Documentation:SESDK:Developer:DefaultConfigSettings:8.5.2NET

Default SipEndpoint.config Settings

Using the Default Configuration File

You can find the default configuration file in the following location:

<installation folder>/Configuration/SipEndpoint.config

This file contains XML configuration details that affect how your SIP Endpoint SDK application behaves. The inital settings are the same as those specified for use with the QuickStart application that is included with your SIP Endpoint SDK release.

Configuration settings are separated into two containers: the Basic Container holds the connectivity details that are required to connect to your SIP Server, while the Genesys Container holds a variety of configuration settings.

Basic Container

The first Container ("Basic") holds the basic connectivity details that are required to connect to your SIP Server. This container has at least one connection (Connectivity) element with the following attributes:

<Connectivity user="DN" server="SERVER:PORT" protocol="TRANSPORT"/>

If you are using a configuration that supports Disaster Recovery and Geo-Redundancy, there may be multiple connection elements present with each specifying a separate possible connection. Refer to the configuration settings of that feature for details. You will have to make the following changes and save the updated configuration file before using the SIP Endpoint SDK:

• user="DN" — Supply a valid DN for the user attribute.
• server="SERVER:PORT" — Replace SERVER with the host name where your SIP Server is deployed, and PORT with the SIP port of the SIP Server host. (The default SIP port value is 5060.)
• protocol="TRANSPORT" — Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are UDP, TCP, or TLS.

Genesys Container

The second Container ("Genesys") holds a number of configurable settings that are organized into domains and sections. These settings do not have to be changed, but can be customized to take full control over your SIP Endpoint SDK applications.

An overview of the settings in this container and the valid values for these settings is provided here:

Domain	Section	Setting	Values	Description
policy
	endpoint
		audio_qos	Number	The integer value representing the DSCP bits to set for RTP audio packets. The value should be 4 * Preferred DSCP value.
		include_os_version_in_user_agent_header	Number	If set to 1, the user agent field includes the OS version the client is currently running on. Default: 1.
		include_sdk_version_in_user_agent_header	Number	If set to 1, the user agent field includes the SDK version the client is currently running on. Default: 1.
		ip_versions	IPv4 empty	A value of IPv4 or an empty value means that the application selects an available local IPv4 address. Default: IPv4. NOTE: This parameter has no effect if the public_address option specifies an explicit IP address.
		public_address	String	Local IP address of the machine. This setting can be an explicit setting or a special value that the SDK uses to automatically obtain the public address. Valid Values: This setting may have one of the following explicit values: An IP address. For example, 192.168.16.123 for IPv4. A bare host name, for example, epsipwin2. This setting may have one of the following special values: $auto—The SDK selects the first valid IP address on the first network adapter that is active (status=up) and has the default gateway configured. IP family preference is specified by the policy.endpoint.ip_versions setting. $ipv4—Same behavior as the $auto setting but the SDK restricts the address to a particular IP family. $host—The SDK retrieves the standard host name for the local computer using the gethostname system function. An adapter name or part of an adapter name prefixed with $. For example, $Local Area Connection 2 or $Local. The specified name must be different from the special values $auto, $ipv4, and $host. Default Value: Empty string which is fully equivalent to the $auto value. If the value is specified as an explicit host name, the Contact header includes the host name for the recipient of SIP messages (SIP Server or SIP proxy) to resolve on their own. For all other cases, including $host, the resolved IP address is used for Contact. The value in SDP is always the IP address.
		rtp_inactivity_timeout	Number	Timeout interval for RTP inactivity. Valid values are positive integers. A value of 0 means that this feature is not activated. A value 1 or higher indicates the inactivity timeout interval in seconds. Default: 30. Suggested values: 1 through 150.
		rtp_port_min	Number	The integer value representing the minimum value for an RTP port range. Must be within the valid port range of 8000 to 65535. If the minimum value is not specified or set to an invalid value, the default value of (8000) is used for rtp_port_min. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.
		rtp_port_max	Number	The integer value representing the maximum value for an RTP port range. Must be within the valid port range of 8000 to 65535. If the maximum value is not specified or set to an invalid value, the default maximum 9000 is used for rtp_port_max. Setting the maximum to a value that is less than the minimum is considered an error and results in a failure to initialize the endpoint.
		signaling_qos	Number	The integer value representing the DSCP bits to set for SIP packets. Integer value should be configured to 4 * Preferred DSCP value.
		sip_port_min	Number	The integer value representing the minimum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the minimum value is not specified or set to an invalid value, the default value of 5060 is used for sip_port_min. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.
		sip_port_max	Number	The integer value representing the maximum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the maximum value is not specified or set to an invalid value, the default value of 5080 is used for sip_port_max. Setting the maximum to a value that is less than the minimum is considered an error and will result in a failure to initialize the endpoint.
		sip_transaction_timeout	Number	SIP transaction timeout value in milliseconds. Valid values are 1 through 32000, with a default value of 4000. The recommended value is 4000.
		vq_report_collector		See Producing RTCP Extended Reports
		vq_report_publish		See Producing RTCP Extended Reports
		answer_sdp_priority	config NULL Empty offer	Valid values: config, NULL, or Empty—the endpoint selects the first codec from the codec configuration listed in both the codec configuration and the SDP offer. offer—the endpoint selects the first codec in the SDP offer listed in both the codec configuration and the SDP offer. Default value: config
		sip_port_binding	0 NULL Empty 1	Valid values: 0, NULL, or Empty—opens the SIP port to listen on any interface 1—the SIP port binds to the interface specified by the public_address setting and listens only on this IP address. Default value: 0
	session
		agc_mode	0 1	If set to 0, AGC (Automatic Gain Control) is disabled; if set to 1, it is enabled. Default: 1. Other values are reserved for future extensions. This configuration is applied at startup, after which time the agc_mode setting can be changed to 1 or 0 from the main sample application. NOTE: It is not possible to apply different AGC settings for different channels in multi-channel scenarios.
		auto_answer	0 1	If set to 1, all incoming calls should be answered automatically.
		auto_answer_delay	Number	Valid values: Number in milliseconds. Time in milliseconds to wait before auto-answering. The recommended and default value is 1500 milliseconds.
		dtmf_method	Rfc2833 Info InbandRtp	Method to send DTMF
		echo_control	0 1	Valid values: 0 or 1. If set to 1, echo control is enabled.
		noise_suppression	0 1	Valid values: 0 or 1. If set to 1, noise suppresion is enabled.
		dtx_mode	0 1	Valid values: 0 or 1. If set to 1, DTX is activated.
		reject_session_when_headset_na	0 1	Valid values: 0 or 1. If set to 1, the SDK should reject the incoming session if a USB headset is not available.
		sip_code_when_headset_na	Number	Defaul Value: 480 If a valid SIP error code is supplied, the SDK rejects the incoming session with the specified SIP error code if a USB headset is not available.
		vad_level	Number	Sets the degree of bandwidth reduction. Valid values: 0 – 3 — from 0 (conventional VAD) to 3 (aggressive high).
		ringing_enabled	Number	Valid values: 0, 1, 2, or 3. 0 = event Ringing disabled 1 = event Ringing enabled 2 = play ringtone internally (event Ringing disabled) 3 = play ringtone internally and enable event Ringing. Default Value: 3 Specifies whether to enable the ringing tone.
		ringing_timeout	Number	Valid Values: Empty, 0, or a positive number Default Value: 0 Specifies the duration, in seconds, of the ringing tone. If set to 0 or if the value is empty, the ringing time is unlimited.
		ringing_file	String	Valid values: Empty or the path to the ringing sound file. The path may be a file name in the current directory or the full path to the sound file. Default Value: ringing.wav Specifies the audio file that is played when the ringing tone is enabled with the ringing_enabled option. Note that WebRTC does not support MP3 playback. The ringtone file for built-in ringing should be a RIFF (little-endian) WAVE file using one of the following formats: kWavFormatPcm = 1, PCM, each sample of size bytes_per_sample kWavFormatALaw = 6, 8-bit ITU-T G.711 A-law kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law Uncompressed PCM audio must be 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.
	device
		audio_in_device For more information, see Audio Device Settings	String	Microphone device name
		audio_out_device	String	Speaker device name
		capture_device	String	Capture device name
		headset_name	String	The name of the headset model
		use_headset	Number	Valid values: 0 or 1. If set to 0, the audio devices specified in audio_in_device and audio_out_device are used by the SDK. If set to 1, the SDK uses a headset as the preferred audio input and output device and the audio devices specified in audio_in_device and audio_out_device are ignored.
codecs — See Working with Codec Priorities
proxies
	proxy<n>
		display_name	String	Proxy display name
		domain	String	Valid Values: Any valid SIP domain Default Value: Empty string A SIP domain is an application layer configuration defining the management domain of a SIP proxy. The configured value should include hostport and may include uri-parameters as defined by RFC 3261. The scheme, userinfo, and transport URI parameters are included automatically. If set to an empty string, SIP Endpoint SDK uses the parameters from the Connectivity section to construct the SIP domain value as it did in previous versions.
		password	String	Proxy password. Password configured for DN object.
		reg_interval	Number	The period, in seconds, after which the endpoint starts a new registration cycle when a SIP proxy is down. Valid values are integers greater than or equal to 0. If the setting is empty or negative, the default value is 0, which means no new registration cycle is allowed. If the setting is greater than 0, a new registration cycle is allowed and will start after the period specified by regInterval.
		reg_match_received_rport	Number	Valid Values: 0 or 1 Default Value: 0 This setting controls whether or not SIP Endpoint SDK should re-register itself when receiving a mismatched IP address in the received parameter of a REGISTER response. This helps resolve the case where SIP Endpoint SDK for OS X has multiple network interfaces and obtains the wrong local IP address. A value of 0 (default) disables this feature and a value of 1 enables re-registration.
		reg_timeout	Number	The period, in seconds, after which registration should expire. A new REGISTER request will be sent before expiration. Valid values are integers greater than or equal to 0. If the setting is 0, then registration is disabled, putting the endpoint in standalone mode. If the setting is empty/null, the default value of 1800 seconds is used.
	mailbox
		password	String	Mailbox password
		server	String	Proxy server address and port for this mailbox
		timeout	Number	Subscription expiration timeout in seconds. Valid values: any positive integer. If the setting is missing or set to 0, the default value of 1800 is used.
		transport	udp tcp tls	Transport protocol to use when communicating with server
		user	String	Mailbox number this line subscribes to.
system
	diagnostics
		enable_logging	Number	Valid values: 0 or 1. Disable or enable logging.
		log_file	String	Log file name, for example, SipEndpoint.log Valid values: Full path or relative path to the log file.
		log_level	Number	Valid values: 0 – 4. Log levels: 0 = "Fatal"; 1 = "Error"; 2 = "Warning"; 3 = "Info"; 4 = "Debug".
		log_options_provider	String	Valid values for webrtc = (warning, state, api, debug, info, error, critical). For example: gsip=2, webrtc=(error,critical)
		log_options_endpoint	Number	Valid values: 0 – 4, same as log_level. 10 = Logging disabled.
		logger_type	external	If set to external, an external logger is used.
	security
		cert_file	String	Full path pointing to the certificate file used as a client-side certificate for outgoing TLS connections and as a server-side certificate for incoming TLS connections. For example, /users/Desktop/certificate/hostname_cer.pem
		tls_enabled	Number	If set to 1, connection with TLS transport will be registered. Default: 0.
		use_srtp	String disabled optional mandatory	Indicates whether to use SRTP
		priv_key_file	String	Ful path to the endpoint private key file used for client-side authentication for outgoing TLS connections. For example, /users/Desktop/certificate/hostname_priv_key.pem
		ca_list_file	String	Full path to the Certificat Authority's list file. For example, /users/Desktop/certificate/ca/ca_cert.pem

Additional Configuration Options

The default configuration file may not contain all settings that may be used with the SIP Endpoint SDK; additional settings can be added to change certain behaviors. Check Configuring SIP Endpoint SDK for .NET for a discussion of these additional settings.