- Cisco Unified Border Element SIP Support
- SIP Core SIP Technology Enhancements
- Reporting End-of-Call Statistics in SIP BYE Message
- Configurable Hostname in Locally Generated SIP Headers
- SIP-to-SIP Basic Functionality for Session Border Controller
- SIP Session Timer Support
- SIP-to-SIP Supplementary Services for Session Border Controller
- Session Refresh with Reinvites
- Cisco UBE Out-of-dialog OPTIONS Ping
- SIP Out-of-Dialog OPTIONS Ping Group
- Configuring an Error Response Code upon an Out-of-Dialog OPTIONS Ping Failure
- Configurable SIP Error Codes
- Configuring SIP Error Message Pass Through
- SIP INFO Method for DTMF Tone Generation
- SIP Enhanced 180 Provisional Response Handling
- Configuring SIP 181 Call is Being Forwarded Message
- SIP UPDATE Message per RFC 3311
- Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Configurable Pass-Through of SIP INVITE Parameters
- Transparent Tunneling of QSIG and Q.931
- SIP Diversion Header Enhancements
- SIP History INFO
- SIP Ability to Send a SIP Registration Message on a Border Element
- Expires Timer Reset on Receiving or Sending SIP 183 Message
- Dynamic Refer Handling
- Configurable SIP Parameters via DHCP
- Multiple Registrars on SIP Trunks
- Handle Multiple Early Dialog Messages
- Monitoring of Phantom Packets
- V150.1 MER Support in SDP Passthrough Mode
- Feature Information for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Prerequisites for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Restrictions for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Configuring P-Header and Random-Contact Support on the Cisco Unified Border Element
- Configuring P-Header Translation on a Cisco Unified Border Element
- Configuring P-Header Translation on an Individual Dial Peer
- Configuring P-Called-Party-Id Support on a Cisco Unified Border Element
- Configuring P-Called-Party-Id Support on an Individual Dial Peer
- Configuring Privacy Support on a Cisco Unified Border Element
- Configuring Privacy Support on an Individual Dial Peer
- Configuring Random-Contact Support on a Cisco Unified Border Element
- Configuring Random-Contact Support for an Individual Dial Peer
Support for PAID PPID Privacy
PCPID and PAURI Headers on the Cisco Unified Border Element
The figure below shows a typical network topology where the Cisco Unified Border Element is configured to route messages between a call manager system (such as the Cisco Unified Call Manager) and a Next Generation Network (NGN).
Devices that connect to an NGN must comply with the User-Network Interface (UNI) specification. The Cisco Unified Border Element supports the NGN UNI specification and can be configured to interconnect NGN with other call manager systems, such us the Cisco Unified Call Manager.
The Cisco Unified Border Element supports the following:
-
the use of P-Preferred Identity (PPID), P-Asserted Identity (PAID), Privacy, P-Called Party Identity (PCPID), in INVITE messages
-
the translation of PAID headers to PPID headers and vice versa
-
the translation of RPID headers to PAID or PPID headers and vice versa
-
the configuration and/or pass through of privacy header values
-
the use of the PCPID header to route INVITE messages
-
the use of multiple PAURI headers in the response messages (200 OK) it receives to REGISTER messages
P-Preferred Identity and P-Asserted Identity Headers
NGN servers use the PPID header to identify the preferred number that the caller wants to use. The PPID is part of INVITE messages sent to the NGN. When the NGN receives the PPID, it authorizes the value, generates a PAID based on the preferred number, and inserts it into the outgoing INVITE message towards the called party.
However, some call manager systems, such as Cisco Unified Call Manager 5.0, use the Remote-Party Identity (RPID) value to send calling party information. Therefore, the Cisco Unified Border Element must support building the PPID value for an outgoing INVITE message to the NGN, using the RPID value or the From: value received in the incoming INVITE message. Similarly, CUBE supports building the RPID and/or From: header values for an outgoing INVITE message to the call manager, using the PAID value received in the incoming INVITE message from the NGN.
In non-NGN systems, the Cisco Unified Border Element can be configured to translate between PPID and PAID values, and between From: or RPID values and PAID/PPID values, at global and dial-peer levels.
In configurations where all relevant servers support the PPID or PAID headers, the Cisco Unified Border Element can be configured to transparently pass the header.
Note | If the NGN sets the From: value to anonymous, the PAID is the only value that identifies the caller. |
The table below describes the types of INVITE message header translations supported by the Cisco Unified Border Element. It also includes information on the configuration commands to use to configure P-header translations.
The table below shows the P-header translation configuration settings only. In addition to configuring these settings, you must configure other system settings (such as the session protocol).
The CUBE can be configured to transparently pass the PAID and PPID headers in the incoming and outgoing Session Initiation Protocol (SIP) requests or response messages from end-to-end.
Note | The priority of P-headers are in the following order: PAID, PPID, and RPID. |
Incoming Header |
Outgoing Header |
Configuration Notes |
||
---|---|---|---|---|
PAID |
PPID |
To enable the translation to PPID headers in the outgoing header at a global level, use the asserted-id ppi command in voice service VoIP SIP configuration mode. For example: Router(conf-serv-sip)# asserted-id ppi To enable the translation to PPID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id ppi command in dial peer voice configuration mode. For example: Router(config-dial-peer)# voice-class sip asserted-id ppi |
||
RPID |
PPID |
To enable the translation to PPID headers in the outgoing header at a global level, use the asserted-id ppi command in voice service VoIP SIP configuration mode. For example: Router(conf-serv-sip)# asserted-id ppi To enable the translation to PPID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id ppi command in dial peer voice configuration mode. For example: Router(config-dial-peer)# voice-class sip asserted-id ppi |
||
PPID |
PPID |
To enable the translation to PPID headers in the outgoing header at a global level, use the asserted-id ppi command in voice service VoIP SIP configuration mode. To enable the translation to PPID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id ppi command in dial peer voice configuration mode. For example: Router(config-dial-peer)# voice-class sip asserted-id ppi |
||
PAID |
PAID |
To enable the translation to PAID headers in the outgoing header at a global level, use the asserted-id pai command in voice service VoIP SIP configuration mode. To enable the translation to PAID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id pai command in dial peer voice configuration mode. For example: Router(config-dial-peer)# voice-class sip asserted-id pai |
||
RPID |
PAID |
To enable the translation to PAID headers in the outgoing header at a global level, use the asserted-id pai command in voice service VoIP SIP configuration mode. To enable the translation to PAID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id pai command in dial peer voice configuration mode. |
||
PPID |
PAID |
To enable the translation to PAID headers in the outgoing header at a global level, use the asserted-id pai command in voice service VoIP SIP configuration mode. To enable the translation to PAID headers in the outgoing header on a specific dial peer, use the voice-class sip asserted-id pai command in dial peer voice configuration mode. |
||
PAID |
RPID |
|
||
RPID |
RPID |
|
||
PPID |
RPID |
To enable the translation to RPID headers in the outgoing header, use the remote-party-id command in SIP user-agent configuration mode. For example: Router(config-sip-ua)# remote-party-id |
||
FROM |
FROM |
No configuration required except for the remote-party-id header. | ||
FROM |
RPID |
To enable the translation to RPID headers in the outgoing header, use the remote-party-id command in SIP user-agent configuration mode. For example: Router(config-sip-ua)# remote-party-id |
||
PAID |
PAID |
Enables PPID headers on the incoming dial-peer and PAID headers on the outgoing dial-peer. |
||
RPID |
PAID |
Enables PPID headers on incoming dial-peer and PAID headers on outgoing dial-peer. |
||
PPID |
PAID |
Enables PPID headers on incoming dial-peer and PAID headers on outgoing dial-peer. |
||
PAID |
PAID |
Enables RPID headers on incoming dial-peer and PAID headers on outgoing dial-peer. |
||
RPID |
PAID |
Enables RPID headers on incoming dial-peer and PAID headers on outgoing dial-peer. |
||
PPID |
PAID |
Enables RPID headers on incoming dial-peer and PAID headers on outgoing dial-peer. |
||
PAID |
PPID |
Enables PAID headers on incoming dial-peer and PPID headers on outgoing dial-peer. |
||
RPID |
PPID |
Enables PAID headers on incoming dial-peer and PPID headers on outgoing dial-peer. |
||
PPID |
PPID |
Enables PAID headers on incoming dial-peer and PPID on outgoing dial-peer. |
||
PAID |
PPID |
Enables RPID headers on incoming dial-peer and PPID headers on outgoing dial-peer. |
||
RPID |
PPID |
Enables RPID headers on incoming dial-peer and PPID headers on outgoing dial-peer. |
||
PPID |
PPID |
Enables RPID headers on incoming dial-peer and PPID headers on outgoing dial-peer. |
||
PAID |
RPID |
Enables PPID headers on incoming dial-peer and RPID headers on outgoing dial-peer.
|
||
RPID |
RPID |
Enables PPID headers on incoming dial-peer and RPID headers on outgoing dial-peer. |
||
PPID |
RPID |
Enables PPID headers on incoming dial-peer and RPID headers on outgoing dial-peer.
|
||
PAID |
RPID |
Enables PAID headers on incoming dial-peer and RPID headers on outgoing dial-peer.
|
||
RPID |
RPID |
Enables PAID headers on incoming dial-peer and RPID headers on outgoing dial-peer. |
||
PPID |
RPID |
Enables PAID headers on incoming dial-peer and RPID headers on outgoing dial-peer.
|
Privacy
If the user is subscribed to a privacy service, the Cisco Unified Border Element can support privacy using one of the following methods:
The NGN dial plan can specify prefixes to enable privacy settings. For example, the dial plan may specify that if the caller dials a prefix of 184, the calling number is not sent to the called party.
The dial plan may also specify that the caller can choose to send the calling number to the called party by dialing a prefix of 186. Here, the Cisco Unified Border Element transparently passes the prefix as part of the called number in the INVITE message.
The actual prefixes for the network are specified in the dial plan for the NGN, and can vary from one NGN to another.
If the Privacy header is set to None, the calling number is delivered to the called party. If the Privacy header is set to a Privacy:id value, the calling number is not delivered to the called party.
If the incoming INVITE has a Privacy header or a RPID with privacy on, the outgoing INVITE can be set to Privacy: id. This behavior is enabled by configuring privacy pstn command globally or voice-class sip privacy pstn command on the selected dial-per.
Incoming INVITE can have multiple privacy header values, id, user, session, and so on. Configure the privacy-policy passthru command globally or voice-class sip privacy-policy passthru command to transparently pass across these multiple privacy header values.
Some NGN servers require a Privacy header to be sent even though privacy is not required. In this case the Privacy header must be set to none. The Cisco Unified Border Element can add a privacy header with the value None while forwarding the outgoing INVITE to NGN. Configure the privacy-policy send-always globally or voice-class sip privacy-policy send-always command in dial-peer to enable this behavior.
If the user is not subscribed to a privacy service, the Cisco Unified Border Element can be configured with no Privacy settings.
P-Called Party Identity
The Cisco Unified Border Element can be configured to use the PCPID header in an incoming INVITE message to route the call, and to use the PCPID value to set the To: value of outgoing INVITE messages.
The PCPID header is part of the INVITE messages sent by the NGN, and is used by Third Generation Partnership Project (3GPP) networks. The Cisco Unified Border Element uses the PCPID from incoming INVITE messages (from the NGN) to route calls to the Cisco Unified Call Manager.
Note | The PCPID header supports the use of E.164 numbers only. |
P-Associated URI
The Cisco Unified Border Element supports the use of PAURI headers sent as part of the registration process. After the Cisco Unified Border Element sends REGISTER messages using the configured E.164 number, it receives a 200 OK message with one or more PAURIs. The number in the first PAURI (if present) must match the contract number. The Cisco Unified Border Element supports a maximum of six PAURIs for each registration.
Note | The Cisco Unified Border Element performs the validation process only when a PAURI is present in the 200 OK response. |
The registration validation process works as follows:
-
The Cisco Unified Border Element receives a REGISTER response message that includes PAURI headers that include the contract number and up to five secondary numbers.
-
The Cisco Unified Border Element validates the contract number against the E.164 number that it is registering: - If the values match, the Cisco Unified Border Element completes the registration process and stores the PAURI value. This allows administration tools to view or retrieve the PAURI if needed.
- If the values do not match, the Cisco Unified Border Element unregisters and then reregisters the contract number. The Cisco Unified Border Element performs this step until the values match.
Random Contact Support
The Cisco Unified Border Element can use random-contact information in REGISTER and INVITE messages so that user information is not revealed in the contact header.
To provide random contact support, the Cisco Unified Border Element performs SIP registration based on the random-contact value. The Cisco Unified Border Element then populates outgoing INVITE requests with the random-contact value and validates the association between the called number and the random value in the Request-URI of the incoming INVITE. The Cisco Unified Border Element routes calls based on the PCPID, instead of the Request-URI which contains the random value used in contact header of the REGISTER message.
The default contact header in REGISTER messages is the calling number. The Cisco Unified Border Element can generate a string of 32 random alphanumeric characters to replace the calling number in the REGISTER contact header. A different random character string is generated for each pilot or contract number being registered. All subsequent registration requests will use the same random character string.
The Cisco Unified Border Element uses the random character string in the contact header for INVITE messages that it forwards to the NGN. The NGN sends INVITE messages to the Cisco Unified Border Element with random-contact information in the Request URI. For example: INVITE sip:FefhH3zIHe9i8ImcGjDD1PEc5XfFy51G@10.12.1.46:5060.
The Cisco Unified Border Element will not use the To: value of the incoming INVITE message to route the call because it might not identify the correct user agent if supplementary services are invoked. Therefore, the Cisco Unified Border Element must use the PCPID to route the call to the Cisco Unified Call Manager. You can configure routing based on the PCPID at global and dial-peer levels.
- Feature Information for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Prerequisites for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Restrictions for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
- Configuring P-Header and Random-Contact Support on the Cisco Unified Border Element
Feature Information for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Feature Name |
Releases |
Feature Information |
---|---|---|
PAID and PPID Headers in mid-call re-INVITE and UPDATE request and responses on Cisco Unified Border Element |
Cisco IOS 15.5(3)M Cisco IOS XE 3.16S |
This feature enables CUBE platforms to support:
|
Feature History Table entry for the Cisco Unified Border Element and Cisco Unified Border Element (Enterprise).
Feature Name |
Releases |
Feature Information |
---|---|---|
PAID, PPID, Privacy, PCPID, and PAURI Headers on the Cisco Unified Border Element |
12.4(22)YB 15.0(1)M Cisco IOS XE Release 3.1S |
This feature enables CUBE platforms to support:
The following commands were introduced: call-route p-called-party-id, privacy-policy, random-contact, random-request-uri validate, voice-class sip call-route p-called-party-id, voice-class sip privacy-policy, voice-class sip random-contact, and voice-class sip random-request-uri validate. |
Prerequisites for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
Cisco Unified Border Element
Cisco Unified Border Element (Enterprise)
Restrictions for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element
-
To enable random-contact support, you must configure the Cisco Unified Border Element to support SIP registration with random-contact information. In addition, you must configure random-contact support in VoIP voice-service configuration mode or on the dial peer.
-
If random-contact support is configured for SIP registration only, the system generates the random-contact information, includes it in the SIP REGISTER message, but does not include it in the SIP INVITE message.
-
If random-contact support is configured in VoIP voice-service configuration mode or on the dial peer only, no random contact is sent in either the SIP REGISTER or INVITE message.
-
Passing of “+" is not supported with PAID PPID Privacy PCPID and PAURI Headers.
Configuring P-Header and Random-Contact Support on the Cisco Unified Border Element
To enable random contact support you must configure the Cisco Unified Border Element to support Session Initiation Protocol (SIP) registration with random-contact information, as described in this section.
To enable the Cisco Unified Border Element to use the PCPID header in an incoming INVITE message to route the call, and to use the PCPID value to set the To: value of outgoing INVITE messages, you must configure P-Header support as described in this section.
- Configuring P-Header Translation on a Cisco Unified Border Element
- Configuring P-Header Translation on an Individual Dial Peer
- Configuring P-Called-Party-Id Support on a Cisco Unified Border Element
- Configuring P-Called-Party-Id Support on an Individual Dial Peer
- Configuring Privacy Support on a Cisco Unified Border Element
- Configuring Privacy Support on an Individual Dial Peer
- Configuring Random-Contact Support on a Cisco Unified Border Element
- Configuring Random-Contact Support for an Individual Dial Peer
Configuring P-Header Translation on a Cisco Unified Border Element
To configure P-Header translations on a Cisco Unified Border Element, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
asserted-id
header-type
6.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
voice
service
voip
Example: Router(config)# voice service voip |
Enters VoIP voice-service configuration mode. |
Step 4 |
sip
Example: Router(conf-voi-serv)# sip |
Enters voice service VoIP SIP configuration mode. |
Step 5 |
asserted-id
header-type
Example: Router(conf-serv-sip)# asserted-id ppi |
Specifies the type of privacy header in the outgoing SIP requests and response messages. |
Step 6 |
exit
Example: Router(conf-serv-sip)# exit |
Exits the current mode. |
Configuring P-Header Translation on an Individual Dial Peer
To configure P-Header translation on an individual dial peer, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
dial-peer
voice
tag
voip
4.
voice-class
sip
asserted-id
header-type
5.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
dial-peer
voice
tag
voip
Example: Router(config)# dial-peer voice 2611 voip |
Defines the dial peer, specifies the method of voice encapsulation, and enters dial peer voice configuration mode. |
Step 4 |
voice-class
sip
asserted-id
header-type
Example: Router(config-dial-peer)# voice-class sip asserted-id ppi |
Specifies the type of privacy header in the outgoing SIP requests and response messages, on this dial peer. |
Step 5 |
exit
Example: Router(config-dial-peer)# exit |
Exits the current mode. |
Configuring P-Called-Party-Id Support on a Cisco Unified Border Element
To configure P-Called-Party-Id support on a Cisco Unified Border Element, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
call-route
p-called-party-id
6.
random-request-uri
validate
7.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
voice
service
voip
Example: Router(config)# voice service voip |
Enters VoIP voice-service configuration mode. |
Step 4 |
sip
Example: Router(conf-voi-serv)# sip |
Enters voice service VoIP SIP configuration mode. |
Step 5 |
call-route
p-called-party-id
Example: Router(conf-serv-sip)# call-route p-called-party-id |
Enables the routing of calls based on the PCPID header. |
Step 6 |
random-request-uri
validate
Example: Router(conf-serv-sip)# random-request-uri validate |
Enables the validation of the random string in the Request URI of the incoming INVITE message. |
Step 7 |
exit
Example: Router(conf-serv-sip)# exit |
Exits the current mode. |
Configuring P-Called-Party-Id Support on an Individual Dial Peer
To configure P-Called-Party-Id support on an individual dial peer, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
dial-peer
voice
tag
voip
4.
voice-class
sip
call-route
p-called-party-id
5.
voice-class
sip
random-request-uri
validate
6.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
dial-peer
voice
tag
voip
Example: Router(config)# dial-peer voice 2611 voip |
Defines the dial peer, specifies the method of voice encapsulation, and enters dial peer voice configuration mode. |
Step 4 |
voice-class
sip
call-route
p-called-party-id
Example: Router(config-dial-peer)# voice-class sip call-route p-called-party-id |
Enables the routing of calls based on the PCPID header on this dial peer. |
Step 5 |
voice-class
sip
random-request-uri
validate
Example: Router(config-dial-peer)# voice-class sip random-request-uri validate |
Enables the validation of the random string in the Request URI of the incoming INVITE message on this dial peer. |
Step 6 |
exit
Example: Router(config-dial-peer)# exit |
Exits the current mode. |
Configuring Privacy Support on a Cisco Unified Border Element
To configure privacy support on a Cisco Unified Border Element, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
voice
service
voip
4.
sip
5.
privacy
privacy-option
6.
privacy-policy
privacy-policy-option
7.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
voice
service
voip
Example: Router(config)# voice service voip |
Enters VoIP voice-service configuration mode. |
Step 4 |
sip
Example: Router(conf-voi-serv)# sip |
Enters voice service VoIP SIP configuration mode. |
Step 5 |
privacy
privacy-option
Example: Router(conf-serv-sip)# privacy id |
Enables the privacy settings for the header. |
Step 6 |
privacy-policy
privacy-policy-option
Example: Router(conf-serv-sip)# privacy-policy passthru |
Specifies the privacy policy to use when passing the privacy header from one SIP leg to the next. |
Step 7 |
exit
Example: Router(conf-serv-sip)# exit |
Exits the current mode. |
Configuring Privacy Support on an Individual Dial Peer
To configure privacy support on an individual dial peer, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
dial-peer
voice
tag
voip
4.
voice-class
sip
privacy
privacy-option
5.
voice-class
sip
privacy-policy
privacy-policy-option
6.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
dial-peer
voice
tag
voip
Example: Router(config)# dial-peer voice 2611 voip |
Defines the dial peer, specifies the method of voice encapsulation, and enters dial peer voice configuration mode. |
Step 4 |
voice-class
sip
privacy
privacy-option
Example: Router(config-dial-peer)# voice-class sip privacy id |
Enables the privacy settings for the header on this dial peer. |
Step 5 |
voice-class
sip
privacy-policy
privacy-policy-option
Example: Router(config-dial-peer)# voice-class sip privacy-policy passthru |
Specifies the privacy policy to use when passing the privacy header from one SIP leg to the next, on this dial peer. |
Step 6 |
exit
Example: Router(config-dial-peer)# exit |
Exits the current mode. |
Configuring Random-Contact Support on a Cisco Unified Border Element
To configure random-contact support on a Cisco Unified Border Element, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
sip-ua
4.
credentials
username
username
password
password
realm
domain-name
5.
registrar
ipv4:
destination-address
random-contact
expires
expiry
6.
exit
7.
voice
service
voip
8.
sip
9.
random-contact
10.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode.
|
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
sip-ua
Example: Router(config)# sip-ua |
Enters SIP user-agent configuration mode. |
Step 4 |
credentials
username
username
password
password
realm
domain-name
Example: Router(config-sip-ua)# credentials username 123456 password cisco realm cisco |
Sends a SIP registration message from the Cisco Unified Border Element. |
Step 5 |
registrar
ipv4:
destination-address
random-contact
expires
expiry
Example: Router(config-sip-ua)# registrar ipv4:10.1.2.2 random-contact expires 200 |
Enables the SIP gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and Skinny Client Control Protocol (SCCP) phones with an external SIP proxy or SIP registrar.
|
Step 6 |
exit
Example: Router(config-sip-ua)# exit |
Exits the current mode. |
Step 7 |
voice
service
voip
Example: Router(config)# voice service voip |
Enters VoIP voice-service configuration mode. |
Step 8 |
sip
Example: Router(conf-voi-serv)# sip |
Enters voice service VoIP SIP configuration mode. |
Step 9 |
random-contact
Example: Router(conf-serv-sip)# random-contact |
Enables random-contact support on a Cisco Unified Border Element. |
Step 10 |
exit
Example: Router(conf-serv-sip)# exit |
Exits the current mode. |
Configuring Random-Contact Support for an Individual Dial Peer
To configure random-contact support for an individual dial peer, perform the steps in this section.
1.
enable
2.
configure
terminal
3.
sip-ua
4.
credentials
username
username
password
password
realm
domain-name
5.
registrar
ipv4:
destination-address
random-contact
expires
expiry
6.
exit
7.
dial-peer
voice
tag
voip
8.
voice-class
sip
random-contact
9.
exit
DETAILED STEPS
Command or Action | Purpose | |
---|---|---|
Step 1 |
enable
Example: Router> enable |
Enables privileged EXEC mode. |
Step 2 |
configure
terminal
Example: Router# configure terminal |
Enters global configuration mode. |
Step 3 |
sip-ua
Example: Router(config)# sip-ua |
Enters SIP user-agent configuration mode. |
Step 4 |
credentials
username
username
password
password
realm
domain-name
Example: Router(config-sip-ua)# credentials username 123456 password cisco realm cisco |
Sends a SIP registration message from the Cisco Unified Border Element. |
Step 5 |
registrar
ipv4:
destination-address
random-contact
expires
expiry
Example: Router(config-sip-ua)# registrar ipv4:10.1.2.2 random-contact expires 200 |
Enables the SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
|
Step 6 |
exit
Example: Router(config-sip-ua)# exit |
Exits the current mode. |
Step 7 |
dial-peer
voice
tag
voip
Example: Router(config)# dial-peer voice 2611 voip |
Defines the dial peer, specifies the method of voice encapsulation, and enters dial peer voice configuration mode. |
Step 8 |
voice-class
sip
random-contact
Example: Router(config-dial-peer)# voice-class sip random-contact |
Enables random-contact support on this dial peer. |
Step 9 |
exit
Example: Router(config-dial-peer)# exit |
Exits the current mode. |