Contents
Configuration 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. For SRV resolution, specify the SRV record without including the port number in the server's URI. Also see SRV Resolution below.
- protocol="TRANSPORT"—Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are UDP, TCP, or TLS.
SRV Resolution
When using an SRV record for the server parameter, note the following:
- SIP Endpoint SDK selects the SRV target based on the priority field only and does not consider the weight field of an SRV record.
- You can not combine IPv4 and IPv6 for a single FQDN.
- The maximum number of targets (SRV records) per service is 20.
- You can only specify SRV records in the server parameter of the Connectivity element. You can not use SRV records for the mailbox section or the vq_report_collector setting.
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:
policy Domain
endpoint Section
audio_qos
Valid Values: Integer
Integer value representing the DSCP bits to set for RTP audio packets. Note: QoS is not supported for Windows Vista, Windows 7, or higher.
include_os_version_in_user_agent_header
Valid Values: 0, 1
Default Value: 1
If set to 1, the user agent field includes the OS version the client is currently running on.
include_sdk_version_in_user_agent_header
Valid Values: 0, 1
Default Value: 1
If set to 1, the user agent field includes the SDK version the client is currently running on.
ip_versions
Valid Values: IPv4, IPv6, IPv4, IPv6, IPv6, IPv4, or empty
Default Value: IPv4, IPv6
- IPv4—the application selects an available local IPv4 address; IPv6 addresses are ignored.
- IPv6—the application selects an available local IPv6 address; IPv4 addresses are ignored.
- IPv4,IPv6 or an empty—the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.
- IPv6,IPv4—the application selects an IPv6 address if one exists. If not, an available IPv4 address is selected.
Note: This parameter has no effect if the public_address option specifies an explicit IP address.
public_address
Valid Values: See description below
Default Value: Empty string which is fully equivalent to the $auto value
Local IP address or Fully Qualified Domain Name (FQDN) 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 or FE80::0202:B3FF:FE1E:8329 for IPv6.
- A bare host name or fully qualified domain name (FQDN). For example, epsipwin2 or epsipwin2.us.example.com.
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 or $ipv6—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.
- $fqdn—The SDK retrieves the fully qualified DNS name of the local computer. The SDK uses the GetComputerNameEx function with parameter ComputerNameDnsFullyQualified.
- 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, $host, and $fqdn.
If the value is an explicit host name, FQDN, or $fqdn, the Contact header includes the host name or FQDN 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. - $net:subnet - The SDK will select the IP address matching the given network ( from any local interface). The subnet is the full CIDR name as per RFC 4632. For example, $net:192.168.0.0/16.
include_mac_address
Valid Values: 0, 1
Default Value: 0
If set to 1, the MAC address is included in the Contact header of the REGISTER message of the host's network interface in a format compatible with RFC 5626.
refer_to_proxy
Valid Values: 0, 1
Default Value: 0
Specifies the destination of a referred INVITE.
- 0—Send the INVITE to the URL specified in the Refer-To header of the REFER message.
- 1—Send the INVITE to your configured SIP Proxy.
rtp_inactivity_timeout
Valid Values: 5-150
Default Value: 150
Suggested Value: 30
Timeout interval in seconds for RTP inactivity.
rtp_port_min
Valid Values: 9000-65535
The integer value representing the minimum value for an RTP port range. Must be within the valid port range of 9000 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (9000) and maximum (minimum value + 999) are used. 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
Valid Values: 9000-65535
The integer value representing the maximum value for an RTP port range. Must be within the valid port range of 9000 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (9000) and maximum (minimum value + 999) are used. 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.
tcp_port_min
Valid Values: 0-65535
The integer value representing the minimum value for a TCP client-side port range. Must be within the valid port range of 1 to 65535. If set to 0 (default) or if the configured range is not valid, SIP connections over TCP and TLS use ephemeral ports, assigned by the operating system.
tcp_port_max
Valid Values: 0-65535
The integer value representing the maximum value for a TCP client-side port range. Must be within the valid port range of 1 to 65535. If set to 0 (default) or if the configured range is not valid, SIP connections over TCP and TLS use ephemeral ports, assigned by the operating system.
If the value is non-zero and greater than the tcp_port_min value, this value specifies the maximum value for a TCP client-side SIP port range that will be used for all outgoing SIP connections over TCP and TLS transport.
signaling_qos
Valid Values: Integer
The integer value representing the DSCP bits to set for SIP packets. Note: QoS is not supported for Windows Vista, Windows 7, or higher.
sip_port_min
Valid Values: 1-65535
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 and maximum values are not specified or are set to an invalid value, the default minimum (5060) and maximum (minimum value + 6) are used. 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
Valid Values: 1-65535
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 minimum and maximum values are not specified or are set to an invalid value, the default minimum (5060) and maximum (minimum value + 6) are used. 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
Valid Values: 1-32000
Default Value: 4000
SIP transaction timeout value in milliseconds. Valid values are 1 through 32000, with a default value of 4000. The recommended value is 4000.
video_max_bitrate
Valid Values: Integer
Integer value representing the maximum video bitrate.
video_qos
Valid Values: Integer
The integer value representing the DSCP bits to set for RTP Video packets. Note: QoS is not supported for Windows Vista, Windows 7, or higher.
vq_report_collector
See Producing RTCP Extended Reports.
vq_report_publish
See Producing RTCP Extended Reports.
vq_alarm_threshold
Valid Values: 0 or a number from 1.0 to 5.0
Default Value: 0
Specifies the MOS threshold for generating Voice Quality Alarms. A 0 value disables the alarms. The recommended threshold value is 3.5. Genesys recommends that you avoid using values above 4.2 as an MOS that high might not be obtainable with some codecs, even in perfect network conditions.
webrtc_audio_layer
Valid Values: 0, 1, 2, 1000, 2000, 3000
Default Value: 0
Specifies which audio layer is used for WebRTC.
- 0 — The audio layer is defined by the GCTI_AUDIO_LAYER environment variable — Core audio is used if this environment variable is not specified.
- 1 — Wave audio layer is used.
- 2 — Core audio layer is used.
- 1000 — Instructs the audio layer to open the microphone channel when the endpoint starts up, using the audio layer type defined by option 0, and to keep it open until the endpoint is terminated.
- 2000 — Opens the speaker channel for the life of the endpoint, using the audio layer type defined by option 0. Eliminates any delay in opening the audio device when an incoming or outgoing call is connected, for example in environments where audio device startup is slow due to a required restart of the Windows MMCSS service.
- 3000 — Opens the microphone and speaker channels for the life of the endpoint, using the audio layer type defined by option 0.
answer_sdp_priority
Valid Values: config, offer
Default Value: config
- config—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.
sip_port_binding
Valid Values: 0, 1
Default Value: 0
- 0—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.
The sip_port_binding must be set to 0 for automatic IP address change detection to work properly. When set to 1 for specific deployments, agents have to re-register connections manually after an IP address change.
defer_device_release
Valid Values: Any integer
Default Value: 200
If set to a non-zero value, releasing of audio devices will be deferred for a given time (in milliseconds) after the audio stream has been stopped, to avoid any potential service interruptions when the audio is going to be quickly restarted, and if audio device operations are too slow on the user workstation or have other problems with restart. A zero value disables the deferred device release.
session Section
agc_mode
Valid Values: 0, 1
Default Value: 1
If set to 0, AGC (Automatic Gain Control) is disabled; if set to 1, it is enabled. 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_accept_video
Valid Values: 0, 1
This setting is only used in auto-answer scenarios when auto_answer=1.
If auto_accept_video is set to 1, both audio and video streams are accepted, otherwise incoming calls are answered as audio only, even if video is present in the offer.
auto_accept_video applies to a 3pcc answer when make-callrfc3275 is configured to 1 on the originating DN and a video codec is configured in the endpoint. auto_accept_video is not applied to a 3pcc answer when make-call-rfc3275 is configured to 2 on an originating DN, even if auto_accept_video is set to 1 and a video codec is configured in the endpoint.
auto_answer
Valid Values: 0, 1
If set to 1, all incoming calls should be answered automatically.
auto_answer_delay
Valid Values: Number in milliseconds
Default Value: 0
Time in milliseconds to wait before auto-answering (only applicable when auto_answer=1 and auto-answer is not blocked by missing headset).
callwait_tone_enabled
Valid Values: 0, 1
Default Value: 0
Specifies whether the call waiting tone is enabled for incoming calls, to be played when a new call arrives while user is on active call.
callwait_tone_file
Valid Values: Empty or the path to the call waiting tone sound file. The path can be a file in the current directory or the full path to the sound file.
Default Value: Empty
Specifies the audio file that is played when the call waiting tone is enabled with the callwait_tone_enabled option.
The call waiting tone file must 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 (8 KHz sampling rate)
- kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law (8 KHz sampling rate)
Uncompressed PCM audio must be 16-bit mono or stereo with a sampling rate of 8, 16, or 32 KHz.
dtmf_feedback
Valid Values: 0 (disable feedback), 1 (enable feedback)
Default Value: 1
Controls whether local feedback tone is played when Endpoint sends DTMF to remote party.
dtmf_method
Valid Values: Rfc2833, Info, InbandRtp
Method to send DTMF.
echo_control
Valid Values: 0, 1
If set to 1, echo control is enabled.
noise_suppression
Valid Values: 0, 1
If set to 1, noise suppresion is enabled.
dtx_mode
Valid Values: 0, 1
If set to 1, DTX is activated.
reject_session_when_headset_na
Valid Values: 0, 1
If the SDK is configured to use the headset setting (policy.device.use_headset=1) and the reject_session_when_headset_na option is set to 1, the SDK should reject the incoming session if a USB headset is not available.
sip_code_when_headset_na
Valid Values: SIP Error Code
Default 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
Valid Values: 0-3
Sets the degree of bandwidth reduction, from 0 for conventional VAD to 3 for aggressive high.
ringback_enabled
Valid Values: 0, 1, 2, 3, 4, 6
Default Value: 2
Specifies whether the ringback tone is enabled for outgoing calls.
- 0 — The ringback is not played when the INVITE dialog is not yet established. In scenarios where ringback is provided by Media Server, the ringback tone would be still present.
- 1 — The incoming media stream is played if provided by the Media gateway in a reliable provisional response with SDP.
- 2 — A local file is used for the ringback.
- 3 — The ringback is always played using either a local file or media provided by the gateway, if the provisional response is reliable.
- 4 — Same as 1, but the incoming media stream is played even if the provisional response from Media gateway is not reliable.
- 6 — The ringback is always played using either a local file or media provided by the gateway (regardless of whether the provisional response is reliable or not).
ringback_file
Valid Values: Empty or the path to the ringback sound file. The path can be a file in the current directory or the full path to the sound file.
Default Value: Empty
Specifies the audio file that is played when the ringing tone is enabled with the ringing_enabled option.
WebRTC does not support MP3 playback. The ringtone file for built-in ringback must 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 (8 KHz sampling rate)
- kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law (8 KHz sampling rate)
Uncompressed PCM audio must be 16-bit mono or stereo with a sampling rate of 8, 16, or 32 KHz.
ringing_enabled
Valid values: An integer between 0 to 7.
Default Value: 3
- 0—event Ringing disabled
- 1—event Ringing enabled
- 2—play ringtone internally (event Ringing disabled)
- 3—play ringtone internally and enable event Ringing.
- 4—play ringtone through a separate ringer device.
- 5—play ringtone through the ringer device and enable event Ringing.
- 6—play ringtone internally once for full duration (ringing_timeout is not used and the ringing does not stop when call is answered).
- 7—play ringtone once for full duration and enable event Ringing.
Suppressing the Ringtone
The ringtone is generated for all incoming calls to the Genesys SIP Endpoint SDK. To suppress the ringtone for third-party call control for the originating DN, configure the following SIP Server option:
- make-call-alert-info=<urn:alert:service:3pcc@genesys>
or
- make-call-alert-info=<file://null>;service=3pcc
ringing_timeout
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
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 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.
ringing_while_call_held
Valid Values: 0 or 1
Default Value: 1
If set to 0, when another call is held, playing ringtone is suppressed and call wait tone is played instead (if configured). If set to 1 (default value), ringtone is played whenever a new call arrives and there are no other active calls; held calls are not considered active in this case.
restart_audio_if_stuck
Valid Values: Empty, 0, 1
Default Value: 0
- 0 or Empty—disable auto restart for stuck audio
- 1—enable auto restart for stuck audio
reject_session_when_busy
Valid Values: Empty, 0, 1
Default Value: 0
- 0 or Empty—disable rejection of a session when busy
- 1—enable rejection of a session when busy
number_sessions_for_busy
Valid Values: Positive integer
Default Value: 1
Sets the number of sessions before busy. Must be a positive integer.
sip_code_when_busy
Valid Values: Empty, 4xx, 5xx, 6xx
Default value: Empty
SIP error response code to use when busy. Can be set to any valid SIP error response code in the 4xx, 5xx, or 6xx range, for example, 486.
rx_agc_mode
Valid Values: 0, 1
Default value: 0
When set to 1, the SDK enables the receiving-side AGC allowing the volume of the received RTP stream to be adjusted automatically. When set to 0 (default), the feature is disabled.
device Section
audio_in_device
Valid Values: A regex that matches the ECMAScript standard.
Microphone device name.
For more information, see Audio Device Settings
audio_out_device
Valid Values: A regex that matches the ECMAScript standard.
Speaker device name.
capture_device
Valid Values: A regex that matches the ECMAScript standard.
Capture device name.
headset_name
Valid Values: A regex that matches the ECMAScript standard.
The name of the headset model.
exclude_headset
Valid Values: A regex that matches the ECMAScript standard.
Default Value: Empty
When set, this will exclude devices whose name matches the regex pattern from being considered a valid headset for automatic device selection. Microphone and speaker parts of the excluded headset can still be selected manually (or even automatically, if no better device is found), but such a selection will not set "headset available" flag.
use_headset
Valid Values: 0, 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.
include_headset
Valid Values: Pair of device names or name parts with microphone and speaker names separated by colon, or comma-separated list of such pairs, for example: External Mic : Headphones
Default Value: Empty
When set, specifies the list of audio in/out devices to be considered as headset for automatic device selection, applicable to case when use_headset = 1. The names including delimiter character (quotes, colon or comma) must be enclosed in single or double quotes.
codecs Domain
See Working with Codec Priorities
proxies Domain
Configure a proxy section for each connectivity line. For example, for three connectivity lines, configure sections for proxy0, proxy1, and proxy2.
When the proxy section does not exist in the configuration file for a particular connectivity line, the framework takes the configurations settings from the proxy0 section. You can use this feature in use cases where the proxy sections are the same for all connectivity lines.
proxy Section
display_name
Valid Values: String
Proxy display name.
domain
Valid Values:Any valid SIP domain
Default Value: Empty
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 for .NET uses the parameters from the Connectivity section to construct the SIP domain value as it did in previous versions.
password
Valid Values: String
Proxy password.
reg_interval
Valid Values: Integer
Default Value: 0
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.
reg_match_received_rport
Valid Values: 0 or 1
Default Value: 0
This setting controls whether or not SIP Endpoint SDK should re-register itself when receiving an IP address (in the received parameter of a REGISTER response) that is different from the address supplied in the Contact header and does not match any local network interfaces. A value of 0 (default) disables this feature and a value of 1 enables re-registration.
Starting from 9.0.003, this setting is deprecated and is not recommended for use, unless suggested by Genesys Technical Support to fix specific problems. When the received parameter of a REGISTER response matches a local IP address, changing the IP address and re-registering is now done automatically.
reg_timeout
Valid Values: Number in seconds
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 or empty/null, then registration is disabled, putting the endpoint in standalone mode.
mailbox Sub-section
password
Valid Values: String
Mailbox password.
server
Valid Values: String
Proxy server address and port for this mailbox.
timeout
Valid Values: Number in seconds
Default Value: 1800
Subscription expiration timeout in seconds. If the setting is missing or set to 0, the SDK uses a default timeout of 1800 seconds (30 minutes).
transport
Valid Values: udp, tcp, tls
Transport protocol to use when communicating with the server.
user
Valid Values: String
Mailbox ID for this mailbox.
nat Sub-section
ice_enabled
Valid Values: Boolean
Enable or disable ICE.
stun_server
Valid Values: String
STUN server address. An empty or null value indicates this feature is not used.
stun_server_port
Valid Values: Valid port number
Default Value: 3478
STUN server port value.
turn_password
Valid Values: String
Password for TURN authentication.
turn_relay_type
Valid Values: 0, udp, 1, or tcp
Type of TURN relay.
- 0 or udp for TURN over UDP.
- 1 or tcp for TURN over TCP.
turn_server
Valid Values: String
TURN server address. An empty or null value indicates this feature is not used.
turn_server_port
Valid Values:Valid port number
Default Value: 3478
TURN server port value.
turn_user_name
Valid Values: String
User ID for TURN authorization
system Domain
diagnostics Section
enable_logging
Valid Values: 0 or 1
Default Value: 1
Disable or enable logging.
log_file
Valid Values: String
Log file name, for example, SipEndpoint.log.
log_filter
Valid Values: Empty, dtmf
Default Value: Empty
Specifies the list of log filters to be applied to hide sensitive data from the endpoint log. Currently the only supported filter is dtmf, which hides all occurrences of DTMF data from the log (by replacing entered digits with 'x').
log_level
Valid Values: 0-4
Default Value: 3
Log levels: 0 = "Fatal"; 1 = "Error"; 2 = "Warning"; 3 = "Info"; 4 = "Debug"
log_options_provider
Valid Values: Comma-separated list of log setting for various low-level components, including:
- gsip=N - Genesys SIP library log level, default is 2
- webrtc=(level) - Log level(s) for third-party WebRTC native code component. The default is error to include error messages and important WebRTC diagnostic, value api adds low-level API printouts.
Example value: gsip=2,webrtc=(error)
This settings control low-level debug information used for troubleshooting. Please don't change the value unless instructed by Genesys Technical Support engineer.
log_options_endpoint
Valid Values: 0-4, same as log_level
Default Value: 2
Log levels: 0 = "Fatal"; 1 = "Error"; 2 = "Warning"; 3 = "Info"; 4 = "Debug"
5 = Logging disabled.
This setting should not be set higher than log_level setting.
logger_type
Valid Values: file
If set to file the log data will be printed to the file specified by the log_file value.
log_segment
Valid Values: false, number, or number in KB,MB, or hr
Default Value: 10 MB
- false: No segmentation is allowed
- or KB: Size in kilobytes
- MB: Size in megabytes
- hr: Number of hours for segment to stay open
Specifies the segmentation limit for a log file. If the current log segment exceeds the size set by this option, the file is closed and a new one is created. This option is ignored if log output is not configured to be sent to a logfile.
log_expire
Valid Values: false, number, number file, number day
Deafult Value: 10 (store 10 log fragments and purge the rest)
- false: No expiration; all generated segments are stored.
- or file: Sets the maximum number of log files to store. Specify a number from 1—1000.
- day: Sets the maximum number of days before log files are deleted. Specify a number from 1—100
Determines whether log files expire. If they do, sets the measurement for determining when they expire, along with the maximum number of files (segments) or days before the files are removed. This option is ignored if log output is not configured to be sent to a log file.
log_time_convert
Valid Values: local, utc
Default Value: local
- local: The time of log record generation is expressed as a local time, based on the time zone and any seasonal adjustments. Time zone information of the application’s host computer is used.
- utc: The time of log record generation is expressed as Coordinated Universal Time (UTC).
Specifies the system in which an application calculates the log record time when generating a log file. The time is converted from the time in seconds since the Epoch (00:00:00 UTC, January 1, 1970).
log_time_format
Valid Values: time, locale, ISO8601
Default Value: time
- time: The time string is formatted according to the HH:MM:SS.sss (hours, minutes, seconds, and milliseconds) format
- locale: The time string is formatted according to the system’s locale.
- ISO8601: The date in the time string is formatted according to the ISO 8601 format. Fractional seconds are given in milliseconds.
Specifies how to represent, in a log file, the time when an application generates log records. A log record’s time field in the ISO 8601 format looks like this: 2001-07-24T04:58:10.123.
security Section
certificate
See Configuring certificate option for TLS.
dylib-path
Valid Values: Empty, path to dynamic library location
Default Value: Empty
Specifies the location of Genesys security module's dynamic libraries for TLS support. The name of the dynamic library in different processors are:
- libgsecurity_openssl_64.dylib for Intel processor
- libgsecurity_openssl_arm64.dylib for Apple processor
If the option is not specified or set to empty value (default), SIP Endpoint SDK requires the dynamic libraries to be located in the Resources folder of the application's main bundle
tls-target-name-check
See Configuring tls-target-name-check option for TLS.
use_srtp
Valid Values: optional, allowed, disabled, off, elective, both, enabled, force, mandatory
Indicates whether to use SRTP:
- optional or allowed—do not send secure offers, but accept them
- disabled or off—do not send secure offers and reject incoming secure offers
- elective or both—send both secure and non-secure offers and accept either
- enabled—send secure offers, accept both secure and non-secure offers
- force or mandatory—send secure offers, reject incoming non-secure offers
Adding either ',UNENCRYPTED_SRTCP' (long form) or ',UEC' (short form) to any value (for example, "enabled,UEC""), would result in the UNENCRYPTED_SRTCP parameter being added to that offer. When this parameter is negotiated, RTCP packets are not encrypted, but are still authenticated.
media Section
ringing_file
Valid Values: Empty, String file name
Default Value: ringing.mp3
The Ringing sound file name in the current directory or the full local path to the ringing sound file.
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.