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! Section
 
! Section
 
! Setting
 
! Setting
! Values
 
! Description
 
 
 
|-
 
|-
| colspan=5 | <div id="sdk_policy">'''policy'''</div>
+
| colspan=3 | '''policy'''
 
|-
 
|-
 
| rowspan=21|  
 
| rowspan=21|  
| colspan=4 | <div id="sdk_endpoint">'''endpoint'''</div>
+
| colspan=2 | '''endpoint'''
 
|-
 
|-
 
| rowspan=20 |  
 
| rowspan=20 |  
  
| audio_qos
+
| [[DefaultConfigSettings#audio_qos|audio_qos]]
| Number
 
| The 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
+
| [[DefaultConfigSettings#include_os_version_in_user_agent_header|include_os_version_in_user_agent_header]]
| <tt>0</tt><br/><tt>1</tt>
 
| 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
+
| [[DefaultConfigSettings#include_sdk_version_in_user_agent_header|include_sdk_version_in_user_agent_header]]
| <tt>0</tt><br/><tt>1</tt>
 
| If set to 1, the user agent field includes the SDK version the client is currently running on. Default: 1.
 
 
|-
 
|-
| ip_versions
+
| [[DefaultConfigSettings#ip_versions|ip_versions]]
| IPv4
 
IPv6<br>
 
IPv4,IPv6<br>
 
IPv6,IPv4<br>
 
empty
 
| A value of <tt>IPv4</tt> means that the application selects an available local IPv4 address; IPv6 addresses are ignored.
 
A value of <tt>IPv6</tt>  means that the application selects an available local IPv6 address; IPv4 addresses are ignored.<br>
 
A value of <tt>IPv4,IPv6</tt> or an empty value means that the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.<br>
 
A value of <tt>IPv6,IPv4</tt> means that the application selects an IPv6 address if one exists. If not, an available IPv4 address is selected.<br>
 
Default: <tt>IPv4,IPv6</tt>.<br>
 
NOTE: This parameter has no effect if the <tt>public_address</tt> option specifies an explicit IP address.
 
 
|-
 
|-
| public_address
+
| [[DefaultConfigSettings#public_address|public_address]]
| String
 
| 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:'''<br>
 
This setting may have one of the following explicit values:
 
 
 
*An IP address. For example, <tt>192.168.16.123</tt> for IPv4 or <tt>FE80::0202:B3FF:FE1E:8329</tt> for IPv6.
 
*A bare host name or fully qualified domain name (FQDN). For example, <tt>epsipwin2</tt> or <tt>epsipwin2.us.example.com</tt>.
 
 
 
This setting may have one of the following special values:
 
 
 
*<tt>$auto</tt>—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.
 
*<tt>$ipv4</tt> or <tt>$ipv6</tt>—Same behavior as the <tt>$auto</tt> setting but the SDK restricts the address to a particular IP family.
 
*<tt>$host</tt>—The SDK retrieves the standard host name for the local computer using the <tt>gethostname</tt> system function.
 
*<tt>$fqdn</tt>—The SDK retrieves the fully qualified DNS name of the local computer. The SDK uses the <tt>GetComputerNameEx</tt> function with parameter <tt>ComputerNameDnsFullyQualified</tt>.
 
*An adapter name or part of an adapter name prefixed with <tt>$</tt>. For example, <tt>$Local Area Connection 2</tt> or <tt>$Local</tt>. The specified name must be different from the special values <tt>$auto</tt>, <tt>$ipv4</tt>, <tt>$host</tt>, and <tt>$fqdn</tt>.
 
 
 
'''Default Value:''' Empty string which is fully equivalent to the <tt>$auto</tt> value.<br>
 
 
 
If the value is specified as an explicit host name, FQDN, or <tt>$fqdn</tt>, the <tt>Contact</tt> header includes the host name or FQDN for the recipient of SIP messages (SIP Server or SIP proxy) to resolve on their own. For all other cases, including <tt>$host</tt>, the resolved IP address is used for <tt>Contact</tt>. The value in SDP is always the IP address.
 
 
|-
 
|-
| {{AnchorDiv|refer_to_proxy}}
+
| [[DefaultConfigSettings#refer_to_proxy|refer_to_proxy]]
refer_to_proxy
 
| 0<br />
 
1
 
| Specifies the destination of a referred <tt>INVITE</tt>.<br />
 
Valid Values:<br />
 
<tt>0</tt>&mdash;Send the <tt>INVITE</tt> to the URL specified in the <b>Refer-To</b> header of the <tt>REFER</tt> message.<br />
 
<tt>1</tt>&mdash;Send the <tt>INVITE</tt> to your configured SIP Proxy.<br />
 
Default Value: <tt>0</tt>
 
 
|-
 
|-
| rtp_inactivity_timeout
+
| [[DefaultConfigSettings#rtp_inactivity_timeout|rtp_inactivity_timeout]]
| Number
 
| Timeout interval in seconds for RTP inactivity. Valid values are integers from <tt>5</tt> to <tt>150</tt>.<br/>
 
Default Value: <tt>150</tt><br/>
 
Suggested Value: <tt>30</tt>
 
 
|-
 
|-
| rtp_port_min
+
| [[DefaultConfigSettings#rtp_port_min|rtp_port_min]]
| Number
 
| The integer value representing the minimum value for an RTP port range. Must be within the valid port range of <tt>9000</tt> to <tt>65535</tt>. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (<tt>9000</tt>) and maximum (minimum value + <tt>999</tt>) 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
+
| [[DefaultConfigSettings#rtp_port_max|rtp_port_max]]
| Number
 
| The integer value representing the maximum value for an RTP port range. Must be within the valid port range of <tt>9000</tt> to <tt>65535</tt>. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (<tt>9000</tt>) and maximum (minimum value + <tt>999</tt>) 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.
 
 
|-
 
|-
| signaling_qos
+
| [[DefaultConfigSettings#signaling_qos|signaling_qos]]
| Number
 
| 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
+
| [[DefaultConfigSettings#sip_port_min|sip_port_min]]
| Number
 
| The integer value representing the minimum value for a SIP port range. Must be within the valid port range of <tt>1</tt> to <tt>65535</tt>. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (<tt>5060</tt>) and maximum (minimum value + <tt>6</tt>) 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
+
| [[DefaultConfigSettings#sip_port_max|sip_port_max]]
| Number
 
| The integer value representing the maximum value for a SIP port range. Must be within the valid port range of <tt>1</tt> to <tt>65535</tt>. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (<tt>5060</tt>) and maximum (minimum value + <tt>6</tt>) 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
+
| [[DefaultConfigSettings#sip_transaction_timeout|sip_transaction_timeout]]
| Number
 
| SIP transaction timeout value in milliseconds. Valid values are <tt>1</tt> through <tt>32000</tt>, with a default value of <tt>4000</tt>. The recommended value is <tt>4000</tt>.
 
 
|-
 
|-
| video_max_bitrate
+
| [[DefaultConfigSettings#video_max_bitrate|video_max_bitrate]]
| Number
 
| The integer value representing the maximum video bitrate.
 
 
|-
 
|-
| video_qos
+
| [[DefaultConfigSettings#video_qos|video_qos]]
| Number
 
| 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
+
| [[DefaultConfigSettings#vq_report_collector|vq_report_collector]]
|
 
| See [[ConfiguringforNET#Producing_RTCP_Extended_Reports|Producing RTCP Extended Reports]]
 
 
|-
 
|-
| vq_report_publish
+
| [[DefaultConfigSettings#vq_report_publish|vq_report_publish]]
|
 
| See [[ConfiguringforNET#Producing_RTCP_Extended_Reports|Producing RTCP Extended Reports]]
 
 
|-
 
|-
| webrtc_audio_layer
+
| [[DefaultConfigSettings#webrtc_audio_layer|webrtc_audio_layer]]
| 0<br>1<br>2
 
| Valid values:<br>
 
<tt>0</tt>—the audio layer is defined by environment variable "GCTI_AUDIO_LAYER"<br>
 
<tt>1</tt>—Wave audio layer is used<br>
 
<tt>2</tt>—Core audio layer is used
 
 
|-
 
|-
| answer_sdp_priority
+
| [[DefaultConfigSettings#answer_sdp_priority|answer_sdp_priority]]
| config<br>offer
 
| Valid values:<br>
 
<tt>config</tt>—the endpoint selects the first codec from the codec configuration listed in both the codec configuration and the SDP offer.<br>
 
<tt>offer</tt>—the endpoint selects the first codec in the SDP offer listed in both the codec configuration and the SDP offer.<br>
 
Default value: <tt>config</tt>
 
 
|-
 
|-
| sip_port_binding
+
| [[DefaultConfigSettings#sip_port_binding|sip_port_binding]]
| 0<br>1
 
| Valid values:<br>
 
<tt>0</tt>—opens the SIP port to listen on any interface<br>
 
<tt>1</tt>—the SIP port binds to the interface specified by the <tt>public_address</tt> setting and listens only on this IP address.<br>
 
 
 
Default value: <tt>0</tt>
 
 
|-
 
|-
 
| rowspan=19|
 
| rowspan=19|
| colspan=5 | <div id="sdk_session">'''session'''</div>
+
| colspan=3 | '''session'''
 
|-
 
|-
 
| rowspan=18 |  
 
| rowspan=18 |  
 
+
| [[DefaultConfigSettings#agc_mode|agc_mode]]
| 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 <tt>agc_mode</tt> setting can be changed to <tt>1</tt> or <tt>0</tt> 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
+
| [[DefaultConfigSettings#auto_accept_video|auto_accept_video]]
| Number
 
| This setting is only used in auto-answer scenarios, that is, in scenarios where <tt>auto_answer=1</tt>.
 
If <tt>auto_accept_video</tt> is set to 1, then both audio and video streams are accepted, otherwise incoming calls are answered as audio only, even if video is present in the offer.<br>
 
<tt>auto_accept_video</tt> applies to a 3pcc answer when <tt>make-callrfc3275</tt> is configured to 1 on the originating DN and a video codec is configured in the endpoint. <tt>auto_accept_video</tt> is not applied to a 3pcc answer when <tt>make-call-rfc3275</tt> is configured to 2 on an originating DN, even if <tt>auto_accept_video</tt> is set to 1 and a video codec is configured in the endpoint.
 
 
|-
 
|-
| auto_answer
+
| [[DefaultConfigSettings#auto_answer|auto_answer]]
| Number
 
| If set to 1, all incoming calls should be answered automatically.
 
 
|-
 
|-
| auto_answer_delay
+
| [[DefaultConfigSettings#auto_answer_delay|auto_answer_delay]]
| Number
 
| Valid values: Number in milliseconds.<br/>
 
Time in milliseconds to wait before auto-answering. The recommended and default value is <tt>1500</tt> milliseconds.
 
 
|-
 
|-
| dtmf_method
+
| [[DefaultConfigSettings#dtmf_method|dtmf_method]]
| Rfc2833
 
Info<br>
 
InbandRtp
 
| Method to send DTMF
 
 
|-
 
|-
| echo_control
+
| [[DefaultConfigSettings#echo_control|echo_control]]
| 0<br>1
 
| Valid values: 0 or 1. If set to 1, echo control is enabled.
 
 
|-
 
|-
| noise_suppression
+
| [[DefaultConfigSettings#noise_suppression|noise_suppression]]
| 0<br>1
 
| Valid values: 0 or 1. If set to 1, noise suppresion is enabled.
 
 
|-
 
|-
| dtx_mode
+
| [[DefaultConfigSettings#dtx_mode|dtx_mode]]
| Number
 
| Valid values: 0 or 1. If set to 1, DTX is activated.
 
 
|-
 
|-
| reject_session_when_headset_na
+
| [[DefaultConfigSettings#reject_session_when_headset_na|reject_session_when_headset_na]]
| Number
 
| 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
+
| [[DefaultConfigSettings#sip_code_when_headset_na|sip_code_when_headset_na]]
| Number
 
| Defaul Value: <tt>480</tt><br>
 
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
+
| [[DefaultConfigSettings#vad_level|vad_level]]
| Number
 
| Sets the degree of bandwidth reduction. Valid values: 0 &ndash; 3 &mdash; from 0 (conventional VAD) to 3 (aggressive high).
 
 
|-
 
|-
| ringing_enabled
+
| [[DefaultConfigSettings#ringing_enabled|ringing_enabled]]
| Number
 
| Valid values: 0, 1, 2, or 3.<br>
 
0 = event Ringing disabled<br>
 
1 = event Ringing enabled<br>
 
2 = play ringtone internally (event Ringing disabled)<br>
 
3 = play ringtone internally and enable event Ringing.<br>
 
Default Value: 1<br>
 
Specifies whether to enable the ringing tone.
 
 
|-
 
|-
| ringing_timeout
+
| [[DefaultConfigSettings#ringing_timeout|ringing_timeout]]
| Number
 
| Valid Values: Empty, 0, or a positive number<br>
 
Default Value: 0<br>
 
Specifies the duration, in seconds, of the ringing tone. If set to <tt>0</tt> or if the value is empty, the ringing time is unlimited.
 
 
|-
 
|-
| ringing_file
+
| [[DefaultConfigSettings#ringing_file|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.<br>
 
Default Value: <tt>ringing.wav</tt><br>
 
Specifies the audio file that is played when the ringing tone is enabled with the <tt>ringing_enabled</tt> option.<br>
 
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:<br>
 
 
 
kWavFormatPcm  = 1, PCM, each sample of size bytes_per_sample<br>
 
kWavFormatALaw  = 6, 8-bit ITU-T G.711 A-law<br>
 
kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law<br>
 
 
 
 
 
Uncompressed PCM audio must 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.
 
 
|-
 
|-
| restart_audio_if_stuck
+
| [[DefaultConfigSettings#restart_audio_if_stuck|restart_audio_if_stuck]]
| 0<br>Empty<br>1
 
| Valid values:<br>
 
<tt>0</tt> or Empty—disable auto restart for stuck audio<br>
 
<tt>1</tt>—enable auto restart for stuck audio<br>
 
Default value: <tt>0</tt>
 
 
|-
 
|-
| reject_session_when_busy
+
| [[DefaultConfigSettings#reject_session_when_busy|reject_session_when_busy]]
| 0<br>Empty<br>1
 
| Valid values:<br>
 
<tt>0</tt> or Empty—disable rejection of a session when busy<br>
 
<tt>1</tt>—enable rejection of a session when busy<br>
 
Default value:<tt>0</tt>
 
 
|-
 
|-
| number_sessions_for_busy
+
| [[DefaultConfigSettings#number_sessions_for_busy|number_sessions_for_busy]]
| Positive Integer
 
| Valid values: Empty, 0, or a positive integer<br>
 
Positive number setting the number of session before busy.<br>
 
Default value: <tt>1</tt>
 
 
|-
 
|-
| sip_code_when_busy
+
| [[DefaultConfigSettings#sip_code_when_busy|sip_code_when_busy]]
| 4xx<br>5xx<br>6xx<br>Empty
 
| Valid values: Empty or any valid SIP error response code.<br>
 
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, <tt>486</tt><br>
 
Default value: Empty
 
 
|-
 
|-
 
| rowspan=6 |
 
| rowspan=6 |
| colspan=4 | <div id="sdk_device">'''device'''</div>
+
| colspan=3 | '''device'''
 
|-
 
|-
 
| rowspan=5 |  
 
| rowspan=5 |  
 
+
| [[DefaultConfigSettings#audio_in_device|audio_in_device]]
| audio_in_device
 
  
 
For more information, see [[AudioDeviceSettings|Audio Device Settings]]
 
For more information, see [[AudioDeviceSettings|Audio Device Settings]]
| String
 
| Microphone device name
 
 
|-
 
|-
| audio_out_device
+
| [[DefaultConfigSettings#audio_out_device|audio_out_device]]
| String
 
| Speaker device name
 
 
|-
 
|-
| capture_device
+
| [[DefaultConfigSettings#capture_device|capture_device]]
| String
 
| Capture device name
 
 
|-
 
|-
| headset_name
+
| [[DefaultConfigSettings#headset_name|headset_name]]
| String
 
| The name of the headset model
 
 
|-
 
|-
| use_headset
+
| [[DefaultConfigSettings#use_headset|use_headset]]
| Number
 
| Valid values: 0 or 1. If set to 0, the audio devices specified in <tt>audio_in_device</tt> and <tt>audio_out_device</tt> 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 <tt>audio_in_device</tt> and <tt>audio_out_device</tt> are ignored.
 
 
|-
 
|-
| colspan=5 <div id="sdk_codec">'''codecs'''</div>&mdash; See [[ConfiguringforNET#Working_with_Codec_Priorities|Working with Codec Priorities]]
+
| colspan=3 |  '''codecs'''
 +
&mdash; See [[ConfiguringforNET#Working_with_Codec_Priorities|Working with Codec Priorities]]
 
|-
 
|-
| colspan=5 | <div id="sdk_proxies">'''proxies'''</div>
+
| colspan=3 | '''proxies'''
 
|-
 
|-
 
| rowspan=22 |
 
| rowspan=22 |
| colspan=4 | <div id="sdk_proxy">'''proxy''<n>'''''</div>
+
| colspan=2 | '''proxy''<n>'''''
 
|-
 
|-
 
| rowspan=21 |  
 
| rowspan=21 |  
| display_name
+
| [[DefaultConfigSettings#display_name|display_name]]
| String
 
| Proxy display name
 
 
|-
 
|-
| domain
+
| [[DefaultConfigSettings#domain|domain]]
| String
 
| Valid Values: Any valid SIP domain<br/>
 
Default Value: Empty string<br/><br/>
 
A SIP domain is an application layer configuration defining the management domain of a SIP proxy. The configured value should include <tt>hostport</tt> and may include <tt>uri-parameters</tt> as defined by RFC 3261. The scheme, <tt>userinfo</tt>, and <tt>transport</tt> URI parameters are included automatically.<br/><br/>
 
If set to an empty string, SIP Endpoint SDK for .NET uses the parameters from the <tt>Connectivity</tt> section to construct the SIP domain value as it did in previous versions.
 
 
|-
 
|-
| password
+
| [[DefaultConfigSettings#password|password]]
| String
 
| Proxy password
 
 
|-
 
|-
| reg_interval
+
| [[DefaultConfigSettings#reg_interval|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 <tt>0</tt>. If the setting is empty or negative, the default value is <tt>0</tt>, which means no new registration cycle is allowed. If the setting is greater than <tt>0</tt>, a new registration cycle is allowed and will start after the period specified by <tt>regInterval</tt>.
 
 
|-
 
|-
| reg_match_received_rport
+
| [[DefaultConfigSettings#reg_match_received_rport|reg_match_received_rport]]
| Number
 
| Valid Values: <tt>0</tt> or <tt>1</tt><br>
 
Default Value: <tt>0</tt><br>
 
This setting controls whether or not SIP Endpoint SDK should re-register itself when receiving a mismatched IP address in the <tt>received</tt> parameter of a <tt>REGISTER</tt> response. This helps resolve the case where SIP Endpoint SDK for .NET 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
+
| [[DefaultConfigSettings#reg_timeout|reg_timeout]]
| Number
 
| The period, in seconds, after which registration should expire. A new <tt>REGISTER</tt> request will be sent before expiration. Valid values are integers greater than or equal to <tt>0</tt>. If the setting is <tt>0</tt> or <tt>empty/null</tt>, then registration is disabled, putting the endpoint in standalone mode.
 
 
|-
 
|-
| colspan=3 | '''mailbox (sub-section of proxy<n>)'''
+
|'''mailbox (sub-section of proxy<n>)'''
 
|-
 
|-
| rowspan=5 |
+
| [[DefaultConfigSettings#password|password]]
 
 
| password
 
| String
 
| Mailbox password
 
 
|-
 
|-
| server
+
| [[DefaultConfigSettings#server|server]]
| String
 
| Proxy server address and port for this mailbox
 
 
|-
 
|-
| {{AnchorDiv|timeout}}timeout
+
| [[DefaultConfigSettings#timeout|timeout]]
| Number
 
| Subscription expiration timeout in seconds. If the setting is missing or set to <tt>0</tt>, the SDK uses a default timeout of 1800 seconds (30 minutes).<br/>
 
Valid Values: Any positive integer<br/>
 
 
|-
 
|-
| transport
+
| [[DefaultConfigSettings#transport|transport]]
| udp
 
tcp<br>
 
tls
 
| Transport protocol to use when communicating with server
 
 
|-
 
|-
| user
+
| [[DefaultConfigSettings#user|user]]
| String
 
| User ID for this mailbox
 
 
 
 
|-
 
|-
| colspan=3 | '''nat (sub-section of proxy<n>)'''
+
|'''nat (sub-section of proxy<n>)'''
 
|-
 
|-
| rowspan=8 |
+
| [[DefaultConfigSettings#ice_enabled|ice_enabled]]
 
 
| ice_enabled
 
| Boolean
 
| Enable or disable ICE
 
 
|-
 
|-
| stun_server
+
| [[DefaultConfigSettings#stun_server|stun_server]]
| String
 
| STUN server address. An empty or null value indicates this feature is not being used.
 
 
|-
 
|-
| stun_server_port
+
| [[DefaultConfigSettings#stun_server_port|stun_server_port]]
| String
 
| STUN server port value
 
 
|-
 
|-
| turn_password
+
| [[DefaultConfigSettings#turn_password|turn_password]]
| Number
 
| Password for TURN authentication
 
 
|-
 
|-
| turn_relay_type
+
| [[DefaultConfigSettings#turn_relay_type|turn_relay_type]]
| Number
 
| Type of TURN relay
 
 
|-
 
|-
| turn_server
+
| [[DefaultConfigSettings#turn_server|turn_server]]
| String
 
| TURN server address. An empty or null value indicates this feature is not being used.
 
 
|-
 
|-
| turn_server_port
+
| [[DefaultConfigSettings#turn_server_port|turn_server_port]]
| String
 
| TURN server port value
 
 
|-
 
|-
| turn_user_name
+
| [[DefaultConfigSettings#turn_user_name|turn_user_name]]
| String
 
| User ID for TURN authorization
 
 
|-
 
|-
| colspan=5 | <div id="sdk_system">'''system'''</div>
+
| colspan=3 | '''system'''
 
|-
 
|-
 
| rowspan=11 |
 
| rowspan=11 |
| colspan=4 | <div id="sdk_diagnostics">'''diagnostics'''</div>
+
| colspan=2 | '''diagnostics'''
 
|-
 
|-
 
| rowspan=10 |  
 
| rowspan=10 |  
 
+
| [[DefaultConfigSettings#enable_logging|enable_logging]]
| enable_logging
 
| Number
 
| Valid values: 0 or 1. Disable or enable logging.
 
|-
 
| log_file
 
| String
 
| Log file name, for example, <tt>SipEndpoint.log</tt>
 
 
|-
 
|-
| log_level
+
| [[DefaultConfigSettings#log_file|log_file]]
| Number
 
| Valid values: 0 &ndash; 4.  Log levels: 0 = "Fatal"; 1 = "Error"; 2 = "Warning"; 3 = "Info"; 4 = "Debug".
 
 
|-
 
|-
| log_options_provider
+
| [[DefaultConfigSettings#log_level|log_level]]
| String
 
| Valid values for webrtc = (warning, state, api, debug, info, error, critical). For example: <tt>gsip=2, webrtc=(error,critical)</tt>
 
 
|-
 
|-
| log_options_endpoint
+
| [[DefaultConfigSettings#log_options_provider|log_options_provider]]
| Number
 
| Default value: <tt>2</tt><br>
 
Valid values: <br/>
 
0 &ndash; 4, same as <tt>log_level</tt>.<br/>
 
5 = Logging disabled.<br/>
 
This setting should not be set higher than <tt>log_level</tt> setting.
 
 
|-
 
|-
| logger_type
+
| [[DefaultConfigSettings#log_options_endpoint|log_options_endpoint]]
| <tt>file</tt>
 
| If set to <tt>file</tt>, the log data will be printed to the file specified by the <tt>log_file</tt> parameter.
 
 
|-
 
|-
| log_segment
+
| [[DefaultConfigSettings#logger_type|logger_type]]
| <tt>false</tt><br>Number<br>Number in <tt>KB</tt>,<tt>MB</tt>, or <tt>hr</tt>
+
|-f
| Valid Values:<br>
+
| [[DefaultConfigSettings#log_segment|log_segment]]
<tt>false</tt>: No segmentation is allowed<br>
 
<tt><number></tt> or <tt><number></tt> KB: Size in kilobytes<br>
 
<tt><number></tt> MB: Size in megabytes<br>
 
<tt><number></tt> hr: Number of hours for segment to stay open<br>
 
Deafult Value: 10 MB<br>
 
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
+
| [[DefaultConfigSettings#log_expire|log_expire]]
| <tt>false</tt><br>Number<br>Number <tt>file</tt><br>Number <tt>day</tt>
 
| Valid Values:<br>
 
<tt>false</tt>: No expiration; all generated segments are stored.<br>
 
<tt><number></tt> or <tt><number></tt> file: Sets the maximum number of log files to store. Specify a number from <tt>1—1000</tt>.<br>
 
<tt><number></tt> day: Sets the maximum number of days before log files are deleted. Specify a number from <tt>1</tt>—<tt>100</tt><br>
 
Deafult Value: 10 (store 10 log fragments and purge the rest)<br>
 
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
+
| [[DefaultConfigSettings#log_time_convert|log_time_convert]]
| <tt>local</tt><br><tt>utc</tt>
 
| Valid Values:<br>
 
<tt>local</tt>: 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.<br>
 
<tt>utc</tt>: The time of log record generation is expressed as Coordinated Universal Time (UTC).<br>
 
Default Value: <tt>local</tt><br>
 
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
+
| [[DefaultConfigSettings#log_time_format|log_time_format]]
| <tt>time</tt><br><tt>locale</tt><br><tt>ISO8601</tt>
 
| Valid Values:<br>
 
<tt>time</tt>: The time string is formatted according to the HH:MM:SS.sss (hours, minutes, seconds, and milliseconds) format<br>
 
<tt>locale</tt>: The time string is formatted according to the system’s locale.<br>
 
<tt>ISO8601</tt>: The date in the time string is formatted according to the ISO 8601 format. Fractional seconds are given in milliseconds.<br>
 
Default Value: <tt>time</tt><br>
 
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: <tt>2001-07-24T04:58:10.123</tt>.
 
 
|-
 
|-
 
| rowspan=4 |
 
| rowspan=4 |
| colspan=4 | <div id="sdk_security">'''security'''</div>
+
| colspan=2| '''security'''
 
|-
 
|-
 
| rowspan=3 |  
 
| rowspan=3 |  
 
| cert_file
 
| String
 
| Thumbprint value of the Public endpoint certificate file, which is used as a client-side certificate for outgoing TLS connection and server-side certificate for incoming TLS connections. For example: <tt>78 44 34 36 7a c2 22 48 bd 5c 76 6b 00 84 5d 66 83 f5 85 d5</tt>
 
 
|-
 
|-
| tls_enabled
+
| [[DefaultConfigSettings#tls_enabled|tls_enabled]]
| Number
 
| If set to <tt>1</tt>, connection with TLS transport will be registered. Default: <tt>0</tt>.
 
 
|-
 
|-
| use_srtp
+
| [[DefaultConfigSettings#use_srtp|use_srtp]]
| String
 
disabled
 
optional<br>
 
mandatory
 
| Indicates whether to use SRTP
 
 
|-
 
|-
 
| rowspan=2|
 
| rowspan=2|
| colspan=4 | <div id="sdk_media">'''media'''</div>
+
| colspan=2| '''media'''
 
|-
 
|-
 
|
 
|
| ringing_file
+
| [[DefaultConfigSettings#ringing_file|ringing_file]]
| String
 
| Valid Values: Empty or String file name<br>
 
Defaul Value: <tt>ringing.mp3</tt><br>
 
The Ringing sound file name in the current directory or the full local path to the ringing sound file.
 
 
|}
 
|}
 +
 +
{{AnchorDiv|policy}}
 +
=='''policy''' Domain==
 +
 +
{{AnchorDiv|enpoint}}
 +
==='''endpoint''' Section===
 +
 +
{{AnchorDiv|audio_qos}}
 +
====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.
 +
 +
{{AnchorDiv|include_os_version_in_user_agent_header}}
 +
====include_os_version_in_user_agent_header====
 +
Valid Values: <tt>0</tt>, <tt>1</tt>
 +
Default Value: <tt>1</tt>
 +
 +
If set to 1, the user agent field includes the OS version the client is currently running on.
 +
 +
{{AnchorDiv|include_sdk_version_in_user_agent_header}}
 +
====include_sdk_version_in_user_agent_header====
 +
Valid Values: <tt>0</tt>, <tt>1</tt>
 +
Default Value: <tt>1</tt>
 +
 +
If set to 1, the user agent field includes the SDK version the client is currently running on.
 +
 +
{{AnchorDiv|ip_versions}}
 +
====ip_versions====
 +
Valid Values: <tt>IPv4</tt>, <tt>IPv6</tt>, <tt>IPv4</tt>, <tt>IPv6</tt>, <tt>IPv6</tt>, <tt>IPv4</tt>, or empty
 +
Default Value: <tt>IPv4</tt>, <tt>IPv6</tt>
 +
 +
 +
* IPv4&mdash;the application selects an available local IPv4 address; IPv6 addresses are ignored.
 +
* IPv6&mdash;the application selects an available local IPv6 address; IPv4 addresses are ignored.
 +
* IPv4,IPv6 or an empty&mdash;the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.
 +
* IPv6,IPv4&mdash;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.
 +
 +
{{AnchorDiv|public_address}}
 +
====public_address====
 +
Valid Values:
 +
 +
{{AnchorDiv|refer_to_proxy}}
 +
====refer_to_proxy====
 +
Valid Values:
 +
 +
{{AnchorDiv|rtp_inactivity_timeout}}
 +
====rtp_inactivity_timeout====
 +
Valid Values:
 +
 +
{{AnchorDiv|rtp_port_min}}
 +
====rtp_port_min====
 +
Valid Values:
 +
 +
{{AnchorDiv|rtp_port_max}}
 +
====rtp_port_max====
 +
Valid Values:
 +
 +
{{AnchorDiv|signaling_qos}}
 +
====signaling_qos====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_port_min}}
 +
====sip_port_min====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_port_max}}
 +
====sip_port_max====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_transaction_timeout}}
 +
====sip_transaction_timeout====
 +
Valid Values:
 +
 +
{{AnchorDiv|video_max_bitrate}}
 +
====video_max_bitrate====
 +
Valid Values:
 +
 +
{{AnchorDiv|video_qos}}
 +
====video_qos====
 +
Valid Values:
 +
 +
{{AnchorDiv|vq_report_collector}}
 +
====vq_report_collector====
 +
Valid Values:
 +
 +
{{AnchorDiv|vq_report_publish}}
 +
====vq_report_publish====
 +
Valid Values:
 +
 +
{{AnchorDiv|webrtc_audio_layer}}
 +
====webrtc_audio_layer====
 +
Valid Values:
 +
 +
{{AnchorDiv|answer_sdp_priority}}
 +
====answer_sdp_priority====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_port_binding}}
 +
====sip_port_binding====
 +
Valid Values:
 +
 +
 +
{{AnchorDiv|session}}
 +
==='''session''' Section===
 +
 +
{{AnchorDiv|agc_mode}}
 +
====agc_mode====
 +
Valid Values:
 +
 +
{{AnchorDiv|auto_accept_video}}
 +
====auto_accept_video====
 +
Valid Values:
 +
 +
{{AnchorDiv|auto_answer}}
 +
====auto_answer====
 +
Valid Values:
 +
 +
{{AnchorDiv|auto_answer_delay}}
 +
====auto_answer_delay====
 +
Valid Values:
 +
 +
{{AnchorDiv|dtmf_method}}
 +
====dtmf_method====
 +
Valid Values:
 +
 +
{{AnchorDiv|echo_control}}
 +
====echo_control====
 +
Valid Values:
 +
 +
{{AnchorDiv|noise_suppression}}
 +
====noise_suppression====
 +
Valid Values:
 +
 +
{{AnchorDiv|dtx_mode}}
 +
====dtx_mode====
 +
Valid Values:
 +
 +
{{AnchorDiv|reject_session_when_headset_na}}
 +
====reject_session_when_headset_na====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_code_when_headset_na}}
 +
====sip_code_when_headset_na====
 +
Valid Values:
 +
 +
{{AnchorDiv|vad_level}}
 +
====vad_level====
 +
Valid Values:
 +
 +
{{AnchorDiv|ringing_enabled}}
 +
====ringing_enabled====
 +
Valid Values:
 +
 +
{{AnchorDiv|ringing_timeout}}
 +
====ringing_timeout====
 +
Valid Values:
 +
 +
{{AnchorDiv|ringing_file}}
 +
====ringing_file====
 +
Valid Values:
 +
 +
{{AnchorDiv|restart_audio_if_stuck}}
 +
====restart_audio_if_stuck====
 +
Valid Values:
 +
 +
{{AnchorDiv|reject_session_when_busy}}
 +
====reject_session_when_busy====
 +
Valid Values:
 +
 +
{{AnchorDiv|number_sessions_for_busy}}
 +
====number_sessions_for_busy====
 +
Valid Values:
 +
 +
{{AnchorDiv|sip_code_when_busy}}
 +
====sip_code_when_busy====
 +
Valid Values:
 +
 +
 +
{{AnchorDiv|device}}
 +
==='''device''' Section===
 +
 +
{{AnchorDiv|audio_in_device}}
 +
====audio_in_device====
 +
Valid Values:
 +
For more information, see [[AudioDeviceSettings|Audio Device Settings]]
 +
 +
{{AnchorDiv|audio_out_device}}
 +
====audio_out_device====
 +
Valid Values:
 +
 +
{{AnchorDiv|capture_device}}
 +
====capture_device====
 +
Valid Values:
 +
 +
{{AnchorDiv|headset_name}}
 +
====headset_name====
 +
Valid Values:
 +
 +
{{AnchorDiv|use_headset}}
 +
====use_headset====
 +
Valid Values:
 +
 +
{{AnchorDiv|codecs}}
 +
=='''codecs''' Domain==
 +
 +
See [[ConfiguringforNET#Working_with_Codec_Priorities|Working with Codec Priorities]]
 +
 +
{{AnchorDiv|proxies}}
 +
 +
=='''proxies''' Domain==
 +
 +
{{AnchorDiv|proxyn}}
 +
===proxy<n> Section===
 +
<span id="sdk_proxy"></span>
 +
 +
{{AnchorDiv|display_name}}
 +
====display_name====
 +
Valid Values:
 +
 +
{{AnchorDiv|domain}}
 +
====domain====
 +
Valid Values:
 +
 +
{{AnchorDiv|password}}
 +
====password====
 +
Valid Values:
 +
 +
{{AnchorDiv|reg_interval}}
 +
====reg_interval====
 +
Valid Values:
 +
 +
{{AnchorDiv|reg_match_received_rport}}
 +
====reg_match_received_rport====
 +
Valid Values:
 +
 +
{{AnchorDiv|reg_timeout}}
 +
====reg_timeout====
 +
Valid Values:
 +
 +
==='''mailbox''' Sub-section===
 +
 +
{{AnchorDiv|password}}
 +
====password====
 +
Valid Values:
 +
 +
{{AnchorDiv|server}}
 +
====server====
 +
Valid Values:
 +
 +
{{AnchorDiv|timeout}}
 +
====timeout====
 +
Valid Values:
 +
 +
{{AnchorDiv|transport}}
 +
====transport====
 +
Valid Values:
 +
 +
{{AnchorDiv|user}}
 +
====user====
 +
Valid Values:
 +
 +
 +
=='''nat''' Sub-section==
 +
 +
{{AnchorDiv|ice_enabled}}
 +
====ice_enabled====
 +
Valid Values:
 +
 +
{{AnchorDiv|stun_server}}
 +
====stun_server====
 +
Valid Values:
 +
 +
{{AnchorDiv|stun_server_port}}
 +
====stun_server_port====
 +
Valid Values:
 +
 +
{{AnchorDiv|turn_password}}
 +
====turn_password====
 +
Valid Values:
 +
 +
{{AnchorDiv|turn_relay_type}}
 +
====turn_relay_type====
 +
Valid Values:
 +
 +
{{AnchorDiv|turn_server}}
 +
====turn_server====
 +
Valid Values:
 +
 +
{{AnchorDiv|turn_server_port}}
 +
====turn_server_port====
 +
Valid Values:
 +
 +
{{AnchorDiv|turn_user_name}}
 +
====turn_user_name====
 +
Valid Values:
 +
 +
{{AnchorDiv|System}}
 +
==<tt>system</tt> Domain==
 +
<span id="sdk_system"></span>
 +
 +
{{AnchorDiv|diagnostics}}
 +
==='''diagnostics''' Section===
 +
<span id="sdk_diagnostics"></span>
 +
 +
 +
{{AnchorDiv|enable_logging}}
 +
====enable_logging====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_file}}
 +
====log_file====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_level}}
 +
====log_level====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_options_provider}}
 +
====log_options_provider====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_options_endpoint}}
 +
====log_options_endpoint====
 +
Valid Values:
 +
 +
{{AnchorDiv|logger_type}}
 +
====logger_type====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_segment}}
 +
====log_segment====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_expire}}
 +
====log_expire====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_time_convert}}
 +
====log_time_convert====
 +
Valid Values:
 +
 +
{{AnchorDiv|log_time_format}}
 +
====log_time_format====
 +
Valid Values:
 +
 +
 +
{{AnchorDiv|security}}
 +
==='''security''' Section===
 +
<span id="sdk_security"></span>
 +
 +
 +
{{AnchorDiv|tls_enabled}}
 +
====tls_enabled====
 +
Valid Values:
 +
 +
{{AnchorDiv|use_srtp}}
 +
====use_srtp====
 +
Valid Values:
 +
 +
 +
{{AnchorDiv|media}}
 +
==='''media''' Section===
 +
<span id="sdk_media">'''media'''</span>
 +
 +
{{AnchorDiv|ringing_file}}
 +
====ringing_file====
 +
Valid Values:
  
 
== Additional Configuration Options ==
 
== Additional Configuration Options ==

Revision as of 23:51, January 27, 2017

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
policy
endpoint
audio_qos
include_os_version_in_user_agent_header
include_sdk_version_in_user_agent_header
ip_versions
public_address
refer_to_proxy
rtp_inactivity_timeout
rtp_port_min
rtp_port_max
signaling_qos
sip_port_min
sip_port_max
sip_transaction_timeout
video_max_bitrate
video_qos
vq_report_collector
vq_report_publish
webrtc_audio_layer
answer_sdp_priority
sip_port_binding
session
agc_mode
auto_accept_video
auto_answer
auto_answer_delay
dtmf_method
echo_control
noise_suppression
dtx_mode
reject_session_when_headset_na
sip_code_when_headset_na
vad_level
ringing_enabled
ringing_timeout
ringing_file
restart_audio_if_stuck
reject_session_when_busy
number_sessions_for_busy
sip_code_when_busy
device
audio_in_device

For more information, see Audio Device Settings

audio_out_device
capture_device
headset_name
use_headset
codecs

— See Working with Codec Priorities

proxies
proxy<n>
display_name
domain
password
reg_interval
reg_match_received_rport
reg_timeout
mailbox (sub-section of proxy<n>)
password
server
timeout
transport
user
nat (sub-section of proxy<n>)
ice_enabled
stun_server
stun_server_port
turn_password
turn_relay_type
turn_server
turn_server_port
turn_user_name
system
diagnostics
enable_logging
log_file
log_level
log_options_provider
log_options_endpoint
logger_type
log_segment
log_expire
log_time_convert
log_time_format
security
tls_enabled
use_srtp
media
ringing_file

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:

refer_to_proxy

Valid Values:

rtp_inactivity_timeout

Valid Values:

rtp_port_min

Valid Values:

rtp_port_max

Valid Values:

signaling_qos

Valid Values:

sip_port_min

Valid Values:

sip_port_max

Valid Values:

sip_transaction_timeout

Valid Values:

video_max_bitrate

Valid Values:

video_qos

Valid Values:

vq_report_collector

Valid Values:

vq_report_publish

Valid Values:

webrtc_audio_layer

Valid Values:

answer_sdp_priority

Valid Values:

sip_port_binding

Valid Values:


session Section

agc_mode

Valid Values:

auto_accept_video

Valid Values:

auto_answer

Valid Values:

auto_answer_delay

Valid Values:

dtmf_method

Valid Values:

echo_control

Valid Values:

noise_suppression

Valid Values:

dtx_mode

Valid Values:

reject_session_when_headset_na

Valid Values:

sip_code_when_headset_na

Valid Values:

vad_level

Valid Values:

ringing_enabled

Valid Values:

ringing_timeout

Valid Values:

ringing_file

Valid Values:

restart_audio_if_stuck

Valid Values:

reject_session_when_busy

Valid Values:

number_sessions_for_busy

Valid Values:

sip_code_when_busy

Valid Values:


device Section

audio_in_device

Valid Values: For more information, see Audio Device Settings

audio_out_device

Valid Values:

capture_device

Valid Values:

headset_name

Valid Values:

use_headset

Valid Values:

codecs Domain

See Working with Codec Priorities

proxies Domain

proxy<n> Section

display_name

Valid Values:

domain

Valid Values:

password

Valid Values:

reg_interval

Valid Values:

reg_match_received_rport

Valid Values:

reg_timeout

Valid Values:

mailbox Sub-section

password

Valid Values:

server

Valid Values:

timeout

Valid Values:

transport

Valid Values:

user

Valid Values:


nat Sub-section

ice_enabled

Valid Values:

stun_server

Valid Values:

stun_server_port

Valid Values:

turn_password

Valid Values:

turn_relay_type

Valid Values:

turn_server

Valid Values:

turn_server_port

Valid Values:

turn_user_name

Valid Values:

system Domain

diagnostics Section


enable_logging

Valid Values:

log_file

Valid Values:

log_level

Valid Values:

log_options_provider

Valid Values:

log_options_endpoint

Valid Values:

logger_type

Valid Values:

log_segment

Valid Values:

log_expire

Valid Values:

log_time_convert

Valid Values:

log_time_format

Valid Values:


security Section


tls_enabled

Valid Values:

use_srtp

Valid Values:


media Section

media

ringing_file

Valid Values:

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.

Comments or questions about this documentation? Contact us for support!