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Cisco IOS Software Releases 12.2 T

SIP Extensions for Caller Identity and Privacy

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Table Of Contents

SIP Extensions for Caller Identity and Privacy

Contents

Prerequisites for SIP Extensions for Caller Identity and Privacy

Restrictions for SIP Extensions for Caller Identity and Privacy

Information About SIP Extensions for Caller Identity and Privacy

Benefits of SIP Extensions for Remote Party ID

Privacy, Screening, and Presentation Indicators

Remote-Party-ID Implementation

Inbound and Outbound Call Flows

Remote-Party-ID in SIP and PSTN Messages

Remote-Party-ID Header

Remote-Party-ID Syntax

ISDN Syntax

Screening and Presentation Information

How to Configure SIP Extensions for Caller Identity and Privacy

Configuring Remote Party-ID

What to Do Next

Configuring SIP to PSTN Calling-Info Policy

What to Do Next

Configuring PSTN to SIP Calling-Info Policy

What to Do Next

Verifying SIP Extensions for Caller Identity and Privacy

Configuration Examples for SIP Extensions for Caller Identity and Privacy

Configuring PSTN to SIP Calling-Info Example

Verifying SIP Extensions for Caller Identity and Privacy Example

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

calling-info pstn-to-sip

calling-info sip-to-pstn

remote-party-id

Glossary


SIP Extensions for Caller Identity and Privacy

The SIP Extensions for Caller Identity and Privacy feature provides support for privacy indication, network verification, and screening of a call participant name and number.

Feature Specifications for SIP Extensions for Caller Identity and Privacy

Feature History
 
Release
Modification

12.2(13)T

This feature was introduced.

Supported Platforms

Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800 and Cisco AS5850 platforms.


Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.

Cisco Feature Navigator is a web-based tool that enables you to determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco  Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

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Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

http://www.cisco.com/go/fn

Availability of Cisco IOS Software Images

Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.

Contents

•Prerequisites for SIP Extensions for Caller Identity and Privacy

•Restrictions for SIP Extensions for Caller Identity and Privacy

•Information About SIP Extensions for Caller Identity and Privacy

•How to Configure SIP Extensions for Caller Identity and Privacy

•Configuration Examples for SIP Extensions for Caller Identity and Privacy

•Additional References

•Command Reference

•Glossary

Prerequisites for SIP Extensions for Caller Identity and Privacy

•Ensure that your Cisco router has the minimum memory requirements necessary for voice capabilities.

•Ensure that the gateway has voice functionality and is configured for SIP. For more information about configuring SIP, refer to

Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2, the "Configuring SIP for VoIP" chapter.

•Establish a working IP network. For more information about configuring IP, refer to

Cisco IOS IP Configuration Guide, Release 12.2.

Restrictions for SIP Extensions for Caller Identity and Privacy

This feature does not support the Anonymity header described in the Internet Engineering Task Force (IETF) specification, draft-ietf-privacy-.02.txt. The new feature implements presentation level anonymity at Layer 5, rather than at the IP address level. Since the SIP gateway assumes that all adjacent signaling devices are trusted, it is recommended that border SIP proxy servers enforce anonymity policies at administrative boundaries.

The IETF specification, draft-ietf-privacy-.02.txt, for mapping of North American Numbering Plan Area (NANPA) defined Automatic Number Identification Information Indicators (ANI II) or Originating Line Information (OLI) digits, is still under development. The current implementation of SIP Extensions for Remote Party ID supports carrying the ANI II digits as digits, rather than as a string representation of the numbering plan-tagged ANI II digits.

Information About SIP Extensions for Caller Identity and Privacy

To configure the SIP Extensions for Caller Identity and Privacy feature, you must understand the following concepts:

•Benefits of SIP Extensions for Remote Party ID

•Privacy, Screening, and Presentation Indicators

•Remote-Party-ID Implementation

•Inbound and Outbound Call Flows

•Remote-Party-ID in SIP and PSTN Messages

Benefits of SIP Extensions for Remote Party ID

•Expands Public Switched Telephone Network (PSTN) inter-operability

•Supports the ability to override privacy and screening indicators

•Enables network verification and screening of a call participant identity by SIP proxy servers

•Supports logging of screened identity information in accounting records for billing information

•Provides enhanced subscriber information that supports the enablement of service creation platforms and application servers for service providers

•Allows the service provider enhanced control of the ability to identify a subscriber and its qualifications within the network

Privacy, Screening, and Presentation Indicators

Cisco implements this feature on SIP trunking gateways by supporting a new header, Remote-Party-ID, as defined in the IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. The Remote-Party-ID header identifies the calling party and carries presentation and screening information. In previous SIP implementations, the From header was used to indicate calling party identity, and once defined in the initial INVITE request, could not be modified for that session. Implementing the Remote-Party-ID header, which can be modified, added, or removed as a call session is being established, overcomes previous limitations and enables call participant privacy indication, screening, and verification. The new feature uses the Remote-Party-ID header to support translation capability between Integrated Services Digital Networks (ISDN) messages and Remote-Party-ID SIP tags. The new SIP header also enables support for certain telephony services, and some regulatory and public safety requirements, by providing screening and presentation indicators.

The SIP Extensions for Caller Identity and Privacy feature introduces new command-line interface (CLI) commands to enable remote-party-id translations and to configure alternative calling information treatments for calls entering the SIP trunking gateway. Configurable treatment options are:

•Calling name and number pass-through (default).

•No calling name or number sent in the forwarded Setup message.

•Calling name unconditionally set to the configured string in the forwarded Setup message.

•Calling number unconditionally set to the configured string in the forwarded Setup message.

A new command allows you to configure alternative calling information treatments for calls exiting the SIP trunking gateway. Configurable treatment options are:

•Calling name and number pass-through (default).

•No calling name or number sent in the forwarded INVITE message.

•Display-name of the From header unconditionally set to the configured string in the forwarded INVITE message.

•User part of the From header unconditionally set to the configured string in the forwarded INVITE message.

•Display-name of the Remote-Party-ID header unconditionally set to the configured string in the forwarded INVITE message.

•User part of the Remote-Party-ID header unconditionally set to the configured string in the forwarded INVITE message.

Remote-Party-ID Implementation

This section discusses the implementation of the Remote-Party-ID feature in a SIP network. Before the implementation of this feature, there was no mechanism to modify the contents of the From header field. With the feature enabled, SIP gateways provide translation capability for ISDN screening and presentation identifiers in call setup messages. SIP gateways and proxy servers require configuration to support Remote-Party-ID functionality.

Figure 1 illustrates a typical network where the feature is implemented. Gateway C is configured for unscreened discard, that is, if the incoming SIP INVITE request does not contain a screened Remote-Party-ID header (;screen=yes), no calling name or number is sent in the forwarded Setup message.

Figure 1 Wholesaler SIP Network

Note With respect to privacy and screening indication, it is the responsibility of the proxy server to protect display-name information and enforce privacy policies at the administrative boundary.

In the following sections, Figure 2 through Figure 9 illustrate various calling information treatment options using the new commands available with the feature. Calling information treatment is determined by the parameters specified in the Setup message and Remote-Party-ID configuration on the SIP gateway.

Inbound and Outbound Call Flows

This section presents inbound and outbound call flows for the Remote-Party-ID feature. Figure 2 shows the PSTN-to-SIP default behavior where the calling party name and number are passed. The feature enables this treatment by default and no configuration is required.

Figure 2 PSTN to SIP Default Call Flow with Remote-Party-ID Translation, No Privacy Requested

Figure 3 shows the SIP to PSTN default behavior where the calling party name and number are passed. This feature enables this treatment by default and no configuration is required.

Figure 3 SIP to PSTN Default Call Flow with Remote-Party-ID

Figure 4 shows the call flow for discarding the calling name and number at Gateway B. The Setup message includes ISDN information elements (IEs) that specify calling information treatment. The INVITE message from Gateway A includes the corresponding Remote-Party-ID SIP tags. To configure Gateway B to discard calling name and number, use the following commands:

•remote-party-id

•calling-info sip-to-pstn unscreened discard

Figure 4 Discarding Calling Name and Number at Gateway

Figure 5 shows Gateway B overriding the calling name and number received in the Setup message from Gateway A. To configure Gateway B to override calling name and number, use the following commands:

•remote-party-id

•calling-info sip-to-pstn name set name

•calling-info sip-to-pstn number set number

Figure 5 Overriding Calling Name and Number at Gateway

In Figure 6 the trunking SIP gateway is configured to override the calling name and number of the From header. To configure this call treatment option, use the following commands:

•remote-party-id

•calling-info pstn-to-sip from name set name

•calling-info pstn-to-sip from number set number

Figure 6 Overriding Calling Name and Number of From Header

Figure 7 illustrates translation of OLI or ANI II digits for a billing application. The Remote-Party-ID feature enables this treatment by default; no configuration tasks are required. If the feature was disabled by using the no remote-party-id command, use the remote-party-id command to reenable the feature.

Figure 7 Passing OLI from CAS to SIP

Figure 8 and Figure 9 illustrate the SIP trunking gateway capability to provide translation between ISDN screening and presentation identifiers and SIP Remote-Party-ID extensions. The two figures show the difference in call treatment, with and without privacy requested. With no privacy requested, as shown in Figure 8, the calling party name and number are passed unchanged. With privacy requested, as shown in Figure 9, screened identity information is still logged in accounting records for billing information, but the user field is not populated in the From header of the outgoing INVITE message, and the display-name is populated with "anonymous."

Figure 8 SIP to PSTN Call Flow with Remote-Party-ID Translation, No Privacy Requested

Figure 9 PSTN to SIP Call Flow with Remote-Party-ID, Privacy Requested

Remote-Party-ID in SIP and PSTN Messages

The ability to provide marking, screening, and PSTN translation of identity information to and from Remote-Party-ID extensions is supported in SIP INVITE and PSTN messages. This section discusses the formats of SIP INVITE and PSTN messages, and has the following subsections:

•Remote-Party-ID Header

•Remote-Party-ID Syntax

•ISDN Syntax

•Screening and Presentation Information

Remote-Party-ID Header

The SIP Remote-Party-ID header identifies the calling party and includes user, party, screen and privacy headers that specify how a call is presented and screened. The new header contains a URL and an optional display name that identifies a user. A valid Remote-Party-ID header may be either a SIP URL or a TEL URL. See the sections Remote-Party-ID Syntax and Screening and Presentation Information for more information on the syntax of the new header. The following example shows representative Remote-Party-ID headers, including user, party, screen, and privacy. 

02:32:17:Received:

INVITE sip:3331000@172.27.184.118:5060;user=phone SIP/2.0

Via:SIP/2.0/UDP 10.0.0.1:5070

Supported:org.ietf.sip.100rel

From:"alice" <sip:555-1001@10.0.0.1:5070>

To:sip:555-1002@172.27.184.118:5060

Remote-Party-ID:"Alice Smith" 
<sip:5551111@161.44.147.67;user=phone>;party=calling;screen=no;privacy=off

Call-ID:00000001@10.0.0.1:5070

CSeq:1 INVITE

Contact:"alice" <sip:10.0.0.1:5070>

Content-Type:application/sdp


v=0

o=- 2890844526 2890844526 IN IP4 A3C47F2146789F0

s=-

c=IN IP4 10.0.0.1

t=36124033 0

m=audio 49170 RTP/AVP 0

Remote-Party-ID Syntax

Remote-Party-ID fields identify the calling party depending upon how the field is marked. If the party is unmarked, a Remote-Party-ID in a header represents the identity of the calling party.

Remote-Party-ID follows the Augmented Backus-Naur Format (ABNF). Refer to draft-ietf-sip-privacy-02.txt for the definitive specification. Fields are defined as follows:

•Remote-Party-ID = "Remote-Party-ID" ":" [display-name] "<" addr-spec ">" *(";" rpi-token)

•rpi-token = rpi-screen | rpi-pty-type | rpi-id-type | rpi-privacy | other-rpi-token

•rpi-screen = "screen" "=" ("no" | "yes" )

•rpi-pty-type = "party" "=" ( "calling" | "called" | token )

•rpi-id-type = "id-type" "=" ( "subscriber" | "user" | "alias" | "return" | "term" | token )

•rpi-privacy = "privacy" "=" 1#( ("full" | "name" | "uri" | "off" | token ) [ "-" ( "network" | token ) ] )

•other-rpi-token = ["-"] token ["=" (token | quoted-string)]

ISDN Syntax

ISDN messages follow the format specified in ISDN Basic Rate Interface Call Control Switching and Signalling Generic Requirements, GR-268-CORE, July 1998, to signal call control. ISDN messages are composed of information elements (IEs). The SIP Extensions for Remote Party ID feature uses Calling Party Number and Display Text IEs to provide specified screening and presentation treatment. The Calling Party Number IE specifies the origin of the calling number and presentation status, and the Display Text IE supplies calling party name information that is formatted for display by a terminal for a human user. See the Setup message in Figure 2 for sample IE information.

Screening and Presentation Information

The Remote-Party-ID header and ISDN Setup messages contain tags used to specify screened identity information. Table 1 lists translation of screening and presentation information included in the Remote-Party-ID SIP tags for SIP to PSTN networks. Table 2 provides the same translation for PSTN to SIP networks. Table 3 lists the corresponding translation for ISDN tags in binary and hex formats.

Table 1 SIP to PSTN Translation of Screening And Presentation Information

Remote-Party-ID SIP Tags
PSTN Octet 3A

;privacy=off;screen=no

Presentation allowed of user-provided number, number not screened (0x00)

;privacy=off;screen=yes

Presentation allowed of user-provided number, number passed network screening (0x01)

;privacy=[full|uri|name];screen=no

Presentation prohibited of user-provided number, number not screened (0x20)

;privacy=[full|uri|name];screen=yes

Presentation prohibited of user-provided number, number passed network screening (0x21)

;screen=no

Presentation allowed of user-provided number, number not screened (0x00)

;screen=yes

Presentation allowed of user-provided number, number passed network screening (0x01)

;privacy=off

Presentation allowed of user-provided number, number not screened (0x00)

;privacy=[full|uri|name]

Presentation prohibited of user-provided number, number not screened (0x20)

(no screen or privacy tags)

Presentation allowed of user-provided number, number not screened (0x00)


Table 2 PSTN to SIP Translation of Screening and Presentation Information

PSTN Octet 3A
Remote-Party-ID SIP Tags

Presentation allowed of user-provided number, number not screened (0x00)

;privacy=off;screen=no

Presentation allowed of user-provided number, number passed network screening (0x01)

;privacy=off;screen=yes

Presentation allowed of user-provided number, number failed network screening (0x02)

;privacy=off;screen=no

Presentation allowed of network-provided number (0x03)

;privacy=off;screen=yes

Presentation prohibited of user-provided number, number not screened (0x20)

;privacy=full;screen=no

Presentation prohibited of user-provided number, number passed network screening (0x21)

;privacy=full;screen=yes

Presentation prohibited of user-provided number, number failed network screening (0x22)

;privacy=full;screen=no

Presentation prohibited of network-provided number (0x23)

;privacy=full;screen=yes

Number not available (0x43)

(no screen or privacy tags are sent)


Table 3 Origin of Number and Presentation Status (Octet 3a)

Binary (Bits)
8 7 6 5 4 3 2 1
Hex
Meaning

1 0 0 0 0 0 0 0

0x80

Presentation allowed of user-provided number, number not screened

1 0 0 0 0 0 0 1

0x81

Presentation allowed of user-provided number, number passed network screening

1 0 0 0 0 0 1 0

0x82

Presentation allowed of user-provided number, number failed network screening

1 0 0 0 0 0 1 1

0x83

Presentation allowed of network-provided number

1 0 1 0 0 0 0 0

0xA0

Presentation prohibited of user-provided number, number not screened

1 0 1 0 0 0 0 1

0xA1

Presentation prohibited of user-provided number, number passed network screening

1 0 1 0 0 0 1 1

0xA3

Presentation prohibited of network-provided number

1 0 0 0 0 0 1 1

0xC3

Number not available


How to Configure SIP Extensions for Caller Identity and Privacy

This section contains the following procedures. Each procedure is identified as either required or optional.

•Configuring Remote Party-ID (optional)

•Configuring SIP to PSTN Calling-Info Policy (optional)

•Configuring PSTN to SIP Calling-Info Policy (optional)

•Verifying SIP Extensions for Caller Identity and Privacy (optional)

Configuring Remote Party-ID

This feature is enabled by default; no configuration tasks are required to enable this feature. If it was disabled by using the no remote-party-id command, perform this task to reenable the feature.

SUMMARY STEPS

1. enable

2. configure {terminal | memory | network}

3. sip-ua

4. remote-party-id

5. exit

6. show running-config

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router# enable

Enables higher privilege levels, such as privileged EXEC mode.

•Enter your password if prompted.

Step 2 

configure {terminal | memory | network}

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

sip-ua

Example:

Router(config)# sip-ua

Enters sip-ua configuration mode.

Step 4 

remote-party-id

Example:

Router(config-sip-ua)# remote-party-id


Enables Remote-Party-ID support.

Step 5 

exit

Example:

Router(config-sip-ua)# exit

Exits sip-ua configuration mode.

Step 6 

show running-config

Example:

Router# show running-config

(Optional) Enables SIP Service Provider Interface (SPI) message tracing.

•Use the show running-config command to verify that remote-party-id support is enabled.


What to Do Next

Proceed to the section "Verifying SIP Extensions for Caller Identity and Privacy."

Configuring SIP to PSTN Calling-Info Policy

When Remote-Party-ID support is enabled, the default calling-info treatment is the following: the calling name and calling number are bidirectionally translated between the display-name and the user part of the Remote-Party-ID header of the SIP INVITE message and the calling name and calling number of the PSTN Setup message. If a PSTN to SIP call is marked as presentation prohibited, the user part is not populated in the From header of the outgoing INVITE message and the display-name is populated with "anonymous". Otherwise, the display-name and user part of the From header of the outgoing INVITE are populated with the calling name and calling number.

To override the default calling-info treatment, perform this task to optionally configure SIP to PSTN calling-info policy.

SUMMARY STEPS

1. enable

2. configure {terminal | memory | network}

3. sip-ua

4. calling-info sip-to-pstn [unscreened discard][name set name][number set number]

5. exit

6. show running-config

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables higher privilege levels, such as privileged EXEC mode.

•Enter your password if prompted.

Step 2 

configure {terminal | memory | network}

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

Router(config)# sip-ua

Enters SIP-UA configuration mode

Step 4 

Router(config-sip-ua# calling-info sip-to-pstn [unscreened discard] [name set name][number set number]

Example:

Router(config-sip-ua# calling-info sip-to-pstn

unscreened discard

Example:

Router(config-sip-ua# calling-info sip-to-pstn name set CompanyA

Example:

Router(config-sip-ua# calling-info sip-to-pstn number set 5551000


(Optional) Configures sip-to-pstn calling-info treatment.

•(Optional) Specifies that the calling name and number be discarded.If the incoming SIP INVITE message does not contain a screened (;screen=yes) Remote-Party-ID header, then no name or number is sent in the forwarded Setup message.

•(Optional) Specifies that the calling name be unconditionally set to the configured name in the forwarded Setup message.

•(Optional) Specifies the calling number be unconditionally set to the configured number in the forwarded Setup message.

Step 5 

Router(config-sip-ua# exit

Example:

Router(config-sip-ua)# exit

Exits sip-ua configuration mode.

Step 6 

Router# show running-config

Verifies that sip-to-pstn calling-info policy is operational.


What to Do Next

Proceed to the section "Verifying SIP Extensions for Caller Identity and Privacy."

Configuring PSTN to SIP Calling-Info Policy

To override the default calling-info treatment, perform this task to optionally configure PSTN to SIP calling-info policy.

SUMMARY STEPS

1. enable

2. configure {terminal | memory | network}

3. sip-ua

4. calling-info pstn-to-sip [unscreened discard] [from [name set name][number set number]][remote-party-id [name set name][number set number]]

5. exit

6. show running-config

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables higher privilege levels, such as privileged EXEC mode.

Enter your password if prompted.

Step 2 

configure {terminal | memory | network}

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

Router(config)# sip-ua

Example:

Router# sip-ua

Enters sip-ua configuration mode.

Step 4 

calling-info pstn-to-sip [unscreened discard] [from [name set name][number set number]][remote-party-id [name set name][number set number]]

Example:

router(config-sip-ua)# calling-info pstn-to-sip unscreened discard

Example:

router(config-sip-ua)# calling-info pstn-to-sip from name set CompanyA

Example:

router(config-sip-ua)# calling-info pstn-to-sip from number set 5552000

Example:

router(config-sip-ua)# calling-info pstn-to-sip remote-party-id name set CompanyA

Example:

router(config-sip-ua)# calling-info pstn-to-sip remote-party-id number set 5552000


Configures pstn-to-sip calling-info treatment.

•(Optional) Specifies that the calling name and number be discarded. If the incoming Setup is not marked as "user-provided, passed screening" or "network-provided," then no calling name or number is sent in the forwarded INVITE message.

•(Optional) Specifies that the display-name of the From header is unconditionally set to the configured name in the forwarded INVITE message.

•(Optional) Specifies that the user part of the From header is unconditionally set to the configured string in the forwarded INVITE message.

•(Optional) Specifies that the display-name of the Remote-Party-ID header is unconditionally set to the configured name in the forwarded INVITE message.

•(Optional) Specifies that the user part of the Remote-Party-ID header is unconditionally set to the configured number in the forwarded INVITE message.

Step 5 

exit

Example:

Router(config-sip-ua)# exit


Exits sip-ua configuration mode.

Step 6 

show running-config

Example:

router# show running-configuration

Verifies that pstn-to-sip calling-info policy is operational.


What to Do Next

Proceed to the section "Verifying SIP Extensions for Caller Identity and Privacy."

Verifying SIP Extensions for Caller Identity and Privacy

Perform this task to verify that the SIP Extensions for Caller Identity and Privacy feature is working.

SUMMARY STEPS

1. enable

2. debug ccsip messages

3. debug isdn q931

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router# enable

Enables higher privilege levels, such as privileged EXEC mode.

•Enter your password if prompted.

Step 2 

debug ccsip messages

Example:

Router# debug ccsip messages

(Optional) Displays all SIP SPI message tracing.

•Use this command to enable traces for SIP messages exchanged between the SIP UA client (UAC) and the access server.

Step 3 

debug isdn q931

Example:

Router# debug isdn q931

(Optional) Use this command to display information about call setup and teardown of ISDN network connections (Layer 3) between the local router (user side) and the network.


Configuration Examples for SIP Extensions for Caller Identity and Privacy

This section provides the following configuration examples:

•Configuring PSTN to SIP Calling-Info Example

•Verifying SIP Extensions for Caller Identity and Privacy Example

Configuring PSTN to SIP Calling-Info Example

In the following example, the PSTN name is set to Company A and the PSTN number is set to 5551000. 

Router(config-sip-ua)# calling-info sip-to-pstn name set CompanyA

Router(config-sip-ua)# calling-info sip-to-pstn number set 5551000

!

Router# show running-config


Building configuration...


Current configuration :2791 bytes

!

version 12.2

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

!

hostname 3640

!

voice-card 2

!

ip subnet-zero

!

no ip domain lookup

ip domain name cisco.com

ip name-server 172.18.195.113

!

isdn switch-type primary-ni

!

fax interface-type fax-mail

mta receive maximum-recipients 0

ccm-manager mgcp

!

controller T1 2/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24

!

controller T1 2/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24

!

interface Ethernet0/0

 ip address 172.18.197.22 255.255.255.0

 half-duplex

!

interface Serial0/0

 no ip address

 shutdown

!

interface TokenRing0/0

 no ip address

 shutdown

 ring-speed 16

!

interface FastEthernet1/0

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface Serial2/0:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn outgoing display-ie

 no cdp enable

!

interface Serial2/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn outgoing display-ie

 no cdp enable

!

ip classless

ip route 0.0.0.0 0.0.0.0 Ethernet0/0

no ip http server

ip pim bidir-enable

!

call rsvp-sync

!

voice-port 2/0:23

!

voice-port 2/1:23

!

voice-port 3/0/0

!

voice-port 3/0/1

!

mgcp ip qos dscp cs5 media

mgcp ip qos dscp cs3 signaling

!

mgcp profile default

!

dial-peer cor custom

!

dial-peer voice 1 voip

 incoming called-number 5552222

 destination-pattern 5552222

 session protocol sipv2

 session target ipv4:172.18.197.27

!

dial-peer voice 2 pots

 destination-pattern 5551111

 no digit-strip

 direct-inward-dial

 port 2/0:23

!

gateway 

!

sip-ua 

calling-info sip-to-pstn name set CompanyA

calling-info sip-to-pstn number set 5551000

!

line con 0

line aux 0

line vty 0 4

 login

!

end!

Verifying SIP Extensions for Caller Identity and Privacy Example

In the following examples, the output is displayed for each command in the section "Verifying SIP Extensions for Caller Identity and Privacy."

In these examples, the output is displayed for a gateway that has Remote-Party-ID header translation enabled. 

Router# debug isdn q931


debug isdn q931 is	 	ON.

Router#

00:14:30:ISDN Se2/1:23 Q931:RX <- SETUP pd = 8  callref = 0x0002 

	Bearer Capability i = 0x8090A2 

		Standard = CCITT 

		Transer Capability = Speech  

		Transfer Mode = Circuit 

		Transfer Rate = 64 kbit/s 

	Channel ID i = 0xA98381 

		Exclusive, Channel 1 

	Display i = 'CallerA' 

	Calling Party Number i = 0x0081, '5551111' 

		Plan:Unknown, Type:Unknown 

	Called Party Number i = 0x80, '5552222' 

		Plan:Unknown, Type:Unknown

00:14:30:ISDN Se2/1:23 Q931:TX -> CALL_PROC pd = 8  callref = 0x8002 

	Channel ID i = 0xA98381 

		Exclusive, Channel 1


Router# debug ccsip messages


SIP Call messages tracing is enabled


00:14:30:Sent:

INVITE sip:5552222@172.18.197.27:5060 SIP/2.0

Via:SIP/2.0/UDP  172.18.197.22:5060

From:"CallerA" <sip:5551111@172.18.197.22>;tag=D47FC-1F9F

To:<sip:5552222@172.18.197.27>

Date:Mon, 01 Mar 1993 00:14:30 GMT

Call-ID:8E97C441-14F011CC-8009E50C-94497697@172.18.197.22

Supported:timer,100rel

Min-SE: 1800

Cisco-Guid:2392230745-351277516-2147935500-2487842455

User-Agent:Cisco-SIPGateway/IOS-12.x

Allow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO

CSeq:101 INVITE

Max-Forwards:6

Remote-Party-ID:"CallerA" <sip:5551111@172.18.197.22>;party=calling;screen=yes;privacy=off

Timestamp:730944870

Contact:<sip:5551111@172.18.197.22:5060>

Expires:180

Allow-Events:telephone-event

Content-Type:application/sdp

Content-Length:242


v=0

o=CiscoSystemsSIP-GW-UserAgent 2388 5877 IN IP4 172.18.197.22

s=SIP Call

c=IN IP4 172.18.197.22

t=0 0

m=audio 17476 RTP/AVP 18 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=ptime:20

Additional References

For additional information related to the SIP Extensions for Caller Identity and Privacy feature, refer to the following references:

Related Documents

Related Topic
Document Title

SIP configuration tasks

Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2, "Configuring Session Initiation Protocol for Voice over IP" chapter

IP configuration tasks

Cisco IOS IP Configuration Guide, Release 12.2

IP configuration commands

Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2

VoIP configuration tasks

Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2


Standards

Standards1
Title

draft-ietf-privacy-.02.txt

SIP Extension for Caller Identity and Privacy

1 Not all supported standards are listed.


MIBs

MIBs1
MIBs Link

No new or modified MIBs are supported by this feature, and support for existing MIBs has not been modified by this feature.

To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

1 Not all supported MIBs are listed.


To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://tools.cisco.com/ITDIT/MIBS/servlet/index

If Cisco  MIB Locator does not support the MIB information that you need, you can also obtain a list of supported MIBs and download MIBs from the Cisco  MIBs page at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

To access Cisco MIB Locator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

http://www.cisco.com/register

RFCs

RFCs1
Title

RFC 2543

SIP: Session Initiation Protocol

1 Not all supported RFCs are listed.


Technical Assistance

Description
Link

Technical Assistance Center (TAC) home page, containing 30,000 pages of searchable technical content, including links to products, technologies, solutions, technical tips, tools, and lots more. Registered Cisco.com users can log in from this page to access even more content.

http://www.cisco.com/public/support/tac/home.shtml


Command Reference

This section documents new commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2T command reference publications.

•calling-info pstn-to-sip

•calling-info sip-to-pstn

•remote-party-id

calling-info pstn-to-sip

To specify calling information treatment for PSTN-to-SIP calls, use the calling-info pstn-to-sip command in SIP user agent configuration mode. To disable calling information treatment for PSTN-to-SIP calls, use the no form of this command.

calling-info pstn-to-sip {unscreened discard | {from | remote-party-id {name set name | number set number}}}

no calling-info pstn-to-sip

Syntax Description

unscreened discard

(Optional) Specifies that the calling name and number be discarded.

from name set name

(Optional) Specifies that the display-name of the From header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

from number set number

(Optional) Specifies that the user part of the From header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

remote-party-id name set name

(Optional) Specifies that the display-name of the Remote-Party-ID header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

remote-party-id number set number

(Optional) Specifies that the user part of the Remote-Party-ID header is unconditionally set to the configured ASCII string in the forwarded INVITE message.


Defaults

This command is disabled.

Command Modes

SIP user agent configuration

Command History

Release
Modification

12.2(13)T

This command was introduced.


Usage Guidelines

When a call exits the gateway, the calling-info pstn-to-sip treatments are applied.

Examples

The following example enables calling information treatment for PSTN-to-SIP calls and sets the company name and number: 

Router(config-sip-ua)# calling-info pstn-to-sip from name set CompanyA

Router(config-sip-ua)# calling-info pstn-to-sip from number set 5551000

Router(config-sip-ua)# exit

Router(config)# exit

Router# show running-config

Building configuration...


.

.

.

!

sip-ua 

 calling-info pstn-to-sip from name set CompanyA

 calling-info pstn-to-sip from number set 5551000

 no remote-party-id

!

.

.

.

Related Commands

Command
Description

calling-info sip-to-pstn

Specifies calling information treatment for SIP-to-PSTN calls.

debug ccsip events

Enables tracing of SIP SPI events.

debug ccsip messages

Enables tracing SIP messages exchanged between the SIP UA client and the access server.

debug isdn q931

Displays call setup and teardown of ISDN connections.

debug voice ccapi error

Enables tracing error logs in the call control API.

debug voip ccapi in out

Enables tracing the execution path through the call control API.


calling-info sip-to-pstn

To specify calling information treatment for SIP-to-PSTN calls, use the calling-info sip-to-pstn command in SIP user agent configuration mode. To disable calling information treatment for SIP-to-PSTN calls, use the no form of this command.

calling-info sip-to-pstn {unscreened discard | name set name | number set number}

no calling-info sip-to-pstn

Syntax Description

unscreened discard

(Optional) Specifies that the calling name and number be discarded.

name set name

(Optional) Specifies that the calling name be unconditionally set to the configured ASCII string in the forwarded Setup mesage.

number set number

(Optional) Specifies that he calling number be unconditionally set to the configured ASCII string in the forwarded Setup message.


Defaults

This command is disabled.

Command Modes

SIP user agent configuration

Command History

Release
Modification

12.2(13)T

This command was introduced.


Usage Guidelines

When a call enters the gateway, the calling-info sip-to-pstn treatments are applied.

Examples

The following example enables calling information treatment for SIP-to-PSTN calls and sets the company name to CompanyA and the number to 5551000: 

Router(config-sip-ua)# calling-info sip-to-pstn name set CompanyA

Router(config-sip-ua)# calling-info sip-to-pstn number set 5551000

Router(config-sip-ua)# exit

Router(config)# exit

Router# show running-config

Building configuration...


.

.

.

!

sip-ua 

 calling-info sip-to-pstn name set CompanyA

	 calling-info sip-to-pstn number set 5551000

!

.

.

.

Related Commands

Command
Description

debug ccsip events

Enables tracing of SIP SPI events.

debug ccsip messages

Enables SIP SPI message tracing.

debug isdn q931

Displays call setup and teardown of ISDN connections.

debug voip ccapi in out

Enables tracing the execution path through the call control API.

calling-info pstn-to-sip

Specifies calling information treatment for PSTN-to-SIP calls.


remote-party-id

To enable translation of the SIP header Remote-Party-ID, use the remote-party-id command in SIP user agent configuration mode. To disable Remote-Party-ID translation, use the no form of this command.

remote-party-id

no remote-party-id

Syntax Description

remote-party-id

Extracts the calling name and number from the Remote-Party-ID header.

no remote-party-id

Extracts the calling name and number from the From header.


Defaults

Remote-Party-ID translation is enabled

Command Modes

SIP user agent configuration

Command History

Release
Modification

12.2(13)T

This command was introduced.


Usage Guidelines

When the remote-party-id command is enabled, one of the following calling information treatments occurs:

•If a Remote-Party-ID header is present in the incoming INVITE message, the calling name and number extracted from the Remote-Party-ID header are sent as the calling name and number in the outgoing Setup message. This is the default behavior. Use the remote-party-id command to enable this option.

•When no Remote-Party-ID header is available, no translation occurs so the calling name and number are extracted from the From header and are sent as the calling name and number in the outgoing Setup message. This treatment also occurs when the feature is disabled.

Examples

The following example shows the Remote-Party-ID translation being enabled:

Router(config-sip-ua)# remote-party-id

Related Commands

Command
Description

debug ccsip events

Enables tracing of SIP SPI events.

debug ccsip messages

Enables SIP SPI message tracing.

debug isdn q931

Displays call setup and teardown of ISDN connections.

debug voice ccapi in out

Enables tracing the execution path through the call control API.


Glossary

call—In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call identifier. A point-to-point IP telephony conversation maps into a single SIP call.

call-ID—A general header that uniquely identifies a particular invitation or all registrations of a particular client.

call leg A logical connection between the router and another endpoint.

CLI—command-line interface.

INVITE—A message that initiates a session. It indicates that a user is invited to participate, provides a session description, indicates the type of media, and provides insight regarding the capabilities of the called and calling parties.

IP— Internet protocol. A connectionless protocol that operates at the network layer (layer 3) of the OSI model. IP provides features for addressing, type-of-service specification, fragmentation and reassemble, and security. Defined in RFC 791. This protocol works with TCP and is usually identified as TCP/IP.

ISDN—Integrated Services Digital Network. Communication protocol offered by telephone companies that permits telephone networks to carry data, voice, and other source traffic.

PSTN—Public Switched Telephone Network. PSTN refers to the local telephone company.

proxy—A SIP UAC or UAS that forwards requests and responses on behalf of another SIP UAC or UAS.

proxy server—An intermediary program that acts both as a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.

SDP—Session Description Protocol. Messages containing capabilities information that are exchanged between gateways.

session—A SIP session is a set of multimedia senders and receivers and the data streams flowing between the senders and receivers. A SIP multimedia conference is an example of a session. The called party can be invited several times by different calls to the same session.

SIP—Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF). Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.

SIP URL—Session Initiation Protocol Uniform Resource Locator. Used in SIP messages to indicate the originator, recipient, and destination of the SIP request. Takes the basic form of user@host, where user is a name or telephone number, and host is a domain name or network address.

TEL URL—Telephone Uniform Resource Locator. Describes voice call connections to a terminal. Can also be any connection through a voice messaging system or a service that can be operated using DTMF tones. Takes the basic form of tel:telephone subscriber number, where tel indicates a URL and requests the local entity to place a voice call, and telephone subscriber number is the number to receive the call.

UA—user agent. A combination of UAS and UAC that initiates and receives calls. See UAS and UAC.

UAC—user agent client. A client application that initiates a SIP request.

UAS—user agent server. A server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.

URL—Uniform Resource Locator. Standard address of any resource on the Internet that is part of the World Wide Web (WWW).

VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet.