, 36

enableAutoCheckMwi, 38

enableCallbackSignaling, 37

enableCallForward, 37

enableEarlyMedia, 35

enablePriorityIPv6Domain, 36

enableRport, 35

enableSessionTimer, 38

getVersion, 35

setAudioSamples, 39

setDoNotDisturb, 38

setKeepAliveTime, 39

setReliableProvisional, 36

setRtpKeepAlive, 38

setRtpPortRange, 37

setSrtpPolicy, 37

setUriUserEncoding, 36

addSipMessageHeader

Access SIP message header functions, 41

addSupportedMimeType

Additional settings functions, 39

addVideoCodec

Audio and video codecs functions, 32

answerCall

Call functions, 49

attendedRefer

Refer functions, 52

attendedRefer2

Refer functions, 53

Audio and video codecs functions, 32

addAudioCodec, 32

addVideoCodec, 32

clearAudioCodec, 33

clearVideoCodec, 33

isAudioCodecEmpty, 32

isVideoCodecEmpty, 33

setAudioCodecParameter, 33

setAudioCodecPayloadType, 33

setVideoCodecParameter, 34

setVideoCodecPayloadType, 33

Audio and video functions, 44

displayLocalVideo, 46

enableAudioManager, 47

enableVideoHardwareCodec, 44

getAudioDevices, 47

sendVideo, 46

setAudioBitrate, 45

setAudioDevice, 47

setChannelInputVolumeScaling, 47

setChannelOutputVolumeScaling, 46

setRemoteScreenWindow, 46

setRemoteVideoWindow, 46

setVideoBitrate, 45

setVideoDeviceId, 44

setVideoFrameRate, 45

setVideoNackStatus, 46

setVideoOrientation, 44

setVideoResolution, 45

audio device changed,Play audio and video file

finished events, 23

onAudioDeviceChanged, 23

onAudioFocusChange, 23

onPlayFileFinished, 23

onStatistics, 23

Audio effect functions, 67

enableAEC, 67

enableAGC, 67

enableANS, 67

enableCNG, 67

enableVAD, 67

audioPlayLoopbackTest

Play audio and video file and

RTMP/RTSP stream functions, 61

call

Call functions, 48

Call events, 10

onDialogStateUpdated, 13

onInviteAnswered, 11

onInviteBeginingForward, 12

onInviteClosed, 12

onInviteConnected, 12

onInviteFailure, 12

onInviteIncoming, 10

onInviteRinging, 11

onInviteSessionProgress, 11

onInviteTrying, 11

onInviteUpdated, 12

onRemoteHold, 13

onRemoteUnHold, 13

Call functions, 48

answerCall, 49

call, 48

forwardCall, 50

hangUp, 49

hold, 49

muteSession, 50

pickupBLFCall, 50

rejectCall, 48

sendDtmf, 51

unHold, 50

updateCall, 49

clearAddedSipMessageHeaders

Access SIP message header functions, 42

clearAudioCodec

Audio and video codecs functions, 33

clearModifiedSipMessageHeaders

Access SIP message header functions, 43

clearVideoCodec

Audio and video codecs functions, 33

com.portsip.OnPortSIPEvent, 75

com.portsip.PortSIPCameraCapturer, 77

onCapturerStarted, 77

onCapturerStopped, 77

onFrameCaptured, 77

com.portsip.PortSipEnumDefine, 78

ENUM_DIRECTION_RECV, 79

ENUM_DIRECTION_SEND, 79

ENUM_DIRECTION_SEND_RECV, 79

ENUM_RECORD_MODE_BOTH, 79

ENUM_RECORD_MODE_NONE, 79

ENUM_RECORD_MODE_RECV, 79

ENUM_RECORD_MODE_SEND, 79

ENUM_VIDEOCODEC_I420, 79

ENUM_VIDEOCODEC_NONE, 79

com.portsip.PortSipEnumDefine.AUDIOCOD

EC, 73

com.portsip.PortSipEnumDefine.AudioDevice,

com.portsip.PortSipErrorcode, 80

com.portsip.PortSipSdk, 83

com.portsip.PortSIPVideoRenderer, 87

PortSIPVideoRenderer, 87

release, 87

setVideoRotation, 87

Conference functions, 62

createAudioConference, 62

createVideoConference, 62

destroyConference, 62

joinToConference, 63

removeFromConference, 63

setConferenceVideoWindow, 62

createAudioConference

Conference functions, 62

createVideoConference

Conference functions, 62

destroyConference

Conference functions, 62

disableCallForward

Additional settings functions, 37

disableSessionTimer

Additional settings functions, 38

displayLocalVideo

Audio and video functions, 46

DTMF events, 18

onRecvDtmfTone, 18

enable3GppTags

Additional settings functions, 36

enableAEC

Audio effect functions, 67

enableAGC

Audio effect functions, 67

enableANS

Audio effect functions, 67

enableAudioManager

Audio and video functions, 47

enableAudioQos

RTP and RTCP QOS functions, 64

enableAudioStreamCallback

RTP packets, Audio stream and video

stream callback, 57

enableAutoCheckMwi

Additional settings functions, 38

enableCallbackSignaling

Additional settings functions, 37

enableCallForward

Additional settings functions, 37

enableCNG

Audio effect functions, 67

enableEarlyMedia

Additional settings functions, 35

enablePriorityIPv6Domain

Additional settings functions, 36

enableRport

Additional settings functions, 35

enableRtpCallback

RTP packets, Audio stream and video

stream callback, 57

enableSendPcmStreamToRemote

Send audio and video stream functions,

enableSendVideoStreamToRemote

Send audio and video stream functions,

enableSessionTimer

Additional settings functions, 38

enableVAD

Audio effect functions, 67

enableVideoHardwareCodec

Audio and video functions, 44

enableVideoQos

RTP and RTCP QOS functions, 65

enableVideoStreamCallback

RTP packets, Audio stream and video

stream callback, 57

ENUM_DIRECTION_RECV

com.portsip.PortSipEnumDefine, 79

ENUM_DIRECTION_SEND

com.portsip.PortSipEnumDefine, 79

ENUM_DIRECTION_SEND_RECV

com.portsip.PortSipEnumDefine, 79

ENUM_RECORD_MODE_BOTH

com.portsip.PortSipEnumDefine, 79

ENUM_RECORD_MODE_NONE

com.portsip.PortSipEnumDefine, 79

ENUM_RECORD_MODE_RECV

com.portsip.PortSipEnumDefine, 79

ENUM_RECORD_MODE_SEND

com.portsip.PortSipEnumDefine, 79

ENUM_VIDEOCODEC_I420

com.portsip.PortSipEnumDefine, 79

ENUM_VIDEOCODEC_NONE

com.portsip.PortSipEnumDefine, 79

forwardCall

Call functions, 50

getAudioDevices

Audio and video functions, 47

getSipMessageHeaderValue

Access SIP message header functions, 41

getStatistics

RTP statistics functions, 66

getVersion

Additional settings functions, 35

hangUp

Call functions, 49

hold

Call functions, 49

INFO/OPTIONS message events, 19

onRecvInfo, 19

onRecvNotifyOfSubscription, 19

onRecvOptions, 19

initialize

Initialize and register functions, 28

initialize, 28

refreshRegistration, 30

registerServer, 30

setDisplayName, 30

setInstanceId, 29

setLicenseKey, 31

setUser, 29

unInitialize, 29

unRegisterServer, 30

isAudioCodecEmpty

Audio and video codecs functions, 32

isVideoCodecEmpty

Audio and video codecs functions, 33

joinToConference

Conference functions, 63

modifySipMessageHeader

Access SIP message header functions, 42

muteSession

Call functions, 50

MWI events, 17

onWaitingFaxMessage, 17

onWaitingVoiceMessage, 17

onACTVTransferFailure

Refer events, 15

onACTVTransferSuccess

Refer events, 15

onAudioDeviceChanged

audio device changed,Play audio and

video file finished events, 23

onAudioFocusChange

audio device changed,Play audio and

video file finished events, 23

onAudioRawCallback

RTP callback events, 25

onCapturerStarted

com.portsip.PortSIPCameraCapturer, 77

onCapturerStopped

com.portsip.PortSIPCameraCapturer, 77

onDialogStateUpdated

Call events, 13

onFrameCaptured

com.portsip.PortSIPCameraCapturer, 77

onInviteAnswered

Call events, 11

onInviteBeginingForward

Call events, 12

onInviteClosed

Call events, 12

onInviteConnected

Call events, 12

onInviteFailure

Call events, 12

onInviteIncoming

Call events, 10

onInviteRinging

Call events, 11

onInviteSessionProgress

Call events, 11

onInviteTrying

Call events, 11

onInviteUpdated

Call events, 12

onPlayFileFinished

audio device changed,Play audio and

video file finished events, 23

onPresenceOffline

Presence events, 20

onPresenceOnline

Presence events, 20

onPresenceRecvSubscribe

Presence events, 20

onReceivedRefer

Refer events, 14

onReceivedSignaling

Signaling events, 16

onRecvDtmfTone

DTMF events, 18

onRecvInfo

INFO/OPTIONS message events, 19

onRecvMessage

Presence events, 21

onRecvNotifyOfSubscription

INFO/OPTIONS message events, 19

onRecvOptions

INFO/OPTIONS message events, 19

onRecvOutOfDialogMessage

Presence events, 21

onReferAccepted

Refer events, 14

onReferRejected

Refer events, 14

onRegisterFailure

onRegisterSuccess

onRemoteHold

Call events, 13

onRemoteUnHold

Call events, 13

onRTPPacketCallback

RTP callback events, 25

onSendingSignaling

Signaling events, 16

onSendMessageFailure

Presence events, 21

onSendMessageSuccess

Presence events, 21

onSendOutOfDialogMessageFailure

Presence events, 22

onSendOutOfDialogMessageSuccess

Presence events, 22

onStatistics

audio device changed,Play audio and

video file finished events, 23

onSubscriptionFailure

Presence events, 22

onSubscriptionTerminated

Presence events, 22

onTransferRinging

Refer events, 15

onTransferTrying

Refer events, 14

onVideoRawCallback

RTP callback events, 26

onWaitingFaxMessage

MWI events, 17

onWaitingVoiceMessage

MWI events, 17

outOfDialogRefer

Refer functions, 53

pickupBLFCall

Call functions, 50

Play audio and video file and RTMP/RTSP

stream functions, 60

audioPlayLoopbackTest, 61

startPlayingFileLocally, 60

startPlayingFileToRemote, 60

stopPlayingFileLocally, 61

stopPlayingFileToRemote, 60

PortSIPVideoRenderer

com.portsip.PortSIPVideoRenderer, 87

Presence events, 20

onPresenceOffline, 20

onPresenceOnline, 20

onPresenceRecvSubscribe, 20

onRecvMessage, 21

onRecvOutOfDialogMessage, 21

onSendMessageFailure, 21

onSendMessageSuccess, 21

onSendOutOfDialogMessageFailure, 22

onSendOutOfDialogMessageSuccess, 22

onSubscriptionFailure, 22

onSubscriptionTerminated, 22

presenceAcceptSubscribe

Send OPTIONS/INFO/MESSAGE

functions, 70

presenceRejectSubscribe

Send OPTIONS/INFO/MESSAGE

functions, 71

presenceSubscribe

Send OPTIONS/INFO/MESSAGE

functions, 70

presenceTerminateSubscribe

Send OPTIONS/INFO/MESSAGE

functions, 70

Record functions, 58

startRecord, 58

stopRecord, 58

refer

Refer functions, 52

Refer events, 14

onACTVTransferFailure, 15

onACTVTransferSuccess, 15

onReceivedRefer, 14

onReferAccepted, 14

onReferRejected, 14

onTransferRinging, 15

onTransferTrying, 14

Refer functions, 52

acceptRefer, 53

attendedRefer, 52

attendedRefer2, 53

outOfDialogRefer, 53

refer, 52

rejectRefer, 53

refreshRegistration

Initialize and register functions, 30

onRegisterFailure, 9

onRegisterSuccess, 9

registerServer

Initialize and register functions, 30

rejectCall

Call functions, 48

rejectRefer

Refer functions, 53

release

com.portsip.PortSIPVideoRenderer, 87

removeAddedSipMessageHeader

Access SIP message header functions, 41

removeFromConference

Conference functions, 63

removeModifiedSipMessageHeader

Access SIP message header functions, 42

RTP and RTCP QOS functions, 64

enableAudioQos, 64

enableVideoQos, 65

setAudioRtcpBandwidth, 64

setVideoMTU, 65

setVideoRtcpBandwidth, 64

RTP callback events, 25

onAudioRawCallback, 25

onRTPPacketCallback, 25

onVideoRawCallback, 26

RTP packets, Audio stream and video stream

callback, 57

enableAudioStreamCallback, 57

enableRtpCallback, 57

enableVideoStreamCallback, 57

RTP statistics functions, 66

getStatistics, 66

SDK Callback events, 8

SDK functions, 27

Send audio and video stream functions, 55

enableSendPcmStreamToRemote, 55

enableSendVideoStreamToRemote, 56

sendPcmStreamToRemote, 55

sendVideoStreamToRemote, 56

Send OPTIONS/INFO/MESSAGE functions,

presenceAcceptSubscribe, 70

presenceRejectSubscribe, 71

presenceSubscribe, 70

presenceTerminateSubscribe, 70

sendInfo, 68

sendMessage, 69

sendOptions, 68

sendOutOfDialogMessage, 69

sendSubscription, 71

setDefaultPublicationTime, 70

setDefaultSubscriptionTime, 70

setPresenceMode, 69

setPresenceStatus, 71

terminateSubscription, 71

sendDtmf

Call functions, 51

sendInfo

Send OPTIONS/INFO/MESSAGE

functions, 68

sendMessage

Send OPTIONS/INFO/MESSAGE

functions, 69

sendOptions

Send OPTIONS/INFO/MESSAGE

functions, 68

sendOutOfDialogMessage

Send OPTIONS/INFO/MESSAGE

functions, 69

sendPcmStreamToRemote

Send audio and video stream functions,

sendSubscription

Send OPTIONS/INFO/MESSAGE

functions, 71

sendVideo

Audio and video functions, 46

sendVideoStreamToRemote

Send audio and video stream functions,

setAudioBitrate

Audio and video functions, 45

setAudioCodecParameter

Audio and video codecs functions, 33

setAudioCodecPayloadType

Audio and video codecs functions, 33

setAudioDevice

Audio and video functions, 47

setAudioRtcpBandwidth

RTP and RTCP QOS functions, 64

setAudioSamples

Additional settings functions, 39

setChannelInputVolumeScaling

Audio and video functions, 47

setChannelOutputVolumeScaling

Audio and video functions, 46

setConferenceVideoWindow

Conference functions, 62

setDefaultPublicationTime

Send OPTIONS/INFO/MESSAGE

functions, 70

setDefaultSubscriptionTime

Send OPTIONS/INFO/MESSAGE

functions, 70

setDisplayName

Initialize and register functions, 30

setDoNotDisturb

Additional settings functions, 38

setInstanceId

Initialize and register functions, 29

setKeepAliveTime

Additional settings functions, 39

setLicenseKey

Initialize and register functions, 31

setPresenceMode

Send OPTIONS/INFO/MESSAGE

functions, 69

setPresenceStatus

Send OPTIONS/INFO/MESSAGE

functions, 71

setReliableProvisional

Additional settings functions, 36

setRemoteScreenWindow

Audio and video functions, 46

setRemoteVideoWindow

Audio and video functions, 46

setRtpKeepAlive

Additional settings functions, 38

setRtpPortRange

Additional settings functions, 37

setSrtpPolicy

Additional settings functions, 37

setUriUserEncoding

Additional settings functions, 36

setUser

Initialize and register functions, 29

setVideoBitrate

Audio and video functions, 45

setVideoCodecParameter

Audio and video codecs functions, 34

setVideoCodecPayloadType

Audio and video codecs functions, 33

setVideoDeviceId

Audio and video functions, 44

setVideoFrameRate

Audio and video functions, 45

setVideoMTU

RTP and RTCP QOS functions, 65

setVideoNackStatus

Audio and video functions, 46

setVideoOrientation

Audio and video functions, 44

setVideoResolution

Audio and video functions, 45

setVideoRotation

com.portsip.PortSIPVideoRenderer, 87

setVideoRtcpBandwidth

RTP and RTCP QOS functions, 64

Signaling events, 16

onReceivedSignaling, 16

onSendingSignaling, 16

startPlayingFileLocally

Play audio and video file and

RTMP/RTSP stream functions, 60

startPlayingFileToRemote

Play audio and video file and

RTMP/RTSP stream functions, 60

startRecord

Record functions, 58

stopPlayingFileLocally

Play audio and video file and

RTMP/RTSP stream functions, 61

stopPlayingFileToRemote

Play audio and video file and

RTMP/RTSP stream functions, 60

stopRecord

Record functions, 58

terminateSubscription

Send OPTIONS/INFO/MESSAGE

functions, 71

unHold

Call functions, 50

unInitialize

Initialize and register functions, 29

unRegisterServer

Initialize and register functions, 30

updateCall

Call functions, 49