This document describes how to configure Cisco Unified Survivable Remote Site Telephony (SRST) on Cisco IOS routers to provide redundancy to Cisco Session Initiation Protocol (SIP) Phones.
Prerequisites for Configuring Cisco Unified SIP SRST:
Before configuring Cisco Unified SIP SRST, you must do the following:
Figure shows the Cisco SIP SRST
Cisco Unified SRST Configuration in CUCM
The SRST feature in Cisco Unified Communication Manager (CUCM) provides SIP IP-Phones with the information needed to find the relative gateway to register with when they lose contact with CUCM servers.
SRST Reference Definition: SRST reference comprises the gateway which can provide
limited CUCM functionality when all other CUCM Servers for IP Phones are unreachable.
Follow the procedure to configure SRST in CUCM at the Main Site:
CUCM Device Pool Configuration
1.The SIP SRST reference as shown above is assigned to IP Phones using Device Pools.
Follow these steps to configure Cisco Unified SIP SRST on a Cisco IOS gateway at the Remote Site
Step1: Enter configuration mode to activate SIP SRST
Step2: Define the IP address and port to which the SIP SRST service binds
Step3: Define the maximum number of DNs to support
Step4: Define the maximum number of IP Phones to support
SIP Configuration on the Router
1) Enable SIP B2BUA and SIP Registrar Functionality on SIP SRST router
MS-VGW-67(config)#voice service voip
MS-VGW-67(conf-voi-serv)#allow-connections sip to sip
MS-VGW-67(config-sip-ua)#registrar server expires max 600 min 60
2) Enter “voice Register global” mode to allow globally assign characteristics to SIP IP phones.
MS-VGW-67(config)#voice register global
MS-VGW-67(config-register-global)#mode srst/esrst >>>>>>>>(ESRST is optional enables more features in the Cisco Unified ESRST mode.)
MS-VGW-67(config-register-global)# system message "SIP SRST 3845 Service” >>> the System Msg display on the Phones
MS-VGW-67(config-register-global)# max-dn 200 >>> Max number of directory number on the SRST system, platform dependent
MS-VGW-67(config-register-global)# max-pool 100 >>> Max number of Pools on the SRST system, platform dependent
3) Configure the backup registrar (SIP SRST) services for SIP phones
voice register pool 1
id network 192.168.1.0 mask 255.255.255.0 >>> See below Note #1
call-forward b2bua busy 9100 >>> See below Note #2
call-forward b2bua noan 9100 timeout 10 >>> See below Note #2
dtmf-relay rtp-nte sip-notify
4) Enable sip-ua for external registrar (required by external registrar CUCM)
registrar ipv4:192.168.0.2 expires 3600 >>> 192.168.0.2 is CUCM IP address
5) Enable MWI-relay server for Messaging Waiting Indicator to work on SIP SRST
mwi-server ipv4:192.168.1.111 expires 3600 port 5060 transport udp >>> IP is CUE IP
6) To disable supplementary services if multiple SRST routers are required on remote office
voice service voip
no supplementary-service sip refer >>> To disable the SS using REFER
Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is not supported for a mix of SCCP and SIP endpoints.
With disabled SS configuration, REFER messaged used for call transfer and redirect responses for call forwarding will be disabled from being sent to the destination by SRST. Hairpin will be used instead.
7) Verify the SIP IP phone failover on SRST
show sip-ua status registrar
Show voice register pool xx
show voice register dial-peers
show voice register statistics
8) Debug the SIP phone registration process
debug voice register errors
debug voice register events
debug ccsip message
How to Verify the SIP IP phone fallback from CUCM to Local SRST router
This session will explore the process between SIP IP Phone and SIP SRST router by either show CLI output or debug messaged on SRST router only.
1) When a SIP IP Phone in normal operation with CUCM registration, “show sip-ua status registrar” on SRST router:
srst#show sip-ua status registrar
Line destination expires(sec) contact
2) Turn on the debug ‘debug ccsip message” on SRST router:
3) “show voice register dial-peer” on SRST router should return nothing because no SIP phone has been registered with SRST yet.
4) Now the WAN link is down, turn on the debug ccsip message on SRST router.
5) Do CLI “show sip-ua status registrar” on SRST router, now the line numbers with IP address should be shown.
6) Do CLI “show voice register dial-peer” on SRST router
7) During SRST mode, SIP phones will keep polling CUCM periodically, with SRST monitor Duration (default = 120 sec) setup on CUCM, SIP IP phones won’t register back to CUCM until WAN link to CUCM is stable and the CUCM is ready to take full registration process.
8) When SIP IP phone re-home to CUCM, it will un-register first from the SRST router and send normal SIP REGISTER message to the primary CUCM with expires >0
9) With debug ccsip message on SRST router.
10) CLI ‘show sip-ua status registrar” and “shoe voice register dial-peer” will both return nothing after SIP phones rehome to the primary CUCM.
Supported Feature in the E-SRST mode: