Table Of Contents
Configuring SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
How to Configure Secure Call Signaling
Configuring Cisco Unified Communications Manager
Feature Information for SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
Configuring SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
First Published: October 7, 2009, OL-20685-01This feature adds Session Initiation Protocol/Transport Layer Security/Transmission Control Protocol (SIP/TLS/TCP) support for secure call signaling and Secure Real-time Transport Protocol (SRTP) for media encryption to establish a secure, encrypted connection between Cisco Unified IP Phones and a failover device using Cisco Unified Survivable Remote Site Telephony (Cisco SRST).
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the "Feature Information for SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST" section.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Contents
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How to Configure Secure Call Signaling
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Feature Information for SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
Prerequisites
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Cisco IOS Release 15.0(1)XA or later
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Cisco Unified IP Phone firmware release 8.5(3) or later
Information About Configuring SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
Beginning with Cisco IP Phone firmware update 8.5(3) and Cisco IOS Release 15.0(1)XA, Cisco SRST now supports SIP/UDP and SIP/TLS/TCP connections. Cisco SRST also now supports RTP and SRTP media connections, based on the security settings of the IP phone.
Cisco SRST SIP-to-SIP and SIP-to-PSTN support includes the following features:
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Basic calling
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Hold/resume
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Conference
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Transfer
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Blind transfer
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Call forward
Cisco SRST SIP-to-other supports only basic calling, although other features may work.
How to Configure Secure Call Signaling
This section contains the following tasks:
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Configuring Cisco Unified Communications Manager
Configuring Cisco Unified Communications Manager
In Cisco Unified Communications Manager, secure endpoints and non-secure endpoints must each have their own separate SRST reference configuration and device pool.
In Cisco Unified Communications Manager Administration, under System > SRST:
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For the secure SRST profile, Is SRST Secure? must be checked. The SIP port must be 5061.
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For the non-secure SRST profile, Is SRST Secure? can be checked or not, depending on your Skinny Call Control Protocol (SCCP) configuration. The SIP port must be 5060 (the default).
Under Device > Phone:
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Secure phones must belong to the pool that uses the secure SRST profile.
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Non-secure phones must belong to the pool that uses the non-secure SRST profile.
Configuring SRTP
This section explains how to configure SRTP on Cisco SRST.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
service voip
4.
srtp fallback
5.
allow-connections sip to h323
6.
allow-connections sip to sip
DETAILED STEPS
Configuring Secure Mode
This section explains how to configure secure mode on Cisco SRST.
SUMMARY STEPS
1.
sip
2.
url sip | sips
3.
srtp negotiate cisco
4.
exit
5.
exit
DETAILED STEPS
Configuring TLS
This section explains how to configure TLS on Cisco SRST.
SUMMARY STEPS
1.
voice register global
2.
security-policy secure
3.
exit
DETAILED STEPS
Configuring Strict Encryption
This section explains how to configure strict encryption on Cisco SRST.
SUMMARY STEPS
1.
sip-ua
2.
registrar ipv4:destination-address expires seconds
3.
xfer target dial-peer
4.
crypto signaling default trustpoint string [strict-cipher]
5.
end
DETAILED STEPS
Verifying the Configuration
The following examples show a sample configuration displayed by the show sip-ua status registrar command and the show voice register global command.
The show sip-ua status registrar command in privileged EXEC mode displays all SIP endpoints that are currently registered with the contact address:
Router# show sip-ua status registrarLine destination expires(sec) contacttransport call-idpeer============ =============== ============ ===============3029991 192.0.2.108 388 192.0.2.108TLS 00120014-4ae40064-f1a3e9fe-8d301072@192.0.2.1400043029993 192.0.2.103 382 192.0.2.103TCP 001bd433-1c840052-655cd596-4e992eed@192.0.2.1400113029982 192.0.2.106 406 192.0.2.106UDP 001d452c-dbba0056-0481d321-1f3f848d@192.0.2.1400013029983 192.0.2.106 406 192.0.2.106UDP 001d452c-dbba0057-1c69b699-d8dc6625@192.0.2.1400033029992 192.0.2.107 414 192.0.2.107TLS 001e7a25-50c9002c-48ef7663-50c71794@192.0.2.140005The show voice register global command in privileged EXEC mode displays all global configuration parameters associated with SIP phones:
Router# show voice register globalCONFIG [Version=7.1]========================Version 7.1Mode is srstMax-pool is 50Max-dn is 100Outbound-proxy is enabled and will use global configured valueSecurity Policy: DEVICE-DEFAULTSystem message is Welcome to ALOA Secure Fallbacktimeout interdigit 10network-locale[0] US (This is the default network locale for this box)network-locale[1] USnetwork-locale[2] USnetwork-locale[3] USnetwork-locale[4] USuser-locale[0] US (This is the default user locale for this box)user-locale[1] USuser-locale[2] USuser-locale[3] USuser-locale[4] USRouter#Additional References
The following sections provide references related to this feature.
Related Documents
Related Topic Document TitleCisco Unified SRST configuration
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Cisco Unified SIP SRST System Administrator Guide
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Cisco Unified SRST System Administrator Guide
Cisco IOS voice configuration
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Cisco IOS Voice Configuration Library
Standards
Standard TitleNo new or modified standards are supported by this feature, and support for existing standards has not been modified by this feature.
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MIBs
RFCs
RFC TitleNo new or modified RFCs are supported by this feature, and support for existing RFCs has not been modified by this feature.
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Technical Assistance
Command Reference
The following commands are introduced or modified in the feature or features documented in this module. For information about these commands, see the Cisco IOS Voice Command Reference at http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html. For information about all Cisco IOS commands, use the Command Lookup Tool at http://tools.cisco.com/Support/CLILookup or the Cisco IOS Master Command List, All Releases, at http://www.cisco.com/en/US/docs/ios/mcl/allreleasemcl/all_book.html.
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crypto signaling
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security-policy
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show sip ua status
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show voice register global
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srtp negotiate
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xfer target dial-peer
Feature Information for SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
Table 1 lists the release history for this feature.
Not all commands may be available in your Cisco IOS software release. For release information about a specific command, see the command reference documentation.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Note
Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. Unless noted otherwise, subsequent releases of that Cisco IOS software release train also support that feature.
Glossary
RTP—Real-time Transport Protocol. Delivers real-time data over IP packet-switched networks.
SCCP—Skinny Call Control Protocol. A communications protocol between certain clients and Cisco Unified Communications Manager.
SIP—Session Initiation Protocol. An Internet protocol for setting up, maintaining, and terminating multimedia services such as voice calls.
SRST—Survivable Remote Site Telephony. Provides Cisco Unified Communications Manager with fallback support for Cisco Unified IP phones attached to a Cisco router on your local network.
SRTP—Secure Real-time Transport Protocol. Provides encryption and message authentication to RTP.
TCP—Transmission Control Protocol. A transport layer protocol, part of the TCP/IP suite of Internet protocols.
TLS—Transport Layer Security. A security protocol that uses public-key cryptography.
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Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.
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