Functions
String	com.portsip.PortSipSdk.getVersion ()

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

int	com.portsip.PortSipSdk.enableEarlyMedia (boolean enable)
	Enable/disable rport(RFC3581). More...

int	com.portsip.PortSipSdk.enableReliableProvisional (boolean enable)

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

void	com.portsip.PortSipSdk.enableCallbackSendingSignaling (boolean enable)

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

int	com.portsip.PortSipSdk.setRtpPortRange (int minimumRtpAudioPort, int maximumRtpAudioPort, int minimumRtpVideoPort, int maximumRtpVideoPort)

int	com.portsip.PortSipSdk.setRtcpPortRange (int minimumRtcpAudioPort, int maximumRtcpAudioPort, int minimumRtcpVideoPort, int maximumRtcpVideoPort)

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

int	com.portsip.PortSipSdk.disableCallForward ()

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

void	com.portsip.PortSipSdk.disableSessionTimer ()

void	com.portsip.PortSipSdk.setDoNotDisturb (boolean forBusyOnly)

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

int	com.portsip.PortSipSdk.setRtpKeepAlive (boolean state, int keepAlivePayloadType, int deltaTransmitTimeMS)

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

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

int	com.portsip.PortSipSdk.addSupportedMimeType (String methodName, String mimeType, String subMimeType)

Detailed Description

Function Documentation

◆ getVersion()

String com.portsip.PortSipSdk.getVersion ( )

Get the version number of the current SDK.

Returns: String with version description

◆ enableRport()

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.

◆ enableEarlyMedia()

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.

◆ enableReliableProvisional()

int com.portsip.PortSipSdk.enableReliableProvisional ( boolean enable )

Enable/Disable PRACK.

Parameters

enable Set to true to enable the SDK support PRACK. In default the PRACK is disabled.

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

◆ enable3GppTags()

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.

◆ enableCallbackSendingSignaling()

void com.portsip.PortSipSdk.enableCallbackSendingSignaling ( boolean enable )

Enable/disable the callback of the sent SIP messages.

Parameters

enable Set as true to enable the callback of the sent SIP messages, or false to disable it. Once enabled, the "onSendingSignaling" event will be triggered once the SDK sent a SIP message.

◆ setSrtpPolicy()

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.

◆ setRtpPortRange()

int com.portsip.PortSipSdk.setRtpPortRange	(	int	minimumRtpAudioPort,
		int	maximumRtpAudioPort,
		int	minimumRtpVideoPort,
		int	maximumRtpVideoPort
	)

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

Parameters

minimumRtpAudioPort	The minimum RTP port for audio stream.
maximumRtpAudioPort	The maximum RTP port for audio stream.
minimumRtpVideoPort	The minimum RTP port for video stream.
maximumRtpVideoPort	The maximum RTP port for video stream.

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.

◆ setRtcpPortRange()

int com.portsip.PortSipSdk.setRtcpPortRange	(	int	minimumRtcpAudioPort,
		int	maximumRtcpAudioPort,
		int	minimumRtcpVideoPort,
		int	maximumRtcpVideoPort
	)

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

Parameters

minimumRtcpAudioPort	The minimum RTCP port for audio stream.
maximumRtcpAudioPort	The maximum RTCP port for audio stream.
minimumRtcpVideoPort	The minimum RTCP port for video stream.
maximumRtcpVideoPort	The maximum RTCP port for video stream.

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.

◆ enableCallForward()

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:xxxx@.nosp@m.sip..nosp@m.ports.nosp@m.ip.c.nosp@m.om.

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

◆ disableCallForward()

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.

◆ enableSessionTimer()

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.

◆ disableSessionTimer()

void com.portsip.PortSipSdk.disableSessionTimer ( )

Disable the session timer.

◆ setDoNotDisturb()

void com.portsip.PortSipSdk.setDoNotDisturb ( boolean forBusyOnly )

Enable/disable the "Do not disturb" status.

Parameters

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

◆ enableAutoCheckMwi()

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.

◆ setRtpKeepAlive()

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;
keepAlivePayloadType	The payload type of the keep-alive RTP packet. It's usually set to 126.
deltaTransmitTimeMS	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.

◆ setKeepAliveTime()

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.

◆ setAudioSamples()

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.

◆ addSupportedMimeType()

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/

Functions

Detailed Description

Function Documentation

◆ getVersion()

◆ enableRport()

◆ enableEarlyMedia()

◆ enableReliableProvisional()

◆ enable3GppTags()

◆ enableCallbackSendingSignaling()

◆ setSrtpPolicy()

◆ setRtpPortRange()

◆ setRtcpPortRange()

◆ enableCallForward()

◆ disableCallForward()

◆ enableSessionTimer()

◆ disableSessionTimer()

◆ setDoNotDisturb()

◆ enableAutoCheckMwi()

◆ setRtpKeepAlive()

◆ setKeepAliveTime()

◆ setAudioSamples()

◆ addSupportedMimeType()