1. ACLI Reference Guide
  2. ACLI Configuration Elements N-Z
  3. sip-interface

sip-interface

The sip-interface element allows you to configure a SIP interface for your Oracle Communications Session Border Controller.

Parameters

state
Enable or disable the SIP interface
  • Default: enabled
  • Values: enabled | disabled

Note:

Oracle does not recommend disabling and re-enabling a sip-interface operating with TCP ports. Depending on conditions and circumstances, you may not be able to re-enable this sip-interface without rebooting the system. If you need to disable, then re-enable a sip-interface, ensure that:
  • There are no ESTABLISHED in-bound sockets
  • The access-control-trust-level of the realm must not be configured to low or medium
realm-id
Enter the name of the realm to which the SIP interface applies
description
Provide a brief description of the sip-interface configuration element
sip-ports
Access the sip-ports subelement
carriers
Enter a list of carriers related to the sip-config. Entries in this field must follow the Carrier Format.
trans-expire
Set the transaction expiration timer in seconds
  • Default: 0
  • Values: Min: 0 | Max: 2147473
initial-inv-trans-expire
Transaction expiration time <TIMER_B> for initial INVITE.
  • Default: 0
  • Values: 0 - 2147473
invite-expire
Set the INVITE transaction expiration timer in seconds
  • Default: 0
  • Values: Min: 0 | Max: 2147473
max-redirect-contacts
Enter the maximum number of contact and route attempts in case of a redirect
  • Default: 0
  • Values: Min: 0 | Max: 10
proxy-mode
Set the default SIP request proxy mode
  • Values:
    • proxy—Forward all SIP requests to other session agents
    • redirect—Send a SIP 3xx redirect response with contacts (found in the local policy) to the previous hop
    • record-route—Forward requests with Record-Route (for stateless and transaction and operation modes only)
redirect-action
Set handling of Redirect (3xx) response messages from a session agent.
  • Default: Empty
  • Values:
    • Proxy—Send the response back to the previous hop
    • Recurse—Recurse on the contacts in the response
    • Recurse-305-only—Recurse on the contacts in the 305 response
contact-mode
Select the contact header routing mode
  • Default: none
  • Values:
    • none
    • maddr
    • strict
    • loose
nat-traversal
Select the type of HNT functionality for SIP
  • Default: none
  • Values:
    • none—NAT Traversal is disabled
    • always—Performs HNT when SIP-Via and transport addresses do not match

      Note:

      The nat-traversal parameter can establish an important media handling behavior. If you set nat-traversal on a sip-interface to always, this setting supersedes any multi-media configuration that would otherwise release the media. Instead, the SBC recognizes when a flow's leg is behind a NAT during the signaling, and ignores any configuration that would release the media. The SBC then sets up the end to end media flow in MBCD and performs its HNT function for that flow.
    • rport—Performs HNT when Via rport parameter is present and SIP-Via and transport addresses do not match
nat-interval
Enter the expiration time in seconds for the system’s cached registration entry for an endpoint doing HNT
  • Default: 30
  • Values: Min: 1 | Max: 4294967295
tcp-nat-interval
Enter the TCP NAT traversal registration interval in seconds
  • Default: 90
  • Values: Min: 0 / Max: 999999999
registration-caching
Enable or disable registration cache used for all UAs rather than those behind NATs
  • Default: disabled
  • Values: enabled | disabled
min-reg-expire
Enter the minimum registration expiration time in seconds for HNT registration caching
  • Default: 300
  • Values: Min: 0 | Max: 999999999
registration-interval
Enter the expiration time in seconds for the Oracle Communications Session Border Controller’s cached registration entry for an endpoint (non-HNT)
  • Default: 3600
  • Values: Min: 1 | Max: 999999999
route-to-registrar
Indicate whether or not the SD should forward a request addressed to the registrar to the SIP registrar as opposed to sending the request to the registered contact in the registration cache
  • Default: disabled
  • Values: enabled | disabled
secured-network
Enable or disable sending messages on unsecured transport
  • Default: disabled
  • Values: enabled | disabled
teluri-scheme
Enable or disable the conversion of SIP URIs to Tel URIs
  • Default: disabled
  • Values: enabled | disabled
uri-fqdn-domain
Change the host part of the URIs to the FQDN value set here. This applies to the Request-URI, From header, and To header in non-dialog requests sent from the SIP interface.
trust-mode
Select the trust mode for this SIP interface
  • Default: all
  • Values:
    • all—Trust all previous and next hops except untrusted session agents
    • agents-only—Trust only trusted session agents
    • realm-prefix—Trust only trusted session agents or address matching realm prefix
    • registered—Trust only trusted session agents or registered endpoints
    • None—Trust nothing
max-nat-interval
Enter the amount of time in seconds that testing should not exceed for adaptive HNT. The system will keep the expires interval at this value.
  • Default: 3600
  • Values: Min: 0 | Max: 999999999
nat-int-increment
Enter the amount of time in seconds to use as the increment in value in the SIP expires header for adaptive HNT
  • Default: 10
  • Values: Min: 0 | Max: 999999999
nat-test-increment
Enter the amount of time in seconds that will be added to the test timer for adaptive HNT
  • Default: 30
  • Values: Min: 0 | Max: 999999999
sip-dynamic-hnt
Enable or disable adaptive HNT
  • Default: disabled
  • Values: enabled | disabled
stop-recurse
Enter a list of returned response codes that this SIP interface will watch for in order to stop recursion on the target’s or contact’s messages
port-map-start
Set the starting port for the range of SIP ports available for SIP port mapping. A value of 0 disables SIP port mapping.
  • Default: 0
  • Values: Min: 1025 | Max: 65535
port-map-end
Set the ending port for the range of SIP ports available for SIP port mapping. A value of 0 disables SIP port mapping. This value must be larger than the port-map-start parameter’s value.
  • Default: 0
  • Values: Min: 1025 | Max: 65535
in-manipulationid
Enter the name of the SIP header manipulations configuration to apply to the traffic entering the Oracle Communications Session Border Controller via this SIP interface
out-manipulationid
Enter the name of the SIP header manipulations configuration to apply to the traffic exiting the Oracle Communications Session Border Controller via this SIP interface
manipulation-pattern
Number of seconds after de-registration to kill TCP connection
manipulation-string
Enter the string used in header manipulation rules for this sip-interface.
sip-ims-feature
Enable or disable IMS functionality on this SIP interface
  • Default: disabled
  • Values: enabled | disabled
subscribe-reg-event
Enables the Oracle Communications Session Border Controller to generate SIP registration events.
  • Default: disabled
  • Values: enabled | disabled
operator-identifier
Set the operator identifier value to be inserted into a P-Charging-Vector header. The direction of the call determines whether this value is inserted into the orig-ioi or the term-ioi parameter in the P-Charging-Vector header. This string value MUST begin with an alpha character.
anonymous-priority
Set the policy priority parameter for this SIP interface. It is used to facilitate emergency sessions from unregistered endpoints. This value is compared against a policy priority parameter in a local policy configuration element.
  • Default: none
  • Values:
    • none
    • normal
    • non-urgent
    • urgent
    • emergency
max-incoming-conns
Enter the maximum number of TCP/TLS connections for this sip interface
  • Default: 0 (disabled)
  • Values: Min: 0 / Max: 20000
per-scr-ip-max-incoming-conns
Enter the maximum number of TCP/TLS connections per peer IP address
  • Default: 0
  • Values: Min: 0 / Max: 20000; setting a value of 0 disables this parameter.
inactive-conn-timeout
Enter the timeout, measured in seconds for idle TCP/TLS connections
  • Default: 0
  • Values: Min: 0 / Max: 999999999; setting a value of 0 disables the timer.
untrusted-conn-timeout
Enter the timeout time, in seconds, for untrusted endpoints on TCP/TLS connections
  • Default: 0
  • Values: Min: 0 (disabled) | Max: 999999999
network-id
Set the value that will be inserted into the P-Visited-Network-ID header
ext-policy-server
Enter the name of external policy server used as the CLF for this SIP interface
default-location-string
Set a default location string to insert into P-Access-Network-Info header when the CLF does not return this value
charging-vector-mode
Set the state of P-Charging-Vector header handling
  • Default pass
  • Values:
    • none—Pass the P-Charging-Vector header received in an incoming SIP message untouched as the message is forwarded out of the Oracle Communications Session Border Controller, not extracting RADIUS information
    • pass—Pass the P-Charging-Vector header received in an incoming SIP message untouched as the message is forwarded out of the Oracle Communications Session Border Controller, extracting RADIUS information.
    • delete—Delete the P-Charging-Vector header received in an incoming SIP message before it is forwarded out of the Oracle Communications Session Border Controller
    • insert—Inserts the P-Charging-Vector header in an incoming SIP message that does not contain the P-Charging-Vector header. If the incoming message contains the P-Charging-Vector header, the Oracle Communications Session Border Controller will overwrite the P-Charging-Vector header with its values.
    • delete-and-respond—Removes the P-Charging-Vector from incoming requests for a session and store it. Then the Oracle Communications Session Border Controller inserts it into outbound responses related to that session in a P-Charging-Vector header.
    • conditional-insert—Inserts the P-Charging-Vector header in an incoming SIP message that does not contain the P-Charging-Vector header. If the incoming message contains the P-Charging-Vector header, the Oracle Communications Session Border Controller passes the P-Charging-Vector header untouched as the message is forwarded, extracting RADIUS information.

    Note:

    Note that the default setting for the charging-vector-mode is pass for new SIP interface configurations. If you are upgrading and there are pre-existing SIP interfaces in your (upgraded) configuration, the default becomes none.
charging-function-address-mode
Set the state of P-Charging-Function-Address header handling
  • Default: pass
  • Values:
    • none—Pass the P-Charging-Function-Address header received in an incoming SIP message untouched as the message is forwarded out of the Oracle Communications Session Border Controller, not extracting RADIUS information
    • pass—Pass the P-Charging-Function-Address header received in an incoming SIP message untouched as the message is forwarded out of the Oracle Communications Session Border Controller, extracting RADIUS information.
    • delete—Delete the P-Charging-Function-Address header received in an incoming SIP message before it is forwarded out of the Oracle Communications Session Border Controller
    • insert—Inserts the P-Charging-Function-Address header in an incoming SIP message that does not contain the P-Charging-Function-Address header. If the incoming message contains the P-Charging-Function-Address header, the Oracle Communications Session Border Controller will prepend its configured values to the header.
    • insert-reg-cache—To be configured on the SIP interface facing the UE, configures the Oracle Communications Session Border Controller to replace the PCFA with the most recently cached values rather than the ccf-address you set to be static in your configuration. The cached values come from one of the following that the Oracle Communications Session Border Controller has received most recently: request, response, registration, or local configuration.
    • delete-and-respond—To be configured on the SIP interface facing the S-CPCF, configures the Oracle Communications Session Border Controller to strip out the latest cached PCFA.
    • conditional-insert—Inserts the P-Charging-Function-Address header in an incoming SIP message that does not contain the P-Charging-Vector header. If the incoming message contains the P-Charging-Function-Address header, the Oracle Communications Session Border Controller passes the P-Charging-Function-Address header untouched as the message is forwarded, extracting RADIUS information.

    Note:

    Note that the default setting for the charging-function-address-mode is pass for new SIP interface configurations. If you are upgrading and there are pre-existing SIP interfaces in your (upgraded) configuration, the default becomes none.
ccf-address
Set the CCF address value that will be inserted into the P-Charging-Function-Address header
ecf-address
Set the ECF address value that will be inserted into the P-Charging-Function-Address header
term-tgrp-mode
Select the mode for routing for terminating trunk group URIs
  • Default: none
  • Values:
    • none—Disable routing based on trunk groups
    • iptel—Use trunk group URI routing based on the IPTEL formats
    • egress-uri—Use trunk group URI routing based on the egress URI format
implicit-service-route
Enable or disable the implicit service route behavior
  • Default: disabled
  • Values:
    • enabled
    • disabled
    • strict
rfc2833-payload
Enter the payload type used by the SIP interface in preferred rfc2833-mode
  • Default: 101
  • Values: Min: 96 | Max: 127
rfc2833-mode
Choose whether the SIP interface will behave exactly the same way as before and the 2833or UII negotiation will be transparent to the Oracle Communications Session Border Controller, transparent, or whether the sip-interface prefers to use 2833 for DTMF transfer and would signal that in its SDP, preferred. However the final decision depends on the remote endpoint.
  • Default: transparent
  • Values: transparent | preferred | dual
constraint-name
Enter the name of the constraint being applied to this interface
response-map
Enter the name of the response map being applied to this interface
local-response-map
Enter the name of the local response map being applied to this interface
sec-agree-feature
Determines if sec-agree feature is enabled.
  • Default disabled
  • Values enabled | disabled
sec-agree-pref
Determines the security protocol preferences used with Sec-agree support
  • Default: ipsec3gpp
  • Values:
    • ipsec3gpp — support only IMS-AKA protocol
    • tls — support only TLS protocol
    • ipsec3gpp-tls — support both IMS-AKA and TLS, preferred protocol is IMS-AKA
    • tls-ipsec3gpp — support both TLS and IMS-AKA, preferred protocol is TLS
ims-aka-feature
This parameter is unsupported.
enforcement-profile
Enter the name of the enforcement profile associated with this SIP interface
route-unauthorized-calls
Enter the name of the SA or SAG you want to route unauthorized calls
tcp-keepalive
Enable or disable standard keepalive probes to determine whether or not connectivity with a remote peer is lost.
  • Default: none
  • Values: none | enabled | disabled
add-sdp-invite
Enable or disable this SIP interface inserting an SDP into either an INVITE or a REINVITE
  • Default: disabled
  • Values:
    • disabled—Do not insert an SDP
    • invite—Insert an SDP in the invite
    • reinvite—Insert an SDP in the reinvite
add-sdp-profile
Enter a list of one or more media profile configurations you want to use when the Oracle Communications Session Border Controller inserts SDP into incoming INVITEs that have no SDP. The media profile contains media information the Oracle Communications Session Border Controller inserts in outgoing INVITE.
add-sdp-in-msg
Identifies the messages in which to insert SDP offers or answers. The only allowable value is 18xresp. The default is null (no value).
  • Default: null
  • Values:
    • 18xresp—For an offerless INVITE that needs preconditions, causes the Oracle Communications Session Border Controller to insert the SDP, as configured in the media profile names listed in add-sdp-profiles-in-msg, in the 18x (183) response towards the UE.
add-sdp-profile-in-msg
Identifies a list of media profiles that contain, based on the codec, the SDP to insert in the 18x response when add-sdp-in-msg is configured.
sip-profile
Enter the name of the sip-profile to apply to this interface.
sip-isup-profile
Enter the name of the sip-isup-profile to apply to this interface.
tcp-conn-dereg
Number of seconds after de-registration to kill TCP connection.
  • Default 0 (disabled)
tunnel-name
Tunnel traffic for load balancer. Traffic sent to/from this interface will be encapsulated in an RFC 2003 compliant tunnel to/from the load balancer usign the associated network-interface's tunnel name.
register-keep-alive
Sets the use of RFC 5626 CRLF Keepalives on this sip interface.
  • Default: none
  • Values:
    • none—disables this feature
    • always— Keepalive always added to SIP-Via
    • bnat— Keepalive added to SIP-Via when SIP-via and transport addresses do not match (indicates endpoint is behind a NAT)
kpml-interworking
Enables or disables the KPML to RFC2833 interwokring feature.
  • Default: disabled
  • Values: enabled | disabled
kpmlRFC2833-iwf-on-hairpin
When enabled, specifies that the system supports KPML to RFC2833 interworking for hairpinned calls. This requires that kpml-interworking to also be enabled.
  • Default: disabled
  • Values: enabled | disabled —When enabled, allows the Oracle Communications Session Border Controller to present the correct digit encapsulation (KPML or RFC2833) when hairpinned back to the original interface.
msrp-delay-egress-bye
Delay egress BYE message.
  • Default: disabled
  • Values: enabled | disabled
send-380-response
The phrase entered in this parameter is inserted into the <reason> element in the <alternative-service> element in the XML body in the 380 response returned to an endpoint when the call cannot be completed. This is in compliance with GSMA's Voice over LTE specification (IR. 92).
pcscf-restoration
Configure a reason phrase, enclosed in quotes, that will be included in the P-CSCF restoration response, the reason field of a 504 response sent back to the UE.
session-timer-profile
A session-timer-profile name is configured here to apply that session timer profile to this SIP interface.
session-recording-server
A maximum of four names of session-recording-servers, or session-recording-groups, or a combination of both exisiting in the realm associated with the session reporting client. Valid values are alpha-numeric characters. session recording groups are indicated by prepending the groupname with SRG:
session-recording-required
Determines whether calls are accepted by the SBC if recording is not available.
  • Default: disabled
  • Values:
    • enabled—Restricts call sessions from being initiated when a recording server is not available.
    • disabled—Allows call sessions to initiate even if the recording server is not available.
service-tag
Service tag
p-early-media-header
Used to enable P-Early-Media SIP header support.
  • Default: Disabled
  • Values:
    • disabled—(the default value) disables support
    • add—enables support and allows the SBC/P-CSCF to add the P-Early-Media header to SIP messages.
    • modify—enables support and allows the SBC/P-CSCF to modify or strip the P-Early-Media header in SIP messages.
    • support—adds additional PEM support, including enforcing PEM from trusted sources, preventing system modification of PEM direction, not adding PEM if absent from SIP replies and adding PEM if it is not advertised in the initial INVITE.
p-early-media-direction
Used to specify the supported directionalities. for P-Early-Media header support.
  • sendrecv—send and accept early media
  • sendonly—send early media
  • recvonly—receive early media
  • inactive—reject/cancel early media
options
Enter optional features and/or parameters
spl-options
Enter any optional features or parameters
.
diversion-info-mapping-mode
Configure this parameter to specify how the Diversion and History-Info headers map to and interwork on the interface.
  • Default none
  • Values:
    • none—no conversion applied
    • div2hist—any Diversion headers in the initial INVITEs going out of this SIP interface will be converted to History-iInfo headers before sending
    • force—behavior is the same as div2hist when a Diversion header is present in the incoming INVITE if there are no Diversion headers, a History-Info header for the current URI is added in the outgoing INVITE
    • hist2div—any History-Info headers in the initial INVITEs going out of this sip interface will be converted to Diversion headers before sending
atcf-icsi-match
Matches the icsi value for atcf call.
asymmetric-preconditions
Identifies whether to enable preconditions interworking on the interface. Allowable values are enabled and disabled. The default is disabled. You cannot enable asymmetric preconditions unless you have first set the value of sip-interface > options to 100rel-interworking.
  • Default: disabled
  • Values:
    • enabled—Enables preconditions interworking on the interface.
    • disabled—Disables preconditions interworking on the interface.
asymmetric-preconditions-mode
Identifies, when the value of asymmetric-preconditions is enabled, whether to send egress INVITEs immediately or to delay them until preconditions have been met. Allowable values are send-with-delay and send-with-nodelay.
  • Default: send-with-nodelay
  • Values:
    • send-with-delay—Delays INVITEs on the egress interface until preconditions are met on the ingress interface.
    • send-with-nodelay—Forwards INVITEs to the egress interface immediately, but holds the responses until preconditions are met on the ingress interface.
sm-icsi-match-for-invite
The ICSI URN to match on to increment the session-based messaging counters. For example - urn:urn-7:3gpp-service.ims.icsi.oma.cpm.largemsg
  • Default: empty
sm-icsi-match-for-message
The ICSI URN to match on to increment the event-based messaging counters.
  • Default: urn:rrn-7:3gpp-service.ims.icsi.oma.cpm.largemsg
s8hr-profile
Enter the name of the S8HR profile to apply to this SIP interface
playback-file
Specifies the name of the media file, stored previously in /code/media, that the system plays when triggered for this sip-interface.
playback-trigger
Specifies when the system triggers the local media playback function.
  • Default: disabled
  • 180-force—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 180 message.
  • 180-no-sdp—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 180 message that does not include SDP.
npli-profile
Enter the name of the NPLI profile to apply to this SIP interface
hist-to-div-for-cause-380
Determines whether to interwork cause 380 messages within history-Info and Diversion header interworking.
  • Default: inherit
  • Values:
    • inherit—uses the setting specified in the sip-config.
    • enabled—enables the message interworking.
    • disabled—disables the message interworking.
user-agent
Reserved for use with Microsoft Teams integrations only.
ringback-file
Specifies the name of the media file, stored previously in /code/media, that the system plays when triggered for this realm.
ringback-trigger
Specifies when the system triggers the local media playback function.
  • Default: none
  • Values:
    • none—The system does not perform local media playback procedures. Based on precedence, however, the system may issue playback based on other element configurations. Local media playback follows the precedence session-agent, realm, then sip-interface.
    • disabled—The system does not perform media playback procedures on this flow, regardless of ensuing configurations.
    • 180-no-sdp—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 180 message that does not include SDP.
    • 180-force—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 180 message.
    • 183—Starts playback to caller when 183 is sent to call originator. The system stops the playback on the final response (either 2xx success or 4xx error). Configure this 183 value on the original INVITE ingress realm/sip-interface/session-agent.
    • refer—Starts playback to the referee when it receives a REFER. This trigger operates only if the OCSBC actually terminates and performs the refer operation. If the REFER is via proxy, playback is not a triggered. Playback stops when the refer operation is complete with a final response (200-299 or 400-699). Configure this refer value on the ingress realm/sip-interface/session-agent of the transferred call.
    • 183-and-refer—Starts playback when both 183 and refer triggers are activated.183-no-sdp—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 183 message that does not include SDP.
    • 183-no-sdp—Defines the trigger by which the system starts local media playback to caller. This parameter causes playback trigger whenever the called leg responds with a 183 message that does not include SDP.
    • playback-on-header—Starts or stops playback based on the presence of the P-Acme-Playback header and its definitions.
sti-as
Specifies the name of an sti-server-group name or a space-separated list of sti-server (up to four allowed) to which the SBC shall send AS requests. When configuring a group name, use the prefix stg: followed by your group name. For example, stg:myStiGroupName.
sti-vs
Specifies the name of an sti-server-group name or a space-separated list of sti-server (up to four allowed) to which the SBC shall send VS requests. When configuring a group name, use the prefix stg: followed by your group name. For example, stg:myStiGroupName.
sti-orig-id
Specifies the UUID v4 to be added to STI-AS requests, if not already present, during STIR/SHAKEN functions.
sti-attest
Specifies the attestation value that is sent in AS request, during STIR/SHAKEN functions. The default is empty
  • full-attestation
  • partial-attestation
  • gateway-attestation
sti-signaling-attest
Enable this parameter to instruct the SBC to use attestation level and origination ID headers from the ingress SIP INVITE in the REST query to the STI-AS, if preferred. When enabled, the Attestation-Info and Origination-ID headers override the configured values, if present. If one of the two requested headers is present, the other value is obtained from configured parameters.
  • Default: Disable—The system does not use the attestation value and origId from SIP headers.
  • Enable—The system uses the attestation value and origId from SIP headers, when present.
allow-diff2833-clock-rate-mode
Specifies whether and how the SBC can present an SDP answer towards ingress that contains a telephone-event clock rate that is not the same as the audio codec clock rate. When this parameter is disabled, the SBC does not send telephone-event with a different clock rate than audio codec as an answer towards the ingress.
  • Default: Disabled
  • Values:
    • use-2833-clock-rate—Allow the use of different clock rates and generate RFC2833 packets using the telephone-event clock rate.
    • use-codec-clock-rate—Allow the use of different clock rates and generate RFC2833 packets using the codec clock rate.
fax-continue-session
Retains an ongoing fax call when a refresh or non-refresh REINVITE is received.
  • Default : None
  • Values:
    • None - If both ingress and egress realms have none value then the fax session/call can get interrupted and terminated on receiving a REINVITE. If one realm is configured with none then the REINVITE can get rejected or the call can switch from fax to voice depending upon the conditions.
    • faxToVoice - If the Oracle Communications Session Border Controller receives a non-refresh REINVITE with non-faxable codec from the UE that initiated the fax call, then the call switches from fax to voice and REINVITE forwards to the other end.
    • faxToVoiceNotAllowed - If the Oracle Communications Session Border Controller receives a non-refresh REINVITE with non-faxable codec from the UE that initiated the fax call, then Oracle Communications Session Border Controller rejects the REINVITE and the fax call/session continues.

Path

sip-interface is an element under the session-router path. The full path from the topmost ACLI prompt is: configure terminal , and then session-router , and then sip-interface.

Note:

This is a multiple instance configuration element.