SIPConfig

java.lang.Object
- com.zoiper.zdk.Configurations.SIPConfig

public class SIPConfig
extends java.lang.Object

SIP specific account configuration

Constructor Summary

Constructors
Constructor and Description

SIPConfig(long handle)
Constructor storing the pointer/address to the underlying native object

Constructors
Constructor and Description
`SIPConfig(long handle)` Constructor storing the pointer/address to the underlying native object

Method Summary

All Methods Instance Methods Concrete Methods
Modifier and Type	Method and Description
`java.util.List<HeaderField>`	`additionalHeaders()` Gets a list with SIP Header fields to be added to a set of SIP Request Methods Adds a list of header fields with its value to the list of custom headers to be added to each SIP request specified in @p HeaderField::Method, made by this SIP user.
`void`	`additionalHeaders(java.util.List<HeaderField> value)` Sets a list with SIP Header fields to be added to a set of SIP Request Methods Adds a list of header fields with its value to the list of custom headers to be added to each SIP request specified in @p HeaderField::Method, made by this SIP user.
`java.lang.String`	`authUsername()` Gets the user name used for SIP authentication The user name to be used when responding to a SIP authentication challenge.
`void`	`authUsername(java.lang.String value)` Changes the user name used for SIP authentication Changes the user name to be used when responding to a SIP authentication challenge.
`int`	`callAutoRejectTimeout()` Gets the auto reject timeout for incoming calls Configures the auto reject timeout for incoming calls in case no action is applied to them (either answer or reject).
`void`	`callAutoRejectTimeout(int value)` Sets the auto reject timeout for incoming calls Configures the auto reject timeout for incoming calls in case no action is applied to them (either answer or reject).
`java.lang.String`	`callerID()` Gets the Caller ID used as display name part in the address of record
`void`	`callerID(java.lang.String value)` Configures the Caller ID used as display name part in the address of record
`boolean`	`connectionPersistence()` Configures connection persistence for a user Enables or disables the connection persistence option for a user (ENABLED by default).
`void`	`connectionPersistence(boolean value)` Configures connection persistence for a user Enables or disables the connection persistence option for a user (ENABLED by default).
`java.lang.String`	`domain()` Gets the user domain This is the domain part of the address of record.
`void`	`domain(java.lang.String value)` Sets the user domain This is the domain part of the address of record.
`boolean`	`enableMediasec()` Gets whether always to force the use of the mediasec Require the use of the "mediasec" draft specification.
`void`	`enableMediasec(boolean value)` Sets whether always to force the use of the mediasec Require the use of the "mediasec" draft specification.
`boolean`	`enablePreconditions()` Gets the use of call's preconditions framework as described in RFC 3312 The preconditions are used to negotiate network resources needed for a call before it starts ringing.
`void`	`enablePreconditions(boolean value)` Sets the use of call's preconditions framework as described in RFC 3312 The preconditions are used to negotiate network resources needed for a call before it starts ringing.
`boolean`	`enablePrivacy()` Gets the use of Privacy mechanism as described in RFC 3323 Whether the user's identity is revealed in the SIP URIs or anonymous one is used.
`void`	`enablePrivacy(boolean value)` Sets the use of Privacy mechanism as described in RFC 3323 Set whether the user's identity to be revealed in the SIP URIs or anonymous one to be used.
`boolean`	`enableSRTP()` Gets the use of user's SRTP
`void`	`enableSRTP(boolean value)` Sets the use of user's SRTP
`boolean`	`enableVideoFMTP()` Gets video FMTP support
`void`	`enableVideoFMTP(boolean value)` Sets video FMTP support This will take effect in calls created/received after the setting has changed.
`protected void`	`finalize()` Ensures the underlying native object destruction
`long`	`handle()` Gets the pointer/address to the underlying native object
`boolean`	`headerDump()` Gets the use of SIP Header dumps for SIP Calls and Registration event Enable or disable the SIP header fields dump for SIP calls and registration events for the specified user.
`void`	`headerDump(boolean value)` Sets the use of SIP Header dumps for SIP Calls and Registration event Enable or disable the SIP header fields dump for SIP calls and registration events for the specified user.
`boolean`	`isEqual(SIPConfig comp)` Compares the current configuration with the given one
`int`	`keepAlive()` Gets the keepalive interval for SIP accounts The keepalive is a SIP packet sent over the signaling socket containing only a new line (CRLF).
`void`	`keepAlive(int value)` Sets the keepalive interval for SIP accounts The keepalive is a SIP packet sent over the signaling socket containing only a new line (CRLF).
`MSRPConfig`	`msrp()` Gets the MSRP configuration
`void`	`msrp(MSRPConfig value)` Sets the MSRP configuration
`java.lang.String`	`outboundProxy()` Gets the configured SIP Outbound Proxy to be used The outbound proxy is a normal SIP proxy.
`void`	`outboundProxy(java.lang.String value)` Sets the SIP Outbound Proxy to be used The outbound proxy is a normal SIP proxy.
`PushConfig`	`push()` Gets the Push configuration
`void`	`push(PushConfig value)` Sets the Push configuration
`RPortType`	`rPort()` Gets the use of rport for SIP users Used to discover the public address and port in case there is a NAT between the user and the server.
`void`	`rPort(RPortType value)` Sets the use of rport for SIP users Used to discover the public address and port in case there is a NAT between the user and the server.
`RTCPFeedbackType`	`rtcpFeedback()` Gets the RTCP Feedback support The purpose of RTCP Feedback is to provide quick inbound status reporting for media streams, error correction and resilience against packet losses.
`void`	`rtcpFeedback(RTCPFeedbackType value)` Sets the RTCP Feedback support The purpose of RTCP Feedback is to provide quick inbound status reporting for media streams, error correction and resilience against packet losses.
`RTPCollisionResolutionType`	`rtpCollisionResolution()` Gets the desired RTP collision resolution
`void`	`rtpCollisionResolution(RTPCollisionResolutionType value)` Sets the desired RTP collision resolution
`int`	`sessionTimerExpiry()` Changes the session timer expiry according to RFC 4028.
`void`	`sessionTimerExpiry(int value)` Changes the session timer expiry according to RFC 4028.
`SessionTimerModeType`	`sessionTimerMode()` Changes the session timers setting according to RFC 4028.
`void`	`sessionTimerMode(SessionTimerModeType value)` Changes the session timers setting according to RFC 4028.
`java.lang.String`	`sipInstance()` Gets the configured RFC2141 URN for SIP registrations Gets the configures the SIP instance URN used for SIP outbound (RFC 5626).
`void`	`sipInstance(java.lang.String value)` Configures RFC2141 URN for SIP registrations Configures the SIP instance URN used for SIP outbound (RFC 5626).
`StunConfig`	`stun()` Gets the STUN configuration
`void`	`stun(StunConfig value)` Sets the STUN configuration
`TransportType`	`transport()` Gets the selected SIP transport
`void`	`transport(TransportType value)` Selects the SIP transport to be used
`boolean`	`useOutboundProxy()` Gets the use of SIP Outbound as described in RFC 5626 Enables or disables RFC 5626 for SIP users.
`void`	`useOutboundProxy(boolean value)` Configures the use of SIP Outbound as described in RFC 5626 Enables or disables RFC 5626 for SIP users.
`ZRTPConfig`	`zrtp()` Gets the ZRTP configuration
`void`	`zrtp(ZRTPConfig value)` Sets the ZRTP configuration

Methods inherited from class java.lang.Object
clone, equals, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait

- Constructor Detail
  - SIPConfig
```
public SIPConfig(long handle)
```
    Constructor storing the pointer/address to the underlying native object
    
    Parameters:
    
    handle - Pointer/address to the underlying native object
- Method Detail
  - handle
```
public long handle()
```
    Gets the pointer/address to the underlying native object
    
    Returns:
    
    Pointer/address to the underlying native object
  - finalize
```
protected void finalize()
```
    Ensures the underlying native object destruction
    
    Overrides:
    
    finalize in class java.lang.Object
  - domain
```
public java.lang.String domain()
```
    Gets the user domain This is the domain part of the address of record. It is a mandatory parameter and is used to construct the account's AoR and also used by the SIP stack to detect server settings (DNS SRV).
    
    Returns:
    
    The used user's SIP Domain
  - domain
```
public void domain(java.lang.String value)
```
    Sets the user domain This is the domain part of the address of record. It is a mandatory parameter and is used to construct the account's AoR and also used by the SIP stack to detect server settings (DNS SRV).
    
    Parameters:
    
    value - The user's SIP Domain to be used
  - useOutboundProxy
```
public boolean useOutboundProxy()
```
    Gets the use of SIP Outbound as described in RFC 5626 Enables or disables RFC 5626 for SIP users. The default setting for new SIP user account is "disabled". The user MUST also have a configured Outbound Proxy.
    Returns:
    0 - disabled
    1 - enabled
  - useOutboundProxy
```
public void useOutboundProxy(boolean value)
```
    Configures the use of SIP Outbound as described in RFC 5626 Enables or disables RFC 5626 for SIP users. The default setting for new SIP user account is "disabled". The user MUST also have a configured Outbound Proxy.
    Parameters:
    value -
    
    0 - disabled
    1 - enabled
  - outboundProxy
```
public java.lang.String outboundProxy()
```
    Gets the configured SIP Outbound Proxy to be used The outbound proxy is a normal SIP proxy. You configure your client, the phone or software, to use the proxy for all SIP sessions, just like when you configure your Web browser to use a Web proxy for all Web transactions. In some cases, the outbound proxy is placed alongside the firewall and is the only way to let SIP traffic pass from the internal network to the Internet.
    
    Returns:
    
    The configured SIP Outbound Proxy
  - outboundProxy
```
public void outboundProxy(java.lang.String value)
```
    Sets the SIP Outbound Proxy to be used The outbound proxy is a normal SIP proxy. You configure your client, the phone or software, to use the proxy for all SIP sessions, just like when you configure your Web browser to use a Web proxy for all Web transactions. In some cases, the outbound proxy is placed alongside the firewall and is the only way to let SIP traffic pass from the internal network to the Internet.
    
    Parameters:
    
    value - The SIP Outbound Proxy to be used
  - sipInstance
```
public java.lang.String sipInstance()
```
    Gets the configured RFC2141 URN for SIP registrations Gets the configures the SIP instance URN used for SIP outbound (RFC 5626). It MUST be a valid RFC 2141 URN and it SHOULD be a valid RFC 4122 UUID URN in the format "urn:uuid:xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx" where 'x' is a hexadecimal number. It can be optionally in angle brackets. Other URN schemes can be used. The recommended one is urn:uuid but most servers will treat the string as a sequence of bytes and will not care if the scheme is different. Use userSipInstance to create a random UUID. It will include the scheme prefix (urn:uuid:) and can be used directly in this function. Other ways of generating/obtaining a URN/UUID are allowed but the URN prefix is REQUIRED by the standard. Global URN S applied for all SIP accounts. SIP Outbound MUST still be enabled individually for each account. The URN SHOULD be stored locally and re-used after restarts. The API user has the responsibility of using the same URN between restarts.
    
    Returns:
    
    The RFC 2141 URN to be used in RFC 5626 (SIP outbound)
    
    See Also:
    
    userSipInstance
  - sipInstance
```
public void sipInstance(java.lang.String value)
```
    Configures RFC2141 URN for SIP registrations Configures the SIP instance URN used for SIP outbound (RFC 5626). It MUST be a valid RFC 2141 URN and it SHOULD be a valid RFC 4122 UUID URN in the format "urn:uuid:xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx" where 'x' is a hexadecimal number. It can be optionally in angle brackets. Other URN schemes can be used. The recommended one is urn:uuid but most servers will treat the string as a sequence of bytes and will not care if the scheme is different. Use userSipInstance to create a random UUID. It will include the scheme prefix (urn:uuid:) and can be used directly in this function. Other ways of generating/obtaining a URN/UUID are allowed but the URN prefix is REQUIRED by the standard. Global URN S applied for all SIP accounts. SIP Outbound MUST still be enabled individually for each account. The URN SHOULD be stored locally and re-used after restarts. The API user has the responsibility of using the same URN between restarts.
    
    Parameters:
    
    value - The RFC 2141 URN to be used in RFC 5626 (SIP outbound)
    
    See Also:
    
    userSipInstance
  - transport
```
public TransportType transport()
```
    Gets the selected SIP transport
    
    Returns:
    
    The selected signalling protocol (UDP, TCP, TLS)
    
    See Also:
    
    TransportType
  - transport
```
public void transport(TransportType value)
```
    Selects the SIP transport to be used
    
    Parameters:
    
    value - The signalling protocol to select (UDP, TCP, TLS)
    
    See Also:
    
    TransportType
  - authUsername
```
public java.lang.String authUsername()
```
    Gets the user name used for SIP authentication The user name to be used when responding to a SIP authentication challenge. The SIP user might be challenged on any SIP transaction (registration, call creation, etc). All authentication is handled automatically.
    
    Returns:
    
    The user name to use for authentication
  - authUsername
```
public void authUsername(java.lang.String value)
```
    Changes the user name used for SIP authentication Changes the user name to be used when responding to a SIP authentication challenge. The SIP user might be challenged on any SIP transaction (registration, call creation, etc). All authentication is handled automatically.
    
    Parameters:
    
    value - The new user name to use for authentication
  - callerID
```
public java.lang.String callerID()
```
    Gets the Caller ID used as display name part in the address of record
    
    Returns:
    
    The configured CallerID
  - callerID
```
public void callerID(java.lang.String value)
```
    Configures the Caller ID used as display name part in the address of record
    
    Parameters:
    
    value - The CallerID to be used
  - rPort
```
public RPortType rPort()
```
    Gets the use of rport for SIP users Used to discover the public address and port in case there is a NAT between the user and the server. It also helps for normal unfirewalled TCP and TLS connections (highly recommended for these two protocols). If rport is enabled for UDP connects along with STUN, STUN will be preferred. The default is to have rport disabled for UDP. A registration must be done to do a full discovery before making any calls if they are to benefit from rport.
    Returns:
    No Do not use rport.
    Signaling Enables usage of rport discovered public address for signaling negotiations.
    SignalingAndMedia Enables usage of rport discovered public address for both signaling and media negotiations. This option is NOT recommended. Enable it only if you absolutely know what you're doing.
    See Also:
    
    RPortType
  - rPort
```
public void rPort(RPortType value)
```
    Sets the use of rport for SIP users Used to discover the public address and port in case there is a NAT between the user and the server. It also helps for normal unfirewalled TCP and TLS connections (highly recommended for these two protocols). If rport is enabled for UDP connects along with STUN, STUN will be preferred. The default is to have rport disabled for UDP. A registration must be done to do a full discovery before making any calls if they are to benefit from rport.
    Parameters:
    value -
    
    No Do not use rport.
    Signaling Enables usage of rport discovered public address for signaling negotiations.
    Media Enables usage of rport discovered public address for media negotiations - This option is NOT recommended. Enable it only if you absolutely know what you're doing.
    SignalingAndMedia Enables usage of rport discovered public address for both signaling and media negotiations.
    See Also:
    
    RPortType
  - enableSRTP
```
public boolean enableSRTP()
```
    Gets the use of user's SRTP
    Returns:
    0 - disabled
    1 - enabled
  - enableSRTP
```
public void enableSRTP(boolean value)
```
    Sets the use of user's SRTP
    Parameters:
    value -
    
    0 - disabled
    1 - enabled
  - enablePreconditions
```
public boolean enablePreconditions()
```
    Gets the use of call's preconditions framework as described in RFC 3312 The preconditions are used to negotiate network resources needed for a call before it starts ringing. The negotiation is internal to the library without the need of special handling by the API user.
    Returns:
    0 - disabled
    1 - enabled
  - enablePreconditions
```
public void enablePreconditions(boolean value)
```
    Sets the use of call's preconditions framework as described in RFC 3312 The preconditions are used to negotiate network resources needed for a call before it starts ringing. The negotiation is internal to the library without the need of special handling by the API user.
    Parameters:
    value -
    
    0 - disabled
    1 - enabled
  - enableVideoFMTP
```
public boolean enableVideoFMTP()
```
    Gets video FMTP support
    Returns:
    0 - disabled
    1 - enabled
  - enableVideoFMTP
```
public void enableVideoFMTP(boolean value)
```
    Sets video FMTP support This will take effect in calls created/received after the setting has changed. Current calls will not be affected. Turned off by default.
    Parameters:
    value -
    
    0 - disable
    1 - enable
  - rtcpFeedback
```
public RTCPFeedbackType rtcpFeedback()
```
    Gets the RTCP Feedback support The purpose of RTCP Feedback is to provide quick inbound status reporting for media streams, error correction and resilience against packet losses. Currently only handling of packet loss is implemented (recovery from lost video configuration frames, resulting into "black" time) - Full Intra Request (FIR) and Picture Loss Indication (PLI).
    Returns:
    Off - Uses only AVP RTP Profile type - RTCP Feedbacks are OFF!
    Compatibility - Use both AVP and AVPF RTP Profile types - RTCP Feedbacks are ON (media is duplicated in SDP)
    On - Use only AVPF RTP Profile type - RTCP Feedbacks are ON (will establish media only if the remote peer also supports AVPF!)
    See Also:
    
    RTCPFeedbackType
  - rtcpFeedback
```
public void rtcpFeedback(RTCPFeedbackType value)
```
    Sets the RTCP Feedback support The purpose of RTCP Feedback is to provide quick inbound status reporting for media streams, error correction and resilience against packet losses. Currently only handling of packet loss is implemented (recovery from lost video configuration frames, resulting into "black" time) - Full Intra Request (FIR) and Picture Loss Indication (PLI).
    Parameters:
    value -
    
    Off - Uses only AVP RTP Profile type - RTCP Feedbacks are OFF!
    Compatibility - Use both AVP and AVPF RTP Profile types - RTCP Feedbacks are ON (media is duplicated in SDP)
    On - Use only AVPF RTP Profile type - RTCP Feedbacks are ON (will establish media only if the remote peer also supports AVPF!)
    See Also:
    
    RTCPFeedbackType
  - stun
```
public StunConfig stun()
```
    Gets the STUN configuration
    
    Returns:
    
    STUN configuration
    
    See Also:
    
    StunConfig
  - stun
```
public void stun(StunConfig value)
```
    Sets the STUN configuration
    
    Parameters:
    
    value - STUN configuration
    
    See Also:
    
    StunConfig
  - msrp
```
public MSRPConfig msrp()
```
    Gets the MSRP configuration
    
    Returns:
    
    MSRP configuration
    
    See Also:
    
    MSRPConfig
  - msrp
```
public void msrp(MSRPConfig value)
```
    Sets the MSRP configuration
    
    Parameters:
    
    value - MSRP configuration
    
    See Also:
    
    MSRPConfig
  - zrtp
```
public ZRTPConfig zrtp()
```
    Gets the ZRTP configuration
    
    Returns:
    
    ZRTP configuration
    
    See Also:
    
    ZRTPConfig
  - zrtp
```
public void zrtp(ZRTPConfig value)
```
    Sets the ZRTP configuration
    
    Parameters:
    
    value - ZRTP configuration
    
    See Also:
    
    ZRTPConfig
  - push
```
public PushConfig push()
```
    Gets the Push configuration
    
    Returns:
    
    Push configuration
    
    See Also:
    
    PushConfig
  - push
```
public void push(PushConfig value)
```
    Sets the Push configuration
    
    Parameters:
    
    value - Push configuration
    
    See Also:
    
    PushConfig
  - enablePrivacy
```
public boolean enablePrivacy()
```
    Gets the use of Privacy mechanism as described in RFC 3323 Whether the user's identity is revealed in the SIP URIs or anonymous one is used.
    Returns:
    0 - disabled
    1 - enabled
  - enablePrivacy
```
public void enablePrivacy(boolean value)
```
    Sets the use of Privacy mechanism as described in RFC 3323 Set whether the user's identity to be revealed in the SIP URIs or anonymous one to be used.
    Parameters:
    value -
    
    0 - disable
    1 - enable
  - keepAlive
```
public int keepAlive()
```
    Gets the keepalive interval for SIP accounts The keepalive is a SIP packet sent over the signaling socket containing only a new line (CRLF). It is automatically enabled for UDP sockets to keep alive any possible NAT mappings. It has the same effect as enabling STUN for this socket but unlike STUN it will always keep the connection alive.
    
    Returns:
    
    Keepalive set in seconds
  - keepAlive
```
public void keepAlive(int value)
```
    Sets the keepalive interval for SIP accounts The keepalive is a SIP packet sent over the signaling socket containing only a new line (CRLF). It is automatically enabled for UDP sockets to keep alive any possible NAT mappings. It has the same effect as enabling STUN for this socket but unlike STUN it will always keep the connection alive. Defaults (for -1) are 30 seconds for UDP and 180 seconds for TCP and TLS. This setting will be applied to all subsequent SIP requests. For best effects set this before registering.
    Parameters:
    value - Keepalive setting in seconds
    
    -1 - Use the protocol's defaults
    0 - Disable any keepalives
    > 0 - Set the keepalive interval to that many seconds
  - headerDump
```
public boolean headerDump()
```
    Gets the use of SIP Header dumps for SIP Calls and Registration event Enable or disable the SIP header fields dump for SIP calls and registration events for the specified user.
    Returns:
    0 - disable
    1 - enable
  - headerDump
```
public void headerDump(boolean value)
```
    Sets the use of SIP Header dumps for SIP Calls and Registration event Enable or disable the SIP header fields dump for SIP calls and registration events for the specified user.
    Parameters:
    value -
    
    0 - disable
    1 - enable
  - additionalHeaders
```
public java.util.List<HeaderField> additionalHeaders()
```
    Gets a list with SIP Header fields to be added to a set of SIP Request Methods Adds a list of header fields with its value to the list of custom headers to be added to each SIP request specified in @p HeaderField::Method, made by this SIP user. This could break the SIP request to the point that we will not even attempt to send it. Use with great care. This function will automatically reject to touch the following header fields: Via, Contact, From, To, CallID, CSeq
    
    Returns:
    
    The list with additional SIP headers
    
    See Also:
    
    HeaderField
  - additionalHeaders
```
public void additionalHeaders(java.util.List<HeaderField> value)
```
    Sets a list with SIP Header fields to be added to a set of SIP Request Methods Adds a list of header fields with its value to the list of custom headers to be added to each SIP request specified in @p HeaderField::Method, made by this SIP user. This could break the SIP request to the point that we will not even attempt to send it. Use with great care. This function will automatically reject to touch the following header fields: Via, Contact, From, To, CallID, CSeq
    
    Parameters:
    
    value - The list with additional SIP headers
    
    See Also:
    
    HeaderField
  - callAutoRejectTimeout
```
public int callAutoRejectTimeout()
```
    Gets the auto reject timeout for incoming calls Configures the auto reject timeout for incoming calls in case no action is applied to them (either answer or reject). Only works for calls started after calling this function.
    
    Returns:
    
    The timeout in seconds. 0 turns this feature off(default: 120 seconds)
  - callAutoRejectTimeout
```
public void callAutoRejectTimeout(int value)
```
    Sets the auto reject timeout for incoming calls Configures the auto reject timeout for incoming calls in case no action is applied to them (either answer or reject). Only works for calls started after calling this function.
    
    Parameters:
    
    value - The timeout in seconds. 0 turns this feature off (default: 120 seconds)
  - connectionPersistence
```
public boolean connectionPersistence()
```
    Configures connection persistence for a user Enables or disables the connection persistence option for a user (ENABLED by default). Connection persistence refers to the outbound TCP or TLS connection for signaling. This option has no effect in some configurations: 1. For SIP accounts with rport, it is always on. 2. For SIP accounts with SIP outbound, it is always on. 3. For accounts not using connection-oriented transports (SIP/UDP for example) it is always off. This is useful for TCP or TLS SIP users that have rport disabled, don't use SIP outbound but desire a connection to be kept alive and the server to be updated when the port changes (it will most probably be different on each connection).
    Returns:
    0 - disable
    1 - enable
  - connectionPersistence
```
public void connectionPersistence(boolean value)
```
    Configures connection persistence for a user Enables or disables the connection persistence option for a user (ENABLED by default). Connection persistence refers to the outbound TCP or TLS connection for signaling. This option has no effect in some configurations: 1. For SIP accounts with rport, it is always on. 2. For SIP accounts with SIP outbound, it is always on. 3. For accounts not using connection-oriented transports (SIP/UDP for example) it is always off. This is useful for TCP or TLS SIP users that have rport disabled, don't use SIP outbound but desire a connection to be kept alive and the server to be updated when the port changes (it will most probably be different on each connection).
    Parameters:
    value -
    
    0 - disable
    1 - enable
  - rtpCollisionResolution
```
public RTPCollisionResolutionType rtpCollisionResolution()
```
    Gets the desired RTP collision resolution
    
    Returns:
    
    The desired collision resolution (default: RTPCollisionResolutionType::Moderate)
    
    See Also:
    
    RTPCollisionResolutionType
  - rtpCollisionResolution
```
public void rtpCollisionResolution(RTPCollisionResolutionType value)
```
    Sets the desired RTP collision resolution
    
    Parameters:
    
    value - The desired collision resolution (default: RTPCollisionResolutionType::Moderate)
    
    See Also:
    
    RTPCollisionResolutionType
  - sessionTimerMode
```
public SessionTimerModeType sessionTimerMode()
```
    Changes the session timers setting according to RFC 4028. The RFC describes two general modes when session timers are not disabled: UAC and UAS. UAC means that the one who makes the call (the client or caller) will try to refresh the session periodically to make sure it is still alive. UAS means that the one who receives the call (the server or callee) will try to refresh the session. We have two more settings, "local" and "remote". When "local" is selected, we'll try to be the ones to do the refreshes. This means that we will use the "UAC" setting for outgoing calls and will prefer the "UAS" setting for incoming calls. The "remote" mode will do the opposite of the "local" mode. It will try to force the other end of the call to do the refreshes. The final decision is always at the one who provides the answer which does not always means this will be the callee (especially when the SIP call was done using the NVITE-no-offer mode). If the session timers are enabled we will always have a periodic refresh attempts, no matter if the remote end supports this feature. This is according to RFC 4028. If a re-NVITE (a refresh) fails the call is considered broken and will be closed with an error. The error can be modified by any proxies in between.
    
    Returns:
    
    The desired Session Timer Mode (default: SessionTimerModeType::Disabled)
    
    See Also:
    
    sessionTimerExpiry
  - sessionTimerMode
```
public void sessionTimerMode(SessionTimerModeType value)
```
    Changes the session timers setting according to RFC 4028. The RFC describes two general modes when session timers are not disabled: UAC and UAS. UAC means that the one who makes the call (the client or caller) will try to refresh the session periodically to make sure it is still alive. UAS means that the one who receives the call (the server or callee) will try to refresh the session. We have two more settings, "local" and "remote". When "local" is selected, we'll try to be the ones to do the refreshes. This means that we will use the "UAC" setting for outgoing calls and will prefer the "UAS" setting for incoming calls. The "remote" mode will do the opposite of the "local" mode. It will try to force the other end of the call to do the refreshes. The final decision is always at the one who provides the answer which does not always means this will be the callee (especially when the SIP call was done using the NVITE-no-offer mode). If the session timers are enabled we will always have a periodic refresh attempts, no matter if the remote end supports this feature. This is according to RFC 4028. If a re-NVITE (a refresh) fails the call is considered broken and will be closed with an error. The error can be modified by any proxies in between.
    
    Parameters:
    
    value - The desired Session Timer Mode (default: SessionTimerModeType::Disabled)
    
    See Also:
    
    sessionTimerExpiry
  - sessionTimerExpiry
```
public int sessionTimerExpiry()
```
    Changes the session timer expiry according to RFC 4028. Specifies the desired session expiration in seconds.
    
    Returns:
    
    Session expiration in seconds, if sessionTimerMode is not sessionTimerModeType::Disabled. Must be >= 90s.
    
    See Also:
    
    sessionTimerMode
  - sessionTimerExpiry
```
public void sessionTimerExpiry(int value)
```
    Changes the session timer expiry according to RFC 4028. Specifies the desired session expiration in seconds.
    
    Parameters:
    
    value - Session expiration in seconds, if sessionTimerMode is not sessionTimerModeType::Disabled. Must be >= 90s.
    
    See Also:
    
    sessionTimerMode
  - enableMediasec
```
public boolean enableMediasec()
```
    Gets whether always to force the use of the mediasec Require the use of the "mediasec" draft specification. This will always send client-initiated "Security-Client" headers with "mediasec". Will fail in the case where SRTP is not enabled.
    Returns:
    0 - disabled
    1 - enabled
  - enableMediasec
```
public void enableMediasec(boolean value)
```
    Sets whether always to force the use of the mediasec Require the use of the "mediasec" draft specification. This will always send client-initiated "Security-Client" headers with "mediasec". Will fail in the case where SRTP is not enabled.
    Parameters:
    value -
    
    0 - disabled
    1 - enabled
  - isEqual
```
public boolean isEqual(SIPConfig comp)
```
    Compares the current configuration with the given one
    Parameters:
    
    comp - SIP configuration to be compared
    
    Returns:
    0 - not equal
    1 - equal

Class SIPConfig

Constructor Summary

Method Summary

Methods inherited from class java.lang.Object

Constructor Detail

SIPConfig

Method Detail

handle

finalize

domain

domain

useOutboundProxy

useOutboundProxy

outboundProxy

outboundProxy

sipInstance

sipInstance

transport

transport

authUsername

authUsername

callerID

callerID

rPort

rPort

enableSRTP

enableSRTP

enablePreconditions

enablePreconditions

enableVideoFMTP

enableVideoFMTP

rtcpFeedback

rtcpFeedback

stun

stun

msrp

msrp

zrtp

zrtp

push

push

enablePrivacy

enablePrivacy

keepAlive

keepAlive

headerDump

headerDump

additionalHeaders

additionalHeaders

callAutoRejectTimeout

callAutoRejectTimeout

connectionPersistence

connectionPersistence

rtpCollisionResolution

rtpCollisionResolution

sessionTimerMode

sessionTimerMode

sessionTimerExpiry

sessionTimerExpiry

enableMediasec

enableMediasec

isEqual