SIP specific account configuration. More...

#include <SIPConfig.h>

Inheritance diagram for ZDK_NET::SIPConfig:

Public Member Functions
	SIPConfig (ZDK::Shared::SIPConfig src)

bool	IsEqual (ZDK_NET::SIPConfig^ comp)
	Compares the current configuration with the given one. More...

virtual long long	Handle () override

virtual void	Initialize () override

virtual void	ReleaseReference () override

Public Attributes
ZDK::ISIPConfig *	cppRef = nullptr

Properties
System::String^	Domain `[get, set]`
	Sets the user domain. More...

bool	UseOutboundProxy `[get, set]`
	Configures the use of SIP Outbound as described in RFC 5626. More...

System::String^	OutboundProxy `[get, set]`
	Sets the SIP Outbound Proxy to be used. More...

System::String^	SipInstance `[get, set]`
	Configures RFC2141 URN for SIP registrations. More...

ZDK_NET::TransportType	Transport `[get, set]`
	Selects the SIP transport to be used. More...

System::String^	AuthUsername `[get, set]`
	Changes the user name used for SIP authentication. More...

System::String^	CallerID `[get, set]`
	Configures the Caller ID used as display name part in the address of record. More...

ZDK_NET::RPortType	RPort `[get, set]`
	Sets the use of rport for SIP users. More...

bool	EnableSRTP `[get, set]`
	Sets the use of user's SRTP. More...

bool	EnablePreconditions `[get, set]`
	Sets the use of call's preconditions framework as described in RFC 3312. More...

bool	EnableVideoFMTP `[get, set]`
	Sets video FMTP support. More...

ZDK_NET::RTCPFeedbackType	RTCPFeedback `[get, set]`
	Sets the RTCP Feedback support. More...

ZDK_NET::StunConfig^	Stun `[get, set]`
	Sets the STUN configuration. More...

ZDK_NET::MSRPConfig^	MSRP `[get, set]`
	Sets the MSRP configuration. More...

ZDK_NET::ZRTPConfig^	ZRTP `[get, set]`
	Sets the ZRTP configuration. More...

ZDK_NET::PushConfig^	Push `[get, set]`
	Sets the Push configuration. More...

bool	EnablePrivacy `[get, set]`
	Sets the use of Privacy mechanism as described in RFC 3323. More...

int	KeepAlive `[get, set]`
	Sets the keepalive interval for SIP accounts. More...

bool	HeaderDump `[get, set]`
	Sets the use of SIP Header dumps for SIP Calls and Registration event. More...

List< HeaderField^>^	AdditionalHeaders `[get, set]`
	Sets a list with SIP Header fields to be added to a set of SIP Request Methods. More...

int	CallAutoRejectTimeout `[get, set]`
	Sets the auto reject timeout for incoming calls. More...

bool	ConnectionPersistence `[get, set]`
	Configures connection persistence for a user. More...

ZDK_NET::RTPCollisionResolutionType	RTPCollisionResolution `[get, set]`
	Sets the desired RTP collision resolution. More...

ZDK_NET::SessionTimerModeType	SessionTimerMode `[get, set]`
	Changes the session timers setting according to RFC 4028. More...

int	SessionTimerExpiry `[get, set]`
	Changes the session timer expiry according to RFC 4028. More...

bool	EnableMediasec `[get, set]`
	Sets whether always to force the use of the mediasec. More...

Detailed Description

SIP specific account configuration.

Member Function Documentation

◆ IsEqual()

bool ZDK_NET::SIPConfig::IsEqual ( ZDK_NET::SIPConfig^ comp )

Compares the current configuration with the given one.

Parameters

[in] comp SIP configuration to be compared

Returns

0 - not equal
1 - equal

Property Documentation

◆ AdditionalHeaders

List< HeaderField^>^ ZDK_NET::SIPConfig::AdditionalHeaders

getset

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 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

[in] value The list with additional SIP headers

See also: HeaderField

◆ AuthUsername

System:: String^ ZDK_NET::SIPConfig::AuthUsername

getset

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

[in] value The new user name to use for authentication

◆ CallAutoRejectTimeout

int ZDK_NET::SIPConfig::CallAutoRejectTimeout

getset

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

[in] value The timeout in seconds. 0 turns this feature off (default: 120 seconds)

◆ CallerID

System:: String^ ZDK_NET::SIPConfig::CallerID

getset

Configures the Caller ID used as display name part in the address of record.

Parameters

[in] value The CallerID to be used

◆ ConnectionPersistence

bool ZDK_NET::SIPConfig::ConnectionPersistence

getset

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:

For SIP accounts with rport, it is always on.
For SIP accounts with SIP outbound, it is always on.
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

[in]

value

0 - disable
1 - enable

◆ Domain

System:: String^ ZDK_NET::SIPConfig::Domain

getset

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

[in] value The user's SIP Domain to be used

◆ EnableMediasec

bool ZDK_NET::SIPConfig::EnableMediasec

getset

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

[in]

value

0 - disabled
1 - enabled

◆ EnablePreconditions

bool ZDK_NET::SIPConfig::EnablePreconditions

getset

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

[in]

value

0 - disabled
1 - enabled

◆ EnablePrivacy

bool ZDK_NET::SIPConfig::EnablePrivacy

getset

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

[in]

value

0 - disable
1 - enable

◆ EnableSRTP

bool ZDK_NET::SIPConfig::EnableSRTP

getset

Sets the use of user's SRTP.

Parameters

[in]

value

0 - disabled
1 - enabled

◆ EnableVideoFMTP

bool ZDK_NET::SIPConfig::EnableVideoFMTP

getset

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

[in]

value

0 - disable
1 - enable

◆ HeaderDump

bool ZDK_NET::SIPConfig::HeaderDump

getset

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

[in]

value

0 - disable
1 - enable

◆ KeepAlive

int ZDK_NET::SIPConfig::KeepAlive

getset

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

[in]

value

Keepalive setting in seconds

-1 - Use the protocol's defaults
0 - Disable any keepalives
> 0 - Set the keepalive interval to that many seconds

◆ MSRP

ZDK_NET:: MSRPConfig^ ZDK_NET::SIPConfig::MSRP

getset

Sets the MSRP configuration.

Parameters

[in] value MSRP configuration

See also: MSRPConfig

◆ OutboundProxy

System:: String^ ZDK_NET::SIPConfig::OutboundProxy

getset

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

[in] value The SIP Outbound Proxy to be used

◆ Push

ZDK_NET:: PushConfig^ ZDK_NET::SIPConfig::Push

getset

Sets the Push configuration.

Parameters

[in] value Push configuration

See also: PushConfig

◆ RPort

ZDK_NET:: RPortType ZDK_NET::SIPConfig::RPort

getset

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

[in]

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

◆ RTCPFeedback

ZDK_NET:: RTCPFeedbackType ZDK_NET::SIPConfig::RTCPFeedback

getset

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

[in]

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

◆ RTPCollisionResolution

ZDK_NET:: RTPCollisionResolutionType ZDK_NET::SIPConfig::RTPCollisionResolution

getset

Sets the desired RTP collision resolution.

Parameters

[in] value The desired collision resolution (default: RTPCollisionResolutionType::Moderate)

See also: RTPCollisionResolutionType

◆ SessionTimerExpiry

int ZDK_NET::SIPConfig::SessionTimerExpiry

getset

Changes the session timer expiry according to RFC 4028.

Specifies the desired session expiration in seconds.

Parameters

[in] value Session expiration in seconds, if SessionTimerMode is not SessionTimerModeType::Disabled. Must be >= 90s.

See also: SessionTimerModeType, SessionTimerMode()

◆ SessionTimerMode

ZDK_NET:: SessionTimerModeType ZDK_NET::SIPConfig::SessionTimerMode

getset

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

[in] value The desired Session Timer Mode (default: SessionTimerModeType::Disabled)

See also: SessionTimerModeType, SessionTimerExpiry()

◆ SipInstance

System:: String^ ZDK_NET::SIPConfig::SipInstance

getset

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

[in] value The RFC 2141 URN to be used in RFC 5626 (SIP outbound)

See also: UserSipInstance()

◆ Stun

ZDK_NET:: StunConfig^ ZDK_NET::SIPConfig::Stun

getset

Sets the STUN configuration.

Parameters

[in] value STUN configuration

See also: StunConfig

◆ Transport

ZDK_NET:: TransportType ZDK_NET::SIPConfig::Transport

getset

Selects the SIP transport to be used.

Parameters

[in] value The signalling protocol to select (UDP, TCP, TLS)

See also: TransportType

◆ UseOutboundProxy

bool ZDK_NET::SIPConfig::UseOutboundProxy

getset

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

[in]

value

0 - disabled
1 - enabled

◆ ZRTP

ZDK_NET:: ZRTPConfig^ ZDK_NET::SIPConfig::ZRTP

getset

Sets the ZRTP configuration.

Parameters

[in] value ZRTP configuration

See also: ZRTPConfig

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

Configurations/SIPConfig.h

Public Member Functions

Public Attributes

Properties

Detailed Description

Member Function Documentation

◆ IsEqual()

Property Documentation

◆ AdditionalHeaders

◆ AuthUsername

◆ CallAutoRejectTimeout

◆ CallerID

◆ ConnectionPersistence

◆ Domain

◆ EnableMediasec

◆ EnablePreconditions

◆ EnablePrivacy

◆ EnableSRTP

◆ EnableVideoFMTP

◆ HeaderDump

◆ KeepAlive

◆ MSRP

◆ OutboundProxy

◆ Push

◆ RPort

◆ RTCPFeedback

◆ RTPCollisionResolution

◆ SessionTimerExpiry

◆ SessionTimerMode

◆ SipInstance

◆ Stun

◆ Transport

◆ UseOutboundProxy

◆ ZRTP