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Table Of Contents
Configuring Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Prerequisites for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Restrictions for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Information About Cisco SIP SRST Version 3.0 Features Using Redirect Mode
How to Configure Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco SIP SRST
Configuring Sending 300 Multiple Choice Support
Configuration Examples for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Cisco SIP SRST Version 3.0: Example
Configuring Cisco SIP SRST Version 3.0 Features Using Redirect Mode
This chapter describes Survivable Remote Site Telephony (SRST) functionality for Session Initiation Protocol (SIP) networks that was introduced in Version 3.0.
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
• Prerequisites for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
• Restrictions for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
• Information About Cisco SIP SRST Version 3.0 Features Using Redirect Mode
• How to Configure Cisco SIP SRST Version 3.0 Features Using Redirect Mode
• Configuration Examples for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Prerequisites for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Complete the prerequisites documented in the "Prerequisites for Configuring Cisco SIP SRST" section in the "Cisco IOS SIP SRST Feature Overview" chapter.
Restrictions for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
See the restrictions documented in the "Restrictions for Configuring Cisco SIP SRST" section in the "Cisco IOS SIP SRST Feature Overview" chapter.
Information About Cisco SIP SRST Version 3.0 Features Using Redirect Mode
Cisco SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect services. These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy. The Cisco SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls.
To make maximum use of the Cisco SIP SRST service, the local SIP IP phones should support dual (concurrent) registration with both their primary SIP proxy or registrar and the Cisco SIP SRST backup registrar. Cisco SIP SRST works for the following types of calls:
•Local SIP IP phone to local SIP phone, if the main proxy is unavailable.
•Additional services like class of restriction (COR) for local SIP IP phones to the outgoing PSTN. For example, to block outgoing 1-900 numbers.
How to Configure Cisco SIP SRST Version 3.0 Features Using Redirect Mode
This section contains the following procedures:
• Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco SIP SRST (required)
• Configuring Sending 300 Multiple Choice Support (required)
Configuring Call Redirect Enhancements to Support Calls Between SIP IP Phones for Cisco SIP SRST
The call redirect enhancement supports calls from a local SIP phone to another local SIP phone through the Cisco IOS voice gateway. Prior to this enhancement, an attempt by a SIP phone to contact another local SIP phone using the Cisco IOS voice gateway as if it were a SIP proxy or redirect server would fail. However, now the Cisco IOS voice gateway can act as a SIP redirect server. The voice gateway responds to the originator with a SIP Redirect message, allowing the SIP phone that originated the call to establish a call to its destination.
The redirect ip2ip (voice service) and redirect ip2ip (dial-peer) commands allow you to enable the SIP functionality, globally or on a specific inbound dial peer. The default application on Cisco SIP SRST supports IP-to-IP redirection.
• Configuring Call Redirect Enhancements to Support Calls Globally
• Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer
Configuring Call Redirect Enhancements to Support Calls Globally
To enable 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. redirect ip2ip
5. end
DETAILED STEPS
Configuring Call Redirect Enhancements to Support Calls on a Specific VoIP Dial Peer
To enable IP-to-IP call redirection for a specific VoIP dial peer, configure it on an inbound dial peer in dial-peer configuration mode. The default application on Cisco SIP SRST supports IP-to-IP redirection.
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.
Restrictions
The redirect ip2ip command must be configured on an inbound dial peer of the gateway.
SUMMARY STEPS
1. enable
2. configure terminal
3. dial-peer voice tag voip
4. application application-name
5. redirect ip2ip
6. end
DETAILED STEPS
Configuring Sending 300 Multiple Choice Support
Prior to Cisco IOS Release 12.2(15)ZJ, when a call was redirected, the SIP gateway would send a 302 Moved Temporarily message. The first longest match route on a gateway (dial-peer destination pattern) was used in the Contact header of the 302 message. With Release 12.2(15)ZJ, if multiple routes to a destination exist for a redirected number (multiple dial peers are matched), the SIP gateway sends a 300 Multiple Choice message, and the multiple routes in the Contact header are listed.
The configuration below allows users to choose the order in which the routes appear in the Contact header.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice service voip
4. sip
5. redirect contact order [best-match | longest-match]
6. end
DETAILED STEPS
Configuration Examples for Cisco SIP SRST Version 3.0 Features Using Redirect Mode
This section provides the following configuration example.
• Cisco SIP SRST Version 3.0: Example
Note IP addresses and hostnames in examples are fictitious.
Cisco SIP SRST Version 3.0: Example
This section provides a configuration example to match the configuration tasks in the previous sections.
!
! Sets up the registrar server and enables IP-to-IP redirection and 300
! Multiple Choice support.
!
voice service voip
redirect ip2ip
sip
registrar server expires max 600 min 60
redirect contact order best-match
!
! Configures the voice-class codec with G.711uLaw and G729 codecs. The codecs are
! applied to the voice register pools.
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729br8
!
! The voice register pools define various pools that are used to match
! incoming REGISTER requests and create corresponding dial peers.
!
voice register pool 1
id mac 0030.94C2.A22A
preference 5
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
voice-class codec 1
!
voice register pool 2
id ip 192.168.0.3 mask 255.255.255.255
preference 5
cor outgoing call95 1 91021
proxy 10.2.161.187 preference 1
voice-class codec 1
!
voice register pool 3
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
preference 5
cor incoming call95 1 95011
cor outgoing call95 1 95011
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
max registrations 5
voice-class codec 1
!
voice register pool 4
id network 10.2.161.0 mask 255.255.255.0
number 1 94... preference 1
preference 5
cor incoming everywhere default
cor outgoing everywhere default
proxy 10.2.161.187 preference 1
max registrations 2
voice-class codec 1
!
! Configures translation rules to be applied in the voice register pools.
!
translation-rule 1
Rule 0 94 91
!
! Sets up proxy monitoring.
!
call fallback active
!
dial-peer cor custom
name 95
name 94
name 91
!
! Configures COR values to be applied to the voice register pool.
!
dial-peer cor list call95
member 95
!
dial-peer cor list call94
member 94
!
dial-peer cor list call91
member 91
!
dial-peer cor list everywhere
member 95
member 94
member 91
!
! Configures a voice port and a POTS dial peer for calls to and from the PSTN endpoints.
voice-port 1/0/0
!
dial-peer voice 91500 pots
corlist incoming call91
corlist outgoing call91
destination-pattern 91500
port 1/0/0
!
Where to Go Next
After configuring basic Cisco SIP SRST, the " Configuring Cisco SIP SRST Version 3.4 Features Using Back-to-Back User Agent Mode" chapter describes additional configurations to increase SIP phone functionality.
Posted: Thu Nov 10 08:40:18 PST 2005
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