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Table Of Contents

Getting Started

Contents

Comparison of Cisco SIP SRST 3.0 and Cisco Unified SIP SRST 4.0

Configuration and Upgrade Tasks

How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0

Disabling Call Redirection

Enabling SIP-to-SIP Connection Capabilities


Getting Started

Note Prior to version 4.0, the name of this product was Cisco SIP SRST.

This chapter describes the main tasks necessary for the following:

•Running Cisco Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) 3.0 for the first time

•Running Cisco Unified SIP SRST 4.0 for the first time

•Upgrading from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0

Note that upgrades from Cisco SIP SRST 3.4 to Cisco Unified SIP SRST 4.0 are not impacted by the issues discussed in this chapter.

Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vcl.htm.

Contents

• Comparison of Cisco SIP SRST 3.0 and Cisco Unified SIP SRST 4.0

• Configuration and Upgrade Tasks

• How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0

Comparison of Cisco SIP SRST 3.0 and Cisco Unified SIP SRST 4.0

Cisco  SIP SRST 3.0, Cisco IOS Release 12.2(15)ZJ Through Cisco IOS Release 12.4

Cisco SIP SRST 3.0 was a predecessor to Cisco Unified SIP SRST 4.0. In Cisco SIP SRST 3.0, you could configure a Cisco IOS voice gateway to act as a SIP redirect server. The voice gateway would respond to the originator of a call with a SIP Redirect message, and the Redirect message allowed the SIP phone that originated the call to establish a call to its destination. In addition, several commands in voice register pool configuration mode were introduced that allowed registration permission control.

Cisco Unified SIP SRST V4.0, Cisco IOS Release 12.4(4)XC

With Cisco Unified  SIP SRST 4.0, a SIP redirect server is not necessary. Instead, a back-to-back user agent (B2BUA) server routes the call as desired. A B2BUA is a separate call agent that has more features than a redirect server, which can accept and forward calls only. With a B2BUA you can also configure call blocking and call forwarding. In call forwarding, the B2BUA forwards calls on behalf of the phone, while maintaining a presence as call middleman in the call path.

Configuration and Upgrade Tasks

The table below lists the high-level steps you need to take to upgrade to Cisco Unified SIP SRST 4.0. It also lists the high-level steps you need to take in order to run Version 3.0 or Version 4.0.

Cisco Unified SIP SRST Version
Instructions and Procedures

If you are interested in Cisco SIP SRST 3.0 (using a redirect server), complete these procedures.

Cisco SIP SRST Version 3.0 provides a backup to an external SIP proxy server by providing basic registrar and redirect services. The following chapters provide full Version 3.0 information, including basic voice register pool configurations.

• Configuring the SIP Registrar

• Configuring Backup Registrar Service to SIP Phones

• Configuring Cisco Unified SIP SRST Features Using Redirect Mode (for Version 3.0 Only)

If you are interested in Cisco Unified SIP SRST 4.0 (using a B2BUA) and have never used Cisco Unified SIP SRST in the past, complete these procedures.

VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls over the VoIP network. The following task describes how to allow SIP connections:

• Enabling SIP-to-SIP Connection Capabilities

SIP registrar functionality in Cisco IOS software is a required part of Cisco Unified SIP SRST. A registrar accepts SIP Register requests and dynamically builds VoIP dial peers, allowing the Cisco IOS voice gateway software to route calls to SIP phones. The following task describes how to configure the SIP registrar:

• Configuring the SIP Registrar

Configure a basic voice register pool:

• Configuring Backup Registrar Service to SIP Phones

You are now ready to configure Version 4.0 features such as call blocking and call forwarding. The following chapter describes the call blocking and call forwarding configurations:

• Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only)

If you are currently running Version 3.0 and want to upgrade to Version 4.0, complete these procedures.

Since Version 4.0 uses a B2BUA and not a redirect server, call redirection must be disabled as described in the following task:

• Disabling Call Redirection

VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls over the VoIP network. The following task describes how to allow SIP connections:

• Enabling SIP-to-SIP Connection Capabilities

You are now ready to configure Version 4.0 features such as call blocking and call forwarding. The following chapter describes the call blocking and call forwarding configurations:

• Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only)


How to Upgrade from Cisco SIP SRST 3.0 to Cisco Unified SIP SRST 4.0

This section contains the following procedures:

• Disabling Call Redirection (required)

• Enabling SIP-to-SIP Connection Capabilities (required)

Disabling Call Redirection

Because Version 4.0 uses a B2BUA and not a redirect server, call redirection must be disabled if it was previously enabled. Complete the following tasks as required, depending on whether call redirection was enabled globally or on a dial-peer basis.

• Disabling Call Redirection Globally

• Disabling Call Redirection on a Specific VoIP Dial Peer

Disabling Call Redirection Globally

To disable global IP-to-IP call redirection for all VoIP dial peers, use voice service configuration mode.

Note When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the specific dial peer takes precedence over the global configuration entered under voice service configuration mode.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. no redirect ip2ip

5. end

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

•Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

voice service voip

Example:

Router(config)# voice service voip

Enters voice service configuration mode.

Step 4 

no redirect ip2ip

Example:

Router(config-voi-srv)# no redirect ip2ip

Disables redirection of SIP phone calls to SIP phone calls globally using the Cisco IOS voice gateway.

Step 5 

end

Example:

Router(config-voi-srv)# end

Returns to privileged EXEC mode.

Disabling Call Redirection on a Specific VoIP Dial Peer

To disable IP-to-IP call redirection for a specific VoIP dial peer, disable it on the inbound dial peer where it was originally enabled.

Note When IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the specific dial peer takes precedence over the global configuration entered under voice service configuration mode.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. no redirect ip2ip

5. end

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

•Enter your password if prompted.

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 25 voip

Enters dial-peer configuration mode.

•tag—A number that uniquely identifies the dial peer (this number has local significance only).

•voip—Indicates that this is a VoIP peer using voice encapsulation on the POTS network and is used for configuring redirect.

Step 4 

no redirect ip2ip

Example:

Router(config-dial-peer)# no redirect ip2ip

Disables redirection of SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS voice gateway.

Step 5 

end

Example:

Router(config-dial-peer)# end

Returns to privileged EXEC mode.

Enabling SIP-to-SIP Connection Capabilities

VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls over the VoIP network. For Cisco Unified SIP SRST 4.0 we enable SIP-to-SIP connections for hairpin call routing. The B2BUA that routes the call uses the SIP-to-SIP connection. Because VoIP-to-VoIP connections are disabled on the router by default, they must be explicitly enabled to use call routing.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. allow-connections sip to sip

5. end

DETAILED STEPS

 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

•Enter your password if prompted.

Step 2 

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3 

voice service voip

Example:

Router(config)# voice service voip

Enters voice service configuration mode to establish global call transfer and forwarding parameters.

Step 4 

allow-connections sip to sip

Example:

Router(config-voi-srv)# allow-connections sip to sip

Enables VoIP-to-VoIP call connections. Use the no form of the command to disable VoIP-to-VoIP connections, which is the default.

Step 5 

end

Example:

Router(conf-voi-serv)# end

Returns to privileged EXEC mode.

What to Do Next

SIP registrar functionality in Cisco IOS software is a required part of Cisco Unified SIP SRST. By default, Cisco Unified SIP SRST is not enabled and cannot accept SIP register messages. To configure the SIP registrar to accept incoming SIP Register messages, see the "Configuring the SIP Registrar" chapter.

To configure a basic voice register pool, see "Configuring Backup Registrar Service to SIP Phones" section on page 26.

To configure call forwarding or call blocking, see the "Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only)" chapter.

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Posted: Mon May 1 14:12:17 PDT 2006
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