SIP phone trunk settings

AI-generated summary

Genesys Cloud provides comprehensive SIP phone trunk settings that enable organizations to manage telecommunications infrastructure through extensive configuration controls spanning basic setup, security, connection management, and diagnostic capabilities. The platform supports multiple transport protocols including UDP, TCP, and TLS with customizable listen ports (typically 8061 for TLS, 8060 for UDP/TCP) and registration features for monitoring trunk availability. Basic configuration requires assignment of Phone Trunk Name, Type selection (SIP), Trunk State configuration (defaulting to In-Service), Protocol selection, Listen Port specification, and Registration Rate parameters for SIP server support. Security capabilities are robust, encompassing SIP Access Control for inbound call restrictions through allow/deny lists of IP or CIDR addresses, with the Always Deny list taking precedence. TLS settings offer mutual authentication options, SSL/TLS version selection (v1 through v1.3, defaulting to v1.2), Far-End NAT Traversal for SRTP symmetric latching, and customizable cipher suite ordering using AES-based encryption. Subject alternative names can be specified using DNS, IP, URI, or email prefixes. Connection management features include language settings for incoming calls, configurable maximum concurrent calls and call rates (expressed as requests per time period), and maximum dial timeout specifications. Transport settings enable DSCP value configuration for QoS on RTP and RTCP packets (range 00-3F), management of retryable SIP reason codes (500-599 by default) and Q.850 cause codes, and TCP connection/idle timeout parameters. Genesys Cloud automatically populates a default list of retryable cause codes to facilitate failed call retry on alternate lines. Media configuration supports various audio codecs including g722, g729, PCMA, PCMU, and Opus, with three SDP offer methods (Normal, Delayed, Auto) and preferred codec selection capabilities. SRTP cipher suite options range from legacy AES_CM_128 to AES_CM_256 variants with HMAC-SHA1 authentication. An idle RTP disconnect feature can terminate calls when RTP packets are not received for extended periods, with extended timeout periods of 5 minutes for normal calls or 12 hours for unidirectional media. DTMF transmission methods include RTP Events (RFC 2833/4733), In-band Audio, or None, with configurable payload types (96–127, default 101). Authentication features encompass inbound digest authentication with configurable authorization methods (BYE, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE), realm specification, and username/password credentials. Protocol settings enable insertion of custom conversation headers with UUID values into SIP messages. Diagnostic capabilities include Media Capture and Protocol Capture functions for troubleshooting with Genesys Cloud Product Support, though both should be disabled for organizations using Secure IVR or with PCI DSS compliance enabled due to performance degradation, QoS impact, and logging of sensitive data. Protocol Capture generates PCAP files and should only be enabled under support guidance. The Capture Until setting allows specification of data collection duration through calendar and clock controls. Phone System Logging (Syslog) enables Edges connected to trunks to receive syslogs from phones on port 514 (default), requiring System Logging enablement on Base Settings or phone configuration. Provisioning options control HTTP authentication requirements, redirect preferences, HTTPS enforcement, external provision server URIs, and HTTP/HTTPS port configuration through centralized management. The Custom option permits Genesys Cloud Product Support personnel to modify External Trunk configurations for troubleshooting by configuring custom properties with Boolean, Text, Number, or List data types. All advanced settings should only be modified as directed by support personnel to ensure system integrity and compliance.

When you configure a phone trunk for SIP phones, you’ll need to configure several basic settings. Depending on your requirements, you may also need to configure some of the more advanced settings. This reference describes all the settings that you’ll find on the Create/Edit Phone Trunk page for SIP phones.

Setting	Description
Phone Trunk Name	Use this field to assign the phone trunk a descriptive name.
Type	Use this drop down to select SIP as the phone trunk type,
Trunk State	Use this switch to change the operational state of the phone trunk. The default setting is In-Service.
Protocol	Use this drop-down to choose the trunk transport protocol variant. There are three choices for the trunk transport protocol: UDP TCP TLS
Listen Port	Use this field to specify the trunk transport listen port. Common values for this setting are 8061 for TLS and 8060 for UDP and TCP.

Registration

If your SIP server or proxy server provides support for registration, then the status will be updated accordingly if the trunk ever becomes unavailable.

Setting	Description
Max Registration Rate	Use this field to specify the average number of REGISTER requests per time period that are allowed on this trunk. You can specify this number as a decimal or a fraction. For example: 40/5s: 40 requests/5 seconds 3.5/2m: 3.5 requests/2 minutes 2.5/1h: 2.5 requests/1 hour

Setting

Description

Max Registration Rate

Use this field to specify the average number of REGISTER requests per time period that are allowed on this trunk.

You can specify this number as a decimal or a fraction. For example:

40/5s: 40 requests/5 seconds
3.5/2m: 3.5 requests/2 minutes
2.5/1h: 2.5 requests/1 hour

SIP Access Control

The access control list that you construct here only apply to inbound calls.

Setting	Description
Use Source Address	Use this switch to determine whether ACL matching should use the SIP messaging source address (Yes) or the VIA header originating address (No). The default setting is Yes.
Allow the Following Addresses	Use the controls in this section to enter and build a list of IP or CIDR addresses to which you want to allow SIP access. Notes: Even if an address is allowed by matching this list, it can be denied if it also appears in the Always Deny list. The addresses in this list will not have any affect on HTTP provisioning.
Always Deny the Following Addresses	Use the controls n this section to enter and build a list of IP or CIDR addresses to which you want to deny SIP access. Notes: Addresses must first match the Allow list before being evaluated against this list. The addresses in this list will not have any affect on HTTP provisioning.
Allow All	Select this check box if you want to allow access to any IP or CIDR address. Note: Allowing all addresses is a security risk.

Connection Configuration

GeneralClick to expand

Setting	Description
Language	Use this drop-down to choose the language that you want to use for all calls that come in on this trunk. Note: This language can be overridden by settings in Architect.

Calls

Setting	Description
Max Calls	Use this field to specify the maximum number of combined active inbound and outbound calls that are allowed on this trunk.
Max Call Rate	Use this field to specify the average number of calls per time period that are allowed on this trunk. This rate applies to both inbound and outbound calls. You can specify this number as a decimal or a fraction. For example: 40/5s: 40 calls/5 seconds 3.5/2m: 3.5 calls/2 minutes 2.5/1h: 2.5 calls/1 hour
Max Dial Timeout	Use this field to specify the maximum number of seconds for a delay before an outgoing call attempt is aborted.

Setting

Description

Max Calls

Use this field to specify the maximum number of combined active inbound and outbound calls that are allowed on this trunk.

Max Call Rate

Use this field to specify the average number of calls per time period that are allowed on this trunk. This rate applies to both inbound and outbound calls.

You can specify this number as a decimal or a fraction. For example:

40/5s: 40 calls/5 seconds
3.5/2m: 3.5 calls/2 minutes
2.5/1h: 2.5 calls/1 hour

Max Dial Timeout

Use this field to specify the maximum number of seconds for a delay before an outgoing call attempt is aborted.

TransportClick to expand

Setting	Description
Transport DSCP Value	Use this drop-down to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP and RTCP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP and RTCP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).
Retryable Reason Codes	Use the field to enter a list of valid SIP reason codes. If an outbound call that is made on this line returns one of the SIP reason codes in this list, then that call is retried on the next line. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, Genesys Cloud automatically enters a list of the most common retryable codes in the Retryable Reason Codes field. The default code list: 500-599
Retryable Cause Codes	Use this field to enter a list of valid Q.850 cause codes. If an outbound call that is made on this line returns one of the Q.850 cause codes in this list, then that call is retried on the next line. You can specify individual reason codes or ranges of reason codes, separated by commas. By default, Genesys Cloud automatically enters a list of the most common retryable codes in the Retryable Cause Codes field. The default code list: 1-5,25,27,28,31,34,38,41,42,44,46,62,63,79,91,96,97,99,100,103

Setting

Description

Transport DSCP Value

Use this drop-down to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP and RTCP packets.

The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP and RTCP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).

Retryable Reason Codes

Use the field to enter a list of valid SIP reason codes. If an outbound call that is made on this line returns one of the SIP reason codes in this list, then that call is retried on the next line.

You can specify individual reason codes or ranges of reason codes, separated by commas.

By default, Genesys Cloud automatically enters a list of the most common retryable codes in the Retryable Reason Codes field.

The default code list: 500-599

Retryable Cause Codes

Use this field to enter a list of valid Q.850 cause codes. If an outbound call that is made on this line returns one of the Q.850 cause codes in this list, then that call is retried on the next line.

You can specify individual reason codes or ranges of reason codes, separated by commas.

By default, Genesys Cloud automatically enters a list of the most common retryable codes in the Retryable Cause Codes field.

The default code list:

1-5,25,27,28,31,34,38,41,42,44,46,62,63,79,91,96,97,99,100,103

TCP Settings

Setting	Description
TCP Connection Timeout	Use this field to specify the number of seconds to delay before marking the TCP Connection to the remote IP address as failed and marking the port as unreachable.
TCP Connection Idle Timeout	Use this field to specify the number of seconds that a TCP connection can remain idle before being automatically closed.

TLS Settings

Setting	Description
Mutual Authentication	Use this switch to enable or disable the mutual authentication requirement when negotiating the TLS handshake. The default setting is Disabled.
SIPS URI scheme	Use this switch to enable or disable the sending of a SIPS URI scheme when it is configured for the TLS transport protocol. The default setting is Disabled.
Method	Use this drop-down to choose which SSL 0r TLS method version to use. Available choices are: TLS v1 TLS v1.1 TLS v1.2 TLS v1.3 The default setting is TLS v1.2.
Far-End NAT Traversal	Use this switch to enable or disable the Far-End NAT Traversal features for the remote user agent. For example, you would use this with SRTP Symmetric Latching of media streams. The default setting is Disabled. For more information, see FENT overview.
Ciphers	Use this drop-down to select and build a preferred order list of TLS ciphers. Available choices are: TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384 TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA TLS_DHE_RSA_WITH_AES_256_CBC_SHA256 TLS_DHE_RSA_WITH_AES_256_CBC_SHA TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256 TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA TLS_DHE_RSA_WITH_AES_128_CBC_SHA256 TLS_DHE_RSA_WITH_AES_128_CBC_SHA TLS_RSA_WITH_AES_128_GCM_SHA256 TLS_RSA_WITH_AES_128_CBC_SHA256 TLS_RSA_WITH_AES_128_CBC_SHA
Subject Alternative Names	Use this field to specify and build a list of subject alternative names to use for the secure interface. Valid values must begin with one of the following prefixes: DNS: IP: URI: email:

MediaClick to expand

Setting	Description
DSCP Value	Use this drop-down to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP and RTCP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP and RTCP packet. The range of values available is 00 (0,000000) through 3F (63, 111111). The default value is 2E (46 101110) EF.
Media Method	Use this drop-down to choose the method that you want to use to offer an SDP (Session Description Protocol) to the other participant when making an outgoing call. The offer proposes the set of media streams and codecs along with the IP addresses and ports to use. There are three choices for the Media Method: Normal: Use the normal method, which sends an SDP Offer in the initial SIP INVITE request. Delayed: Use the delayed method, which waits for an SDP Offer in a response before sending an SDP Answer. Auto: Allow the system to choose between using the normal or the delayed method.
Preferred Codec List	Use the controls in this section to choose and build a preferred list of codecs. Available choices are: audio/g722 audio/g729 audio/PCMA (g711 A-Law) audio/PCMU (g711 µ-Law) audio/opus Note: PCMU and PCMA are also known as the g711 codec (the PCM stands for Pulse Code Modulation). PCMU ( µ-Law) is primarily for use in North America and PCMA (A-Law) is primarily for use in other countries outside of North America.
SRTP Cipher Suite List	Use the controls in this section to choose and build a preferred list of SRTP cipher suites to offer or allow in response. Available choices are: AES_CM_128_HMAC_SHA1_32 – Legacy AES_CM_128_HMAC_SHA1_80 – Legacy AES_CM_192_HMAC_SHA1_32 AES_CM_192_HMAC_SHA1_80 AES_CM_256_HMAC_SHA1_32 AES_CM_256_HMAC_SHA1_80
Disconnect on Idle RTP	Use this switch to enable or disable the ability to disconnect a call when RTP is not received for an extended period of time. This setting is designed to account for intentional hold/conference scenarios. Note: An extended period of time is defined as 5 minutes for normal calls or 12 hours for media that is sent in one direction (not send and receive). The default setting is Enabled.

DTMF Settings

Setting	Description
DTMF Payload	Specify the payload type value to use when the DTMF Method type is RTP Events. Valid range is 96–127. The default value is 101. Valid only when DTMF Method value is set to RTP Events.
DTMF Method	Use the drop-down to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. The default value is RTP Events. There are three choices for the DTMF Method: RTP Events: Enables out-of-band processing of events from the RTP stream (RFC 2833 or 4733). In-band Audio: Enables the processing, detection, and synthesis of events from the audio codec stream. None: Don’t use a DTMF method.

Setting

Description

DTMF Payload

Specify the payload type value to use when the DTMF Method type is RTP Events. Valid range is 96–127. The default value is 101.

Valid only when DTMF Method value is set to RTP Events.

DTMF Method

Use the drop-down to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. The default value is RTP Events.

There are three choices for the DTMF Method:

RTP Events: Enables out-of-band processing of events from the RTP stream (RFC 2833 or 4733).
In-band Audio: Enables the processing, detection, and synthesis of events from the audio codec stream.
None: Don’t use a DTMF method.

ProtocolClick to expand

Header / Invite

Setting	Description
Conversation Headers	Use this switch to enable or disable the ability to insert the custom conversation header: “x-inin-cnv” with the UUID value into SIP messages. The default setting is Disabled.

Setting

Description

Conversation Headers

Use this switch to enable or disable the ability to insert the custom conversation header: “x-inin-cnv” with the UUID value into SIP messages.

The default setting is Disabled.

AuthenticationClick to expand

Inbound Digest Authentication

Setting	Description
	Use this switch to enable or disable the ability to challenge Inbound requests. If you enable digest authentication, you’ll need to specify the associated authorization methods, realm, user name, and password. The default setting is Disabled.
Authorization Methods	Use this field to select the SIP methods that should be challenged using SIP digest authentication. Available choices are: BYE INFO INVITE NOTIFY OPTIONS REFER REGISTER SUBSCRIBE UPDATE
Realm	Use this field to specify the domain name for the SIP realm that is used to authenticate the SIP challenge response.
User Name	Use this field to specify the user name that is authorized to access this trunk when receiving credentials from an external user agent client.
Password	Use this field to specify the password associated with the username that is authorized to access this trunk when receiving credentials from an external user agent client. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.

ProvisioningClick to expand

Setting	Description
Require Authentications	Use this switch to enable or disable the requirement that an HTTP request must be authenticated by either a digest or MTLS before being processed. The default setting is Disabled.
Prefer HTTP Redirects	Use this switch to enable or disable a preference for HTTP redirects over a proxy when the configuration data is not found on the local Edge. The default setting is Disabled.
Require HTTPS	Use this switch to enable or disable the requirement that HTTPS forces an HTTP redirect if there is an available HTTPS server. The default setting is Enabled.
External Provision Server	Use this field to specify the URI to an external provision server to redirect requests for phone configuration when the phone’s configuration is not found.
HTTP Port	Use this field to specify the port on which the HTTP receiver should listen.
HTTPS Port	Use this field to specify the port on which the HTTPS receiver should listen.

DiagnosticClick to expand

Setting	Description
Media Capture	Use this switch to enable or disable media capture. The default setting is Disabled. The Media Capture setting is designed to be enabled while you are working with Genesys Cloud Product Support personnel. Enabling it will generate a HPAA Packet File Format (HPAACAP) file that contains live packet streams that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Media Capture setting as directed by Genesys Cloud Product Support. Warnings: Enabling media capture can degrade performance and affect QoS. Media capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Media Capture setting. If you are a PCI compliant Genesys Cloud organization and have the PCI DSS setting enabled, then you will not be able to enable media capture – the Media Capture setting will not be available.
Protocol Capture	Use this switch to enable or disable protocol capture. The default setting is Disabled. The Protocol Capture setting is designed to be enabled while you are working with Genesys Cloud Product Support personnel. Enabling it will generate a PCAP file that contains protocol specific network information that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Protocol Capture setting as directed by Genesys Cloud Product Support. Warnings: Enabling protocol capture can degrade performance and affect QoS. Protocol capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Protocol Capture setting. If you are a PCI compliant Genesys Cloud organization and have the PCI DSS setting enabled, then you will not be able to enable protocol capture – the Protocol Capture setting will not be available.
Capture Until	Use the calendar and clock controls to specify how long you want to collect data.

Setting

Description

Media Capture

Use this switch to enable or disable media capture.

The default setting is Disabled.

The Media Capture setting is designed to be enabled while you are working with Genesys Cloud Product Support personnel. Enabling it will generate a HPAA Packet File Format (HPAACAP) file that contains live packet streams that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Media Capture setting as directed by Genesys Cloud Product Support.

Warnings:

Enabling media capture can degrade performance and affect QoS.
Media capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Media Capture setting.
If you are a PCI compliant Genesys Cloud organization and have the PCI DSS setting enabled, then you will not be able to enable media capture – the Media Capture setting will not be available.

Protocol Capture

Use this switch to enable or disable protocol capture.

The default setting is Disabled.

The Protocol Capture setting is designed to be enabled while you are working with Genesys Cloud Product Support personnel. Enabling it will generate a PCAP file that contains protocol specific network information that can be used for diagnostic and troubleshooting purposes. Therefore, you should only enable the Protocol Capture setting as directed by Genesys Cloud Product Support.

Warnings:

Enabling protocol capture can degrade performance and affect QoS.
Protocol capture will log all data entered into the system, including data entered via Secure IVR flows. This could include sensitive data that should not be exposed or captured. As such, if your organization is using Secure IVR, you should not enable the Protocol Capture setting.
If you are a PCI compliant Genesys Cloud organization and have the PCI DSS setting enabled, then you will not be able to enable protocol capture – the Protocol Capture setting will not be available.

Capture Until

Use the calendar and clock controls to specify how long you want to collect data.

Phone System Logging (Syslog)

Setting	Description
Syslog	Use this switch to enable or disable the ability of the the Edges connected to this trunk to receive syslogs from phones The default setting is Disabled. Note: You must also enable System Logging on the Base Setting or Phone configuration settings for your phones. For more information, see Find your phone’s configuration settings.
Syslog Port	Use this field to specify the port on which Edges connected to this trunk will listen for information on receiving syslogs from phones. The default setting is port 514.

Setting

Description

Syslog

Use this switch to enable or disable the ability of the the Edges connected to this trunk to receive syslogs from phones

The default setting is Disabled.

Note: You must also enable System Logging on the Base Setting or Phone configuration settings for your phones. For more information, see .

Syslog Port

Use this field to specify the port on which Edges connected to this trunk will listen for information on receiving syslogs from phones.

The default setting is port 514.

CustomClick to expand

The Custom option is designed to allow Genesys Cloud Product Support personnel to alter a External Trunk configuration for troubleshooting or special circumstances. You should only enter custom property settings as directed by Genesys Cloud Product Support.

Setting	Description
Property Name	The name to assign to the custom property.
Data Type	The data type for the custom property. The available data types include: Boolean Text Number List
Value	The value to assign the custom property. The data allowed in the Value filed changes depending on the Data Type selected: When you select Boolean, the Value field changes to a drop down containing True and False. When you select Text, the Value field will accept any characters you enter into the filed. When you select Number, the Value field will only accept numeric characters. When you select List, the Value field will only accept data entered as a comma separated list. You can enter numbers and letters enclosed in quotes ( “a”,”b”)

Setting

Description

Property Name

The name to assign to the custom property.

Data Type

The data type for the custom property.

The available data types include:

Boolean
Text
Number
List

Value

The value to assign the custom property.

The data allowed in the Value filed changes depending on the Data Type selected:

When you select Boolean, the Value field changes to a drop down containing True and False.
When you select Text, the Value field will accept any characters you enter into the filed.
When you select Number, the Value field will only accept numeric characters.
When you select List, the Value field will only accept data entered as a comma separated list. You can enter numbers and letters enclosed in quotes ( “a”,”b”)

[NEXT] Was this article helpful?

Get user feedback about articles.

Loading...

SIP phone trunk settings

Registration

SIP Access Control

Connection Configuration

[NEXT] Was this article helpful?