I was in the process of migrating roughly 1,000 phones from legacy Cisco 794X and 796X SCCP phones over to newer Cisco 88XX SIP phones, and standing up several branch clinics with 88XX phones and Cisco 44XX ISR routers running MGCP.
The problem? Early on, I didn’t realize that the router SRST configuration is completely different for SIP phones than it is for SCCP phones (call-manager-fallback). So I was loading SCCP SRST configurations onto branch routers that would never see an SCCP phone!
Let’s backtrack a bit and walk through how SIP SRST works and how to configure it properly.
What is SRST?
SRST is Survivable Remote Site Telephony. It allows branch office IP phones to register directly to their local branch voice router in the event of a WAN failure so they can continue processing internal calls locally and routing emergency/PSTN calls out a local PSTN circuit (such as an FXO POTS line, PRI, or local SIP trunk).
Consider a typical centralized UC architecture with a remote medical clinic:
NORMAL OPERATION (WAN UP):
[Branch 88XX SIP Phone] ──(WAN)──> [HQ CUCM Cluster] ──> [Central SIP ITSP]
[Branch 44XX ISR (MGCP)] ──(Local FXO POTS Line Standby for 911)
WAN OUTAGE (SIP SRST FALLBACK):
[Branch 88XX SIP Phone] ──(LAN SIP Register)──> [Branch 44XX ISR (SIP SRST)]
│
└──> [Local FXO POTS Line (911)]
If the branch phone can no longer reach CUCM—whether the WAN link drops, the HQ router goes down, or CUCM is unreachable—our SIP phone automatically re-registers to the local branch router via SIP SRST and utilizes the local POTS FXO line to route emergency calls with the branch clinic’s physical PSAP address.
Important Note: If only the HQ SIP ITSP goes down while the WAN is still up, phones remain registered to CUCM and will not enter SRST. To handle an HQ ITSP outage, include the branch gateway in your CUCM Standard Local Route Group or backup Route List so CUCM can still route calls out the branch FXO port over the WAN.
Configuring SIP SRST on the Cisco IOS XE Router
Assuming you already have a baseline router configuration and MGCP gateway control set up, let’s focus on the SIP SRST configuration.
1. Enable the SIP Registrar in voice service voip
Unlike SCCP phones, SIP phones act as a SIP User-Agent (UA) and require the branch router to act as a SIP Registrar server:
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
registrar server expires max 120 min 60
You can read more about the supplementary-service commands (which are Cisco best practice for SIP SRST call transfers and hold/resume) in the Cisco IOS Voice Command Reference.
2. Configure voice register global
Next, we configure voice register global, which defines the mode, system display message, and the maximum number of directory numbers (max-dn) and endpoints (max-pool) licensed for SRST:
voice register global
default mode
no allow-hash-in-dn
system message "SRST Mode-Emergency Calls Only"
max-dn 10
max-pool 10
3. Configure voice register pool
Now we configure the voice register pool to tell the router which IP subnet is allowed to register, along with DTMF relay and codec settings:
voice register pool 1
id network 10.80.1.0 mask 255.255.255.0
dtmf-relay rtp-nte sip-kpml sip-notify
codec g711ulaw
no vad
Why identify phones using id network 10.80.1.0 mask 255.255.255.0? Because all phones at the branch sit on a dedicated Voice VLAN (10.80.1.0/24) via DHCP. Identifying by Voice VLAN subnet means any phone in 10.80.1.0/24 (up to our max-pool 10 limit) can register during an outage without having to manually update MAC addresses (id mac) or static IPs (id ip) on the router whenever a phone is swapped.
4. Enable MGCP Fallback
Assuming your router’s analog/digital voice ports are normally controlled by CUCM via MGCP, include this command so the router takes local control of its voice ports when CUCM communication is lost:
ccm-manager fallback-mgcp
5. Configure Inbound & Outbound Call Routing (FXO PLAR & 911 Dial Peer)
Inbound (FXO PLAR):
With a single local POTS line on FXO port 0/1/0, we configure Private Line Automatic Ringdown (connection plar) so any incoming call on the analog line rings a designated main extension at the clinic:
voice-port 0/1/0
timing hookflash-out 50
timing guard-out 1000
connection plar 1000
description *** FXO POTS for 911 calls ***
cable-detect
Outbound (911 POTS Dial Peer):
To route outbound emergency calls over the local FXO port 0/1/0, we configure a static POTS dial peer:
dial-peer voice 911 pots
description *** Dial-Peer for 911 calls ***
destination-pattern 911$
port 0/1/0
forward-digits all
6. Verification & Troubleshooting Commands
When testing or troubleshooting phones in SIP SRST mode, use these IOS XE commands:
show sip-ua status registrar ! Shows which SIP phones and DNs are registered to SRST
show voice register pool 1 ! Shows registered SIP pools and dynamically created dial peers
debug voice register errors ! Troubleshoots SIP SRST registration errors
debug voice register events ! Troubleshoots SIP SRST registration events
debug voice dialpeers inout ! Verifies inbound/outbound dial-peer matching during SRST
Configurations in CUCM
Now that the branch router is configured for SIP SRST, we must tell our phones in CUCM where to send failover SIP registrations if their connection to CUCM drops:
- Navigate to System > SRST and click Add New.
- Enter a descriptive name, and place the IP address of your branch voice router (
10.80.1.1in our example) in both the IP Address field and the SIP Network/IP Address field (ports auto-fill to2000and5060). - Navigate to System > Device Pool and open the Device Pool for your branch facility.
- Set the SRST Reference dropdown to the SRST reference you just created.
- Click Save and Apply Config so the phones download the updated SRST IP address in their configuration file.

Jaron Davis
Unified Communications Architect specializing in enterprise Zoom, Microsoft Teams, Cisco UC modernization, SIP/VoIP infrastructure, and systems architecture.