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Cisco Unified Survivable Remote Site Telephony

Cisco Unified Survivable Remote Site Telephony Version 4.1

As the enterprise extends its IP telephony deployments from central sites to remote offices, one of the critical factors in achieving a successful deployment is the ability to support backup call control at the remote branch office. Cisco® Unified Survivable Remote Site Telephony (SRST) provides a cost-effective solution for supporting redundant call control in the remote branch office.

Cisco Unified Communications is a comprehensive IP communications system of voice, video, data, and mobility products and applications. It enables more effective, more secure, more personal communications that directly affect both sales and profitability. It brings people together by enabling a new way of communicating-where your business moves with you, security is everywhere, and information is always available...whenever and wherever it is needed. Cisco Unified Communications is part of an integrated solution that includes network infrastructure, security, mobility, network management products, lifecycle services, flexible deployment and outsourced management options, end-user and partner financing packages, and third-party communications applications.

Benefits of Centralized Call-Processing Architecture

Cisco Unified Survivable Remote Site Telephony is a critical component of a centralized call-processing architecture in which a Cisco Unified Communications Manager cluster, located at a central site, provides telephony services for all sites of a corporation. The architecture provides numerous benefits to enterprises, including centralized and simplified management. Table 1 lists the benefits of a centralized call-processing architecture.

Table 1. Benefits of Centralized Call-Processing Architecture

Centralized Call

Processing Features Benefits

Delivery of full feature set to remote branches, next-generation call centers, unified-messaging services, embedded directory services, and mobility

Improved productivity

Centralized configuration and management

Reduced operating expenses

Simplified maintenance and troubleshooting

Reduced operating expenses

Converged voice and data network

Reduced operating expenses

Reduced installation cost (shared Cisco Unified Communications Manager resource)

Reduced initial expense

However, centralized call-processing architecture must include a strategy for survivability of telephony service at the branch office when access to the centralized call processing is interrupted because of WAN outage or other factors. Call-processing redundancy in the branch office is particularly critical in an emergency-which may be the cause of a WAN outage in the first place.

Components of Centralized Call-Processing Architecture

The Cisco Unified Communications system uses Cisco Unified Communications Manager in combination with Cisco Unified Survivable Remote Site Telephony which is embedded within Cisco IOS® Software, to help provide high-availability IP telephony to branch offices. When access to Cisco Unified Communications Manager from the branch office is impeded, for example, as a result of a WAN link failure, Cisco Unified Survivable Remote Site Telephony provides telephony backup services to help ensure that the branch office has continuous telephony service over the Cisco network infrastructure deployed in the branch. The enhanced reliability provided by Cisco Unified Survivable Remote Site Telephony makes the Cisco Unified Communications system a cost-effective solution to ensure telephony operation to all users in an organization, whether they are located in the headquarters or in a branch office.
Furthermore, in certain environments, the security of telephony communication is a critical requirement. The Cisco Unified Communications system supports secure telephony communication between any two phones in the network, whether those phones are in the headquarters facility or at a branch office. Cisco Unified Survivable Remote Site Telephony contributes to this secure telephony communication solution by supporting the same secure telephony protocols in the branch office when the branch loses communication with the centralized Cisco Unified Communications Manager.

How It Works

Cisco developed Cisco Unified Survivable Remote Site Telephony technology for all Cisco IOS Software platforms that support voice (refer to Table 3 for a complete list). The Cisco Unified Survivable Remote Site Telephony feature integrates network intelligence into Cisco IOS Software, which acts as the call-processing engine for IP phones located in the branch office during a WAN outage (Figure 1).

Figure 1. Centralized Cisco Unified Communications Manager Deployment with Remote Site Experiencing WAN Failure; Cisco Router Using Cisco Unified Survivable Remote Site Telephony

Cisco Unified Survivable Remote Site Telephony functions in the branch-office router to automatically detect a failure in the network and initiate a process to autoconfigure the router, providing call-processing backup redundancy for the IP phones in that office and helping ensure that the telephony capabilities stay operational. Upon restoration of WAN connectivity, the system automatically shifts call processing back to the primary Cisco Unified Communications Manager cluster. The Cisco Unified Survivable Remote Site Telephony configuration needs to be completed only once during install, simplifying deployment, administration, and maintenance. No IT staff is required at the remote sites to manage the Cisco Unified Survivable Remote Site Telephony feature.
Cisco routers with Cisco Unified Survivable Remote Site Telephony also offer secure voice mode with Cisco Unified Survivable Remote Site Telephony 3.3 and higher. If you deploy secure voice with Cisco Unified Communications Manager at your main site, secure Cisco Unified Survivable Remote Site Telephony gives you the option to keep calls secure during Cisco Unified Survivable Remote Site Telephony mode with transparent layer security (TLS) and Secure Real-Time Transport Protocol (SRTP) for signaling and media encryption, respectively. When the WAN link or Cisco Unified Communications Manager is restored, Cisco Unified Communications Manager resumes secure call-handling capabilities.
Cisco Unified Survivable Remote Site Telephony Version 3.4 and higher supports Session Initiation Protocol (SIP) for Cisco Unified IP phones, which provide basic telephony functions when the network SIP proxy or Cisco Unified Communications Manager is no longer available. The Cisco Unified Survivable Remote Site Telephony router with SIP enabled provides SIP registrar services during the outage and supports a back-to-back user agent, allowing for supplementary features such as call transfer and forwarding. Cisco Unified IP phones using SIP register to the Cisco Unified Survivable Remote Site Telephony enabled router when the WAN link is out of service.
Cisco Unified Survivable Remote Site Telephony offers fault monitoring using Simple Network Management Protocol (SNMP) with the SRST Management Information Base (MIB), which gives you the ability to remotely monitor the Cisco Unified Survivable Remote Site Telephony site using existing SNMP tools or CiscoWorks. The CISCO-SRST-MIB provides the network operations center details about Cisco Unified Survivable Remote Site Telephony activity, including duration of SRST usage, IP phones registered or registration failure, and calls processed during Survivable Remote Site Telephony mode. A backup WAN link connection is required to receive CISCO-SRST-MIB data to the central site during Survivable Remote Site Telephony mode. Table 1 lists part numbers for Cisco Unified Survivable Remote Site Telephony. Cisco Unified Survivable Remote Site Telephony Platform Density and Feature License Part Numbers

Platform

Number of Phones Supported***

Part Number

Part Number (Spare)

Cisco 1760-V Modular Access Router and Cisco 2801 Integrated Services Router

Up to 24 phones

FL-SRST-SMALL

FL-SRST-SMALL=

Cisco 2600XM Multiservice Router and Cisco 2811 Integrated Services Router

Up to 36 phones

FL-SRST-36

FL-SRST-36=

Cisco 2650XM Multiservice Router and Cisco 2821 Integrated Services Router

Up to 48 phones

FL-SRST-MEDIUM

FL-SRST-MEDIUM=

Cisco 2851 Integrated Services Router

Up to 96 phones

FL-SRST-96

FL-SRST-96=

Cisco 3725 Multiservice Access Router

Up to 144 phones

FL-SRST-144

FL-SRST-144=

Cisco 3825 Integrated Services Router

Up to 336 phones

FL-SRST-336

FL-SRST-336=

Cisco 3745 Multiservice Access Router

Up to 480 phones

FL-SRST-480

FL-SRST-480=

Cisco uBR7200 Series NPE-400 and NPE-G1 Network Processing Engines**

Up to 480 phones

FL-SRST-480

FL-SRST-480=

Cisco 3845 Integrated Services Router and Cisco Catalyst®6500 Communications Media Module (CMM)

Up to 720 phones

FL-SRST-720

FL-SRST-720=

* The Cisco Catalyst 6500 Series CMM supports Cisco Unified Survivable Remote Site Telephony 4.0 with Cisco IOS Software Release 12.4 and supports Cisco Unified Survivable Remote Site Telephony 2.1 with Cisco IOS Software Release 12.2(13)ZC.
** The Cisco uBR7200 Series supports Cisco Unified Survivable Remote Site Telephony 2.1 only with Cisco IOS Software Release 12.3 Mainline.
*** The maximum number of SIP phones supported in Survivable Remote Site Telephony mode in this release is different as follows: Cisco 1751, Cisco 1760, Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM and Cisco 2801-up to 24 phones; Cisco 2650XM and Cisco 2811-up to 36 phones; Cisco 2821-up to 48 phones; Cisco 2691-up to 72 phones; Cisco 2851-up to 96 phones; Cisco 3725-up to 144 phones; Cisco 3825-up to 168 phones; Cisco 3745-up to 192 phones; and Cisco 3845-up to 480 phones.

Cisco Unified Survivable Remote Site Telephony Platform Information

Cisco platforms with Cisco Unified Survivable Remote Site Telephony support from 24 to 720 phones. Details about currently supported platforms and the number of phones per platform is provided in the Cisco Unified Survivable Remote Site Telephony Specifications Sheet for each version, which can be viewed online at: http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_documentation_roadmap09186a008018912f.html.
Cisco offers integrated services router bundles with Cisco Unified Survivable Remote Site Telephony at a discount when compared to purchasing bundle components separately. These bundles are listed in Table 2.

Table 2. Cisco Unified Survivable Remote Site Telephony Bundles

Bundle Part Number

Includes

CISCO3845-SRST/K9

Cisco 3845 voice bundle with packet voice digital signal processor (DSP) module (PVDM2-64), Cisco Unified SRST feature license for 240 phones, and Cisco IOS SP Services feature set

CISCO3825-SRST/K9

Cisco 3825 voice bundle with packet voice DSP module (PVDM2-64), Cisco Unified SRST feature license for 168 phones, and Cisco IOS SP Services feature set

CISCO2851-SRST/K9

Cisco 2851 voice bundle with packet voice DSP module (PVDM2-48), Cisco Unified SRST feature license for 96 phones, and Cisco IOS SP Services feature set

CISCO2821-SRST/K9

Cisco 2821 voice bundle with packet voice DSP module (PVDM2-32), Cisco Unified SRST feature license for 48 phones, and Cisco IOS SP Services feature set

CISCO2811-SRST/K9

Cisco 2811 voice bundle with packet voice DSP module (PVDM2-16), Cisco Unified SRST feature license for 36 phones, and Cisco IOS SP Services feature set

CISCO2801-SRST/K9

Cisco 2801 voice bundle with packet voice DSP module (PVDM2-8), Cisco Unified SRST feature license for 24 users, and Cisco IOS SP Services feature set

Cisco Unified IP Phone Support

Cisco Unified Survivable Remote Site Telephony is supported with Cisco Unified CallManager Version 3.01 and greater. Cisco Unified Survivable Remote Site Telephony is not dependent on Cisco Unified CallManager versions but on IP phone loads.
Table 3 lists the Cisco Unified IP phones supported by Cisco Unified Survivable Remote Site Telephony with Skinny Call Control Protocol (SCCP) phone loads.

Table 3. Cisco Unified IP Phone Support Using SCCP

Phone

Cisco Unified SRST 2.1

Cisco Unified SRST 3.3

Cisco Unified SRST 3.4

Cisco Unified SRST 4.0

Cisco Unified SRST 4.1

Cisco Unified IP Phone 7970G and 7971G-GE models

-

X

X

X

X

Cisco Unified IP Phone 7960G and 7940G models

X

X

X

X

X

Cisco Unified IP Phone 7961G, 7941G, 7961G-GE, and 7941G-GE models

-

X

X

X

X

Cisco Unified IP Conference Station 7935

X

X

X

X

X

Cisco Unified IP Conference Station 7936

-

X

X

X

X

Cisco Unified IP Phone 7912G

-

X

X

X

X

Cisco Unified IP Phone 7911G

-

-

-

X

X

Cisco Unified IP Phone 7905G

-

X

X

X

X

Cisco Unified IP Phone 7906G

-

-

-

-

X

Cisco Unified IP Phone 7902G

-

X

X

X

X

Cisco Unified Wireless IP Phone 7920

-

X

X

X

X

Cisco Unified Wireless IP Phone 7921G

-

-

-

 

X

Cisco Unified IP Phone Expansion Module 7914

X

X

X

X

X

Cisco VG248 48-Port Analog Phone Gateway

X

X

X

X

X

Cisco ATA 180 Series Analog Telephone Adaptors

-

-

-

X

X

Cisco IP Communicator

-

-

-

X

X

Cisco Unified Video Advantage

-

-

-

X

X

Table 4 lists the Cisco Unified IP phones supported by Cisco Unified Survivable Remote Site Telephony with Session Initiated Protocol (SIP) phone loads.

Table 4. Cisco Unified IP Phone Support Using SIP

Phone

Cisco Unified SRST 4.0

Cisco Unified SRST 4.1

Cisco Unified IP Phone 7970G and 7971G-GE models

X

X

Cisco Unified IP Phone 7960G and 7940G models

X

X

Cisco Unified IP Phone 7961G, 7941G, 7961G-GE, and 7941G-GE models

X

X

Cisco Unified IP Conference Station 7935

-

-

Cisco Unified IP Conference Station 7936

-

-

Cisco Unified IP Phone 7912G

X

X

Cisco Unified IP Phone 7906G

-

X

Cisco Unified IP Phone 7911G

X

X

Cisco Unified IP Phone 7905G

X

X

Cisco Unified IP Phone 7902G

-

-

Cisco Unified Wireless IP Phone 7920 and 7921G models

-

-

Cisco Unified IP Phone Expansion Module 7914

-

-

Cisco ATA 180 Series Analog Telephone Adaptors

-

-

Cisco IOS Software Image Support

Table 5 summarizes the correlation between Cisco Unified Survivable Remote Site Telephony version and Cisco IOS Software.
Secure Cisco Unified Survivable Remote Site Telephony is available with Cisco Unified Survivable Remote Site Telephony 3.3 and higher for Cisco Unified IP phones using SCCP and also requires Cisco Unified CallManager 4.1(2) and higher.
Cisco Unified Survivable Remote Site Telephony for SIP phones is supported with Cisco Unified Survivable Remote Site Telephony 3.4 and higher and only with Cisco Unified IP phones.
For the latest Cisco IOS Software release and features, consult the Feature Navigator at: http://www.cisco.com/go/fn.

Table 5. Cisco IOS Software Release(s)

Cisco Unified SRST Version

Cisco IOS Software Release(s)

Cisco Unified SRST 2.0

12.2(13)T

Cisco Unified SRST 2.1

12.2(15)T and 12.3 Mainline

Cisco Unified SRST 3.0

12.3(4)T

Cisco Unified SRST 3.1

12.3(8)T

Cisco Unified SRST 3.2

12.3(11)T

Cisco Unified SRST 3.3 Plus Secure SRST

12.3(14)T or 12.4 Mainline

Cisco Unified SRST 3.4

12.4(4)T

Cisco Unified SRST 4.0

12.4(9)T

Cisco Unified SRST 4.1

12.4(14)T

Supported Features

Cisco Unified Survivable Remote Site Telephony provides robust support for many IP phone features through the duration of the WAN failure, a feature that is not available from other traditional telephony solutions. Table 6 lists the features supported during failure.

Table 6. Cisco Unified Survivable Remote Site Telephony Features

Cisco Unified SRST Version

Feature Set

Cisco Unified Survivable Remote Site Telephony 2.0

• Support for IP and analog phones
• Rehoming of IP phones upon failure to branch router for call processing
• Maintenance of local extension-to-extension calls upon failure*
• Maintenance of extension-to-public switched telephone network (PSTN) calls upon failure
• Up to six lines per phone
• Call hold and pick up
• Speed and last-number redial
• Up to 24 line appearances per system
• Primary line support
• Maintenance of existing calls upon recovery
• Analog foreign exchange office (FXO) and foreign exchange station (FXS)
• Calling-party name
• Caller ID and asynchronous-network-interface (ANI) support
• WAN link support: Frame Relay, ATM, Multilink Point-to-Point Protocol (MLPPP), serial, ATM Adaption Layer 2 (AAL2), and DSL
• Class of restriction
• Music on hold (MOH), tone on hold, and music and tone on transfer (MOH for endpoint PSTN only)
• Distinctive ringing
• Direct inward dialing (DID) and direct outward dialing (DOD)
• PSTN T1 and E1 channel-associated-signaling (CAS) trunks support
• ISDN Basic Rate Interface (BRI) and Primary Rate Interface (PRI) support
• Call-detail recording and RADIUS server
• Interworking with Cisco Gatekeeper
• Transfer to voicemail pilot number using PSTN
• Alias lists for unregistered phones
• Translation rules support
• Tool Command Language (TCL)-based simple automated attendant and interactive voice response (IVR) on local gateways
• Transfer across H.323 network of Cisco endpoints

Cisco Unified Survivable Remote Site Telephony 2.1

• Cisco Unified CallManager phone language support
• Global call-forwarding enhancement
• In-band dual tone multifrequency (DTMF) voicemail integration
• Enhanced dial-plan pattern

Cisco Unified Survivable Remote Site Telephony 3.0

• E1-R2 signaling support
• Secondary dial tone
• Dual-line appearance per button
• Three-party G711 temporary conferencing
• Call transfer with consult
• MOH multicast from flash .au file in Cisco Unified CallManager mode
• Support for Cisco Unified IP Phone 7905
• European date formats
• Enhanced dialplan-pattern command
• Increased directory-number maximums
• Additional language options for IP phone
• Configurable system message
• Improved debugs for phones
• Symmetric SIP gateway-to-gateway DTMF relay
• Ringing timeout for phones
• Cisco SIP phone support of basic calls only

Cisco Unified Survivable Remote Site Telephony 3.1

• Support for Cisco Unified Wireless IP Phone 7920
• Support for Cisco Unified IP Conference Station 7935 or Cisco Unified IP Conference Station 7936

Cisco Unified Survivable Remote Site Telephony 3.2

• Enhancement to the alias command
• Enhancement to the cor command
• Enhancement to the pickup command
• Enhancement to the user-locale command
• Increased number of phones supported on the Cisco 3745 Multiservice Access Router
• MOH Multicast from live feed in Cisco Unified CallManager mode
• No timeout for call preservation*
• RFC 2833 DTMF relay support
• Translation profile support

Cisco Unified Survivable Remote Site Telephony 3.3

• Support for Cisco Unified IP Phone 7970G, 7971G-GE, 7961G, 7941G, 7961G-GE, 7941G-GE, and 7911G models
• Enhancement to the show ephone command (new Cisco Unified IP phone model keywords)

Secure Cisco Unified Survivable Remote Site Telephony 3.3 with Cisco Unified CallManager 4.1(2)

• Basic call
• Call transfer (consult and blind)
• Call forward (busy, no answer, and all)
• Shared line (IP phones)
• Hold and resume
• Hold and pickup
• Only secure calls between IP phones or Cisco Unified SRST router

Cisco Unified Survivable Remote Site Telephony 3.4

• Fault monitoring with SNMP CISCO-SRST-MIB including;
- Cisco Unified Survivable Remote Site Telephony state and duration
- Phone registration and failure
- Threshold un-registration
- Total calls handled during Cisco Unified Survivable Remote Site Telephony mode