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Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)

This chapter describes Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) and what it does. It also includes information about support for Cisco Unified IP Phones and Platforms, specifications, features, prerequisites, restrictions and where to find additional reference documents.

For the most up-to-date information about Cisco Unified IP Phone support, the maximum number of Cisco Unified IP Phones, the maximum number of directory numbers (DNs) or virtual voice ports, and memory requirements for Cisco Unified SRST and Cisco Unified SIP SRST, see Cisco Unified SRST Supported Firmware, Platforms, Memory, and Voice Products.

Cisco Unified SRST Feature Overview

This chapter describes Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) and what it does. It also includes information about support for Cisco Unified IP Phones and Platforms, specifications, features, prerequisites, restrictions and where to find additional reference documents.

For the most up-to-date information about Cisco Unified IP Phone support, the maximum number of Cisco Unified IP Phones, the maximum number of directory numbers (DNs) or virtual voice ports, and memory requirements for Cisco Unified SRST and Cisco Unified SIP SRST, see Cisco Unified SRST Supported Firmware, Platforms, Memory, and Voice Products.

Cisco Unified SCCP SRST

Information About SCCP SRST

Cisco Unified SRST provides Cisco Unified CM with fallback support for Cisco Unified IP phones that are attached to a Cisco router on your local network. Cisco Unified SRST enables routers to provide call-handling support for Cisco Unified IP phones when they lose connection to remote primary, secondary, or tertiary Cisco Unified CM installations or when the WAN connection is down.

Cisco Unified CM supports Cisco Unified IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to Cisco Unified SRST, when the WAN connection between a router and the Cisco Unified CM failed or when connectivity with Cisco Unified CM was lost for some reason, Cisco Unified IP phones on the network became unusable for the duration of the failure. Cisco Unified SRST overcomes this problem and ensures that the Cisco Unified IP phones offer continuous (although minimal) service by providing call-handling support for Cisco Unified IP phones directly from the Cisco Unified SRST router. The system automatically detects a failure and uses Simple Network Auto Provisioning (SNAP) technology to autoconfigure the branch office router to provide call processing for Cisco Unified IP phones that are registered with the router. When the WAN link or connection to the primary Cisco Unified CM is restored, call handling reverts back to the primary Cisco Unified CM.

When Cisco Unified IP phones lose contact with primary, secondary, and tertiary Cisco Unified CM, they must establish a connection to a local Cisco Unified SRST router to sustain the call-processing capability necessary to place and receive calls. The Cisco Unified IP phone retains the IP address of the local Cisco Unified SRST router as a default router in the Network Configuration area of the Settings menu. The Settings menu supports a maximum of five default router entries; however, Cisco Unified CM accommodates a maximum of three entries. When a secondary Cisco Unified CM is not available on the network, the local Cisco Unified SRST Router's IP address is retained as the standby connection for Cisco Unified CM during normal operation.


Note

Cisco Unified CM fallback mode telephone service is available only to those Cisco Unified IP phones that are supported by a Cisco Unified SRST router. Other Cisco Unified IP phones on the network remain out of service until they re-establish a connection with their primary, secondary, or tertiary Cisco Unified CM.


Typically, it takes three times the keepalive period for a phone to discover that its connection to Cisco Unified CM has failed. The default keepalive period is 30 seconds. If the phone has an active standby connection established with a Cisco Unified SRST router, the fallback process takes 10 to 20 seconds after connection with Cisco Unified CM is lost. An active standby connection to a Cisco Unified SRST router exists only if the phone has the location of a single Cisco Unified CM in its Unified Communications Manager list. Otherwise, the phone activates a standby connection to its secondary Cisco Unified CM.


Note

The time it takes for a Cisco Unified IP Phone to fallback to the SRST router can vary depending on the phone type. Phones such as the Cisco 7902, Cisco 7905, and Cisco 7912 can take approximately 2.5 minutes to fallback to SRST mode.


If a Cisco Unified IP phone has multiple Cisco Unified CM in its Cisco Unified CM list, it progresses through its list of secondary and tertiary Cisco Unified CM before attempting to connect with its local Cisco Unified SRST router. Therefore, the time that passes before the Cisco Unified IP phone eventually establishes a connection with the Cisco Unified SRST router increases with each attempt to contact to a Cisco Unified CM. Assuming that each attempt to connect to a Cisco Unified CM takes about 1 minute, the Cisco Unified IP phone in question could remain offline for 3 minutes or more following a WAN link failure.


Note

During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP phones display a message informing you that they are operating in Cisco Unified CM fallback mode. For example, the Cisco Unified IP Phone 7960G and Cisco Unified IP Phone 7940G display a "CM Fallback Service Operating" message, and the Cisco Unified IP Phone 7910 displays a "CM Fallback Service" message when operating in Cisco Unified CM fallback mode. When the Cisco Unified CM is restored, the message goes away and full Cisco Unified IP phone functionality is restored.


While in Cisco Unified CM fallback mode, Cisco Unified IP phones periodically attempt to re-establish a connection with Cisco Unified CM at the central office. Generally, the default time that Cisco Unified IP phones wait before attempting to re-establish a connection to a remote Cisco Unified CM is 120 seconds. The time can be changed in Cisco Unified CM; see the "Device Pool Configuration Settings" chapter in the Cisco Unified CM Administration Guide. A manual reboot can immediately reconnect Cisco Unified IP phones to Cisco Unified CM.

When a connection is re-established with Cisco Unified CM, Cisco Unified IP phones automatically cancel their registration with the Cisco Unified SRST Router. However, if a WAN link is unstable, Cisco Unified IP phones can bounce between Cisco Unified CM and Cisco Unified SRST. A Cisco Unified IP phone cannot re-establish a connection with the primary Cisco Unified CM at the central office if it is currently engaged in an active call.

Cisco Unified SRST supports the following call combinations:

  • SCCP phone to SCCP phone

  • SCCP phone to PSTN/router voice-port

  • SCCP phone to WAN VoIP using SIP or H.323

  • SIP phone to SIP phone

  • SIP phone to PSTN / router voice-port

  • SIP phone to Skinny Client Control Protocol (SCCP) phone

  • SIP phone to WAN VoIP using SIP

The figure shows a branch office with several Cisco Unified IP phones connected to a Cisco Unified SRST router. The router provides connections to both a WAN link and the PSTN. Typically, the Cisco Unified IP phones connect to their primary Cisco Unified Communications Manager at the central office via the WAN link. When the WAN connection is down, the Cisco Unified IP phones use the Cisco Unified SRST router as a fallback for their primary Cisco Unified Communications Manager. The branch office Cisco Unified IP phones are connected to the PSTN through the Cisco Unified SRST router and are able to make and receive off-net calls.

On H.323 gateways for SCCP SRST, when the WAN link fails, active calls from Cisco Unified IP phones to the PSTN are not maintained by default. Call preservation may work with the no h225 timeout keepalive command.

Under default configuration, the H.323 gateway maintains a keepalive signal with Cisco Unified Communications Manager and terminates H.323-to-PSTN calls if the keepalive signal fails, for example, if the WAN link fails. To disable this behavior and help preserve existing calls from local Cisco Unified IP phones, you can use the no h225 timeout keepalive command. Disabling the keepalive mechanism only affects calls that will be torn down as a result of the loss of the H.225 keepalive signal. For information regarding disconnecting a call when an inactive condition is detected, see the Media Inactive Call Detection document.

Prerequisites for Configuring Cisco Unified SCCP SRST

Before configuring Cisco Unified SRST, you must do the following:

  • An SRST feature license is required to enable the Cisco Unified SCCP SRST feature. Contact your account representative if you have further questions. For more information about Licensing on Unified SRST, refer Licensing.

  • You have an account on Cisco.com to download software.

    To obtain an account on Cisco.com, go to http://www.cisco.com and clickRegister at the top of the screen.

Installing Cisco Unified Communications Manager

When installing Cisco Unified Communications Manager, consider the following:

Installing Cisco Unified SCCP SRST

Installing Cisco Unified SRST V3.0 and Later Versions

Install the Cisco IOS software release image containing the Cisco SRST or Cisco Unified SRST version that is compatible with your Cisco Unified Communications Manager version. See the Cisco Unified Communications Manager Compatibility section. Cisco IOS software can be downloaded from the Cisco Software Center at http://www.cisco.com/public/sw-center/http://www.cisco.com/public/sw-center/.

Cisco SRST and Cisco Unified SRST can be configured to support continuous multicast output of music- on-hold (MOH) from a flash MOH file in flash memory. For more information, see the Defining XML API Schema section. If you plan to use MOH, go to the Technical Support Software Download site at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp and copy the music-on-hold.au file to the flash memory on your Cisco SRST or Cisco Unified SRST router.

Installing Cisco Unified SRST V2.0 and V2.1

Download and install Cisco SRST V2.0 or Cisco SRST V2.1 from the Cisco Software Center at http://www.cisco.com/public/sw-center/.

Installing Cisco Unified SRST V1.0

Cisco SRST V1.0 runs with Cisco Communications Manager V3.0.5 only. It is recommended that you upgrade to the latest Cisco Unified Communications Manager and Cisco Unified SRST versions.

Integrating Cisco Unified SCCP SRST with Cisco Unified Communications Manager

There are two procedures for integrating Cisco Unified SRST with Cisco Unified Communications Manager. Procedure selection depends on the Cisco Unified Communications Manager version that you have.

If You Have Cisco Communications Manager V3.3 or Later Versions

If you have Cisco Communications Manager V3.3 or later versions, you must create an SRST reference and apply it to a device pool. An SRST reference is the IP address of the Cisco Unified SRST Router.

  1. Create an SRST Reference

    • From any page in Cisco Unified Communications Manager, click System and SRST.

    • On the Find and List SRST References page, click Add a New SRST Reference.

    • On the SRST Reference Configuration page, enter a name in the SRST Reference Name field and the IP address of the Cisco SRST router in the IP Address field.

    • Click Insert.

  2. Apply the SRST reference or the default gateway to one or more device pools.

    • From any page in Cisco Unified Communications Manager, click System and Device Pool.

    • On the Device Pool Configuration page, click on the required device pool icon.

    • On the Device Pool Configuration page, choose an SRST reference or Use Default Gateway from the SRST Reference field's menu.

If You Have Cisco Unified Communications Manager Version Prior to V3.3

If you have firmware versions that enable Cisco Unified SRST by default, no additional configuration is required on Cisco Unified Communications Manager to support Cisco Unified SRST. If your firmware versions disable Cisco Unified SRST by default, you must enable Cisco Unified SRST for each phone configuration.

  1. Go to the Cisco Unified Communications Manager Phone Configuration page.

    • From any page in Cisco Unified Communications Manager, click Device and Phone.

    • In the Find and List Phones page, click Find.

    • After a list of phones appears, click on the required device name.

    • The Phone Configuration appears.

  2. In the Phone Configuration page, go to the Product Specific Configuration section at the end of the page, choose Enabled from the Cisco Unified SRST field’s menu, and click Update.

  3. Go to the Phone Configuration page for the next phone and choose Enabled from the Cisco Unified SRST field’s menu by repeating Step 1 and Step 2.

Restrictions for Configuring Cisco Unified SCCP SRST

The following table provides a history of restrictions from Cisco SCCP SRST Version 1.0 to the present version of Cisco Unified SCCP SRST.

Cisco Unified SRST Version

Cisco IOS Release

Restrictions

Version 4.1

12.4.(15)T

  • Enhanced 911 Services for Cisco Unified SRST does not interface with the Cisco Emergency Responder.

  • The information about the most recent phone that called 911 is not preserved after a reboot of Cisco Unified SRST.

  • Cisco Emergency Responder does not have access to any updates made to the emergency call history table when remote IP phones are in Cisco Unified SRST fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP phones register back to Cisco Unified Communications Manager, Cisco Emergency Responder will not have any history of those calls. As a result, those calls will not get routed to the original 911 caller. Instead, the calls are routed to the default destination that is configured on Cisco Emergency Responder for the corresponding ELIN.

  • For Cisco Unified Wireless IP Phone 7920 and 7921, a caller’s location can only be determined by the static information configured by the system administrator. For more information, see the Precautions for Mobile Phones in Configuring Enhanced 911 Services.

  • The extension numbers of 911 callers can be translated to only two emergency location identification numbers (ELINs) for each emergency response location (ERL).

  • Using ELINs for multiple purposes can result in unexpected interactions with existing Cisco Unified SRST features. These multiple uses of an ELIN can include configuring an ELIN for use as an actual phone number (ephone-dn, voice register dn, or FXS destination-pattern), a Call Pickup number, or an alias rerouting number. For more information, see the Multiple Usages of an ELIN in Configuring Enhanced 911 Services .

  • There are a number of other ways that your configuration of Enhanced 911 Services can interact with existing Cisco Unified SRST features and cause unexpected behavior. For a complete description of interactions between Enhanced 911 Services and existing Cisco Unified SRST features, see the Interactions with Existing Cisco Unified CME Features in Configuring Enhanced 911 Services.

Version 4.0

Version 3.4

Version 3.2

Version 3.1

Version 3.0

Version 2.1

Version 2.02

Version 2.01

Version 2.0

12.4(4)XC

12.4(4)T

12.3(11)T

12.3(7)T

12.2(15)ZJ

12.3(4)T

12.2(15)T

12.2(13)T

12.2(11)T

12.2(8)T1

12.2(8)T

12.2(2)XT

  • All of the restrictions in Cisco SRST Version 1.0.

  • Caller-id display on supported Cisco Unified IP phones: SIP phones in fallback mode displays the name and number of the caller. SCCP phones in fallback mode display only the caller-id number assigned to the line; the caller-ID name configuration for SCCP phones is not preserved during SRST fallback.

Call transfer is supported only on the following:

  • VoIP H.323, VoFR, and VoATM between Cisco gateways running Cisco IOS Release 12.2(11)T and using the H.323 nonstandard information element

  • FXO and FXS loop-start (analog)

  • FXO and FXS ground-start (analog)

  • Ear and mouth (E&M) (analog) and DID (analog)

  • T1 channel-associated signaling (CAS) with FXO and FXS ground-start signaling

  • T1 CAS with E&M signaling

  • All PRI and BRI switch types

The following Cisco Unified IP Phone function keys are dimmed because they are not supported during SRST operation:

  • MeetMe

  • GPickUp (group pickup)

  • Park

  • Confrn (conference)

  • Although the Cisco IAD2420 series integrated access devices (IADs) support the Cisco Unified SRST feature, this feature is not recommended as a solution for enterprise branch offices.

Version 1.0

12.2(2)XB

12.2(2)XG

12.1(5)YD

  • Does not support first generation Cisco Unified IP phones, such as Cisco IP Phone 30 VIP and Cisco IP Phone 12 SP+.

  • Does not support other Cisco Unified Communications Manager applications or services: Cisco IP SoftPhone, Cisco One: Voice and Unified Messaging Application, or Cisco IP Contact Center.

  • Does not support Centralized Automatic Message Accounting (CAMA) trunks on the Cisco 3660 routers.

Note 

If you are in one of the states in the United States of America where there is a regulatory requirement for CAMA trunks to interface to 911 emergency services, and you would like to connect more than 48 Cisco Unified IP phones to the Cisco 3660 multiservice routers in your network, contact your local Cisco account team for help in understanding and meeting the CAMA regulatory requirements.


Note

Voice VRF is not supported for SCCP SRST on Cisco Integrated Services Router Generation 2 (ISR G2).


Cisco Unified SIP SRST

Information About SIP SRST

This guide describes Cisco Unified SRST functionality for SIP networks. Cisco Unified SIP SRST provides backup to an external SIP call control (IP-PBX) by providing basic registrar and redirect server or back-to-back user agent (B2BUA) 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.

Cisco Unified SIP SRST can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across SIP networks in the same way as SCCP phones.

Cisco Unified SIP SRST supports the following call combinations:

  • SIP phone to SIP phone

  • SIP phone to PSTN / router voice-port

  • SIP phone to Skinny Client Control Protocol (SCCP) phone

  • SIP phone to WAN VoIP using SIP

SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These servers are usually located in the core of a VoIP network. If SIP phones located at remote sites at the edge of the VoIP network lose connectivity to the network core (because of a WAN outage), they may be unable to make or receive calls. Cisco Unified SIP SRST functionality on a SIP PSTN gateway provides service reliability for SIP-based IP phones in the event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP phones to continue to make and receive calls to and from the PSTN and also to make and receive calls to and from other SIP IP phones.

To see a branch office Cisco Unifed IP Phones connected to a remote central Cisco Unified CM Operating in SRST mode, see Figure Branch Office Cisco Unifed IP Phones Connected to a Remote Central Cisco Unified Communications Manage Operating in SRST Mode.


Note

Cisco Unity Express (CUE) interworking is not supported with secure SIP SRST.


Prerequisites for Configuring Cisco Unified SIP SRST

Before configuring Cisco Unified SIP SRST, you must do the following:

An SRST feature license is required to enable the Cisco Unified SIP SRST feature. Contact your account representative if you have further questions. For more information about Licensing on Unified SRST, refer to Licensing section in Cisco Unified SIP SRST on Cisco 4000 Series Integrated Services Router chapter.

Restrictions for Configuring Cisco Unified SIP SRST

The following table provides a history of restrictions from Cisco SIP SRST Version 3.0 to the present version of Cisco Unified SIP SRST.

Cisco Unified SRST Version

Cisco IOS Release

Restrictions

Version 8.0

15.1(1)T

SIP phones may be configured on the Cisco Unified CM with an Authenticated device security mode. The Cisco Unified CM ensures integrity and authentication for the phone using a TLS connection with NULL-SHA cipher for signaling. If such an Authenticated SIP phone fails over to the Cisco Unified SRST device, and if the Cisco Unified CM and SRST device are configured to support secure SIP SRST, it will register using TCP instead of TLS/TCP, thus disabling the Authenticated mode until the phone fails back to the Cisco Unified CM.

Version 4.1

12.4.(15)T

  • Cisco Unified SRST does not support BLF speed-dial notification, call forward all synchronization, dial plans, directory services, or music-on-hold (MOH).

  • Prior to SIP phone load 8.0, SIP phones maintained dual registration with both Cisco Unified Communications Manager and Cisco Unified SRST simultaneously. In SIP phone load 8.0 and later versions, SIP phones use keepalive to maintain a connection with Cisco Unified SRST during active registration with Cisco Unified Communications Manager. Every two minutes, a SIP phone sends a keepalive message to Cisco Unified SRST. Cisco Unified SRST responds to this keepalive with a 404 message. This process repeats until fallback to Cisco Unified SRST occurs. After fallback, SIP phones send a keepalive message every two minutes to Cisco Unified Communications Manager while the phones are registered with Cisco Unified SRST. Cisco Unified SRST continues to support dual registration for SIP phone loads older than 8.0.

  • Enhanced 911 Services for Cisco Unified SRST does not interface with the Cisco Emergency Responder.

  • The information about the most recent phone that called 911 is not preserved after a reboot of Cisco Unified SRST.

  • Cisco Emergency Responder does not have access to any updates made to the emergency call history table when remote IP Phones are in Cisco Unified SRST fallback mode. Therefore, if the PSAP calls back after the Cisco Unified IP Phones register back to Cisco Unified Communications Manager, Cisco Emergency Responder will not have any history of those calls. As a result, those calls will not get routed to the original 911 caller. Instead, the calls are routed to the default destination that is configured on Cisco Emergency Responder for the corresponding ELIN.

  • For Cisco Unified Wireless 7920 and 7921 IP Phones, a caller’s location can only be determined by the static information configured by the system administrator. For more information, see Precautions for Mobile Phones in Configuring Enhanced 911 Services.

  • The extension numbers of 911 callers can be translated to only two emergency location identification numbers (ELINs) for each emergency response location (ERL).

  • Using ELINs for multiple purposes can result in unexpected interactions with existing Cisco Unified SRST features. These multiple uses of an ELIN can include configuring an ELIN for use as an actual phone number (ephone-dn, voice register dn, or FXS destination-pattern), a Call Pickup number, or an alias rerouting number. For more information, see Multiple Usages of an ELIN in Configuring Enhanced 911 Services.

  • There are a number of other ways that your configuration of Enhanced 911 Services can interact with existing Cisco Unified SRST features and cause unexpected behavior. For a complete description of interactions between Enhanced 911 Services and existing Cisco Unified SRST features, see the Interactions with Existing Cisco Unified CME Features in Configuring Enhanced 911 Services.

Version 4.0

Version 3.4

Version 3.2

Version 3.1

Version 3.0

12.4(4)XC

12.4(4)T

12.3(11)T

12.3(7)T

12.2(15)ZJ 12.3(4)T

Not Supported

  • MOH is not supported for a call hold invoked from a SIP phone. A caller hears only silence when placed on hold by a SIP phone.

  • As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming call screening, paging, SIP presence, call park, call pickup, and SIP location are not supported.

  • SIP-NAT is not supported.

  • Cisco Unity Express is not supported.

  • Transcoding is not supported.

Phone Features

  • For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be configured with the G.711 codec.

Note 

Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco Analog Telephone Adaptor (ATA) 186 are not capable of dual registration; thus they are not supported and have limited functionality with Cisco Unified SIP SRST.

General

  • Call detail records (CDRs) are only supported by standard IOS RADIUS support; CDRs are not supported otherwise.

  • All calls must use the same codec, either G.729r8 or G.711.

  • Calls that have been transferred cannot be transferred a second time.

  • URL dialing is not supported. Only number dialing is supported.

  • The SIP registrar functionality provided by Cisco Unified SIP SRST provides no security or authentication services.

  • SIP IP phones that do not support dual concurrent registration with both their primary and their backup SIP proxy or registrar may be unable to receive incoming calls from the Cisco Unified SIP SRST gateway during a WAN outage. These phones may take a significant amount of time to discover that their primary SIP proxy or registrar is unreachable before they initiate a fallback registration to their backup proxy or registrar (the SIP SRST gateway).

  • SIP-phone-to-SIP-trunk support requires Refer and 302/300 Redirection to be supported by the SIP trunk (Version 3.0).

Interface Support for Cisco Unified Communications Manager Express and Cisco Unified SRST

Cisco Unified Communications Manager Express and Cisco Unified SRST routers have multiple interfaces and is used for signaling and data packet transfers. The two types of interfaces available on a Cisco router include the physical interface and the virtual interface. The types of physical interfaces available on a router depend on its interface processors or port adapters. Virtual interfaces are software-based interfaces that you create in the memory of the networking device using Cisco IOS commands. To configure a virtual interface for connectivity, use the Loopback Interface for Cisco Unified Communications Manager Express and Cisco Unified SRST.

Cisco Unified Communications Manager Express and Cisco Unified SRST supports the following interfaces:

  • Gigabit Ethernet Interface (IEEE 802.3z) (interface gigabitethernet)

  • Loopback Interface (interface loopback)

  • Fast Ethernet Interface (interface fastethernet)

MGCP Gateways and SRST

MGCP fallback is a different feature than SRST and, when configured as an individual feature, can be used by a PSTN gateway. To use SRST as your fallback mode on an MGCP gateway, SRST and MGCP fallback must both be configured on the same gateway. MGCP and SRST have had the capability to be configured on the same gateway since Cisco IOS Release 12.2(11)T.

To make outbound calls while in SRST mode on your MGCP gateway, two fallback commands must be configured on the MGCP gateway. These two commands allow SRST to assume control over the voice port and over call processing on the MGCP gateway. With Cisco IOS earlier than 12.3(14)T, the two commands are the ccm-manager fallback-mgcp and call application alternatecommands. With Cisco IOS releases after 12.3(14)T, the ccm-manager fallback-mgcp and service commands must be configured. A complete configuration for these commands is shown in the section the Enabling Cisco Unified SRST on an MGCP Gateway section.


Note

The commands listed above are ineffective unless both commands are configured. For instance, your configuration will not work if you only configure the ccm-manager fallback-mgcp command.


For more information on the fallback methods for MGCP gateways, see Configuring MGCP Gateway Support for Cisco Unified Communications Manager document or the MGCP Gateway Fallback Transition to Default H.323 Session Application document.

IPv6 Support for Unified SRST SIP IP Phones

IPv6 Support for Unified SRST SIP IP Phones

Internet Protocol version 6 (IPv6) is the latest version of the Internet Protocol (IP). IPv6 uses packets to exchange data, voice, and video traffic over digital networks. Also, IPv6 increases the number of network address bits from 32 bits in IPv4 to 128 bits. From Unified SRST Release 12.0 onwards, Unified SRST supports IPv6 protocols for SIP IP phones.

IPv6 support in Unified SRST allows the network to behave transparently in a dual-stack (IPv4 and IPv6) environment and provides additional IP address space to SIP IP phones that are connected to the network. If you do not have a dual-stack configuration, configure the CLI command call service stop under voice service voip configuration mode before changing to dual-stack mode. For an example of switching to dual-stack mode, see Examples for Configuring IPv6 Pools for SIP IP Phones.

The Cisco IP Phone 7800 Series and 8800 Series are supported on IPv6 for Unified SRST.

For more information on configuring SIP IP phones for IPv6 source address, see Configure IPv6 Pools for SIP IP Phones.

For an example of configuring IPv6 Support on Unified SRST, see Examples for Configuring IPv6 Pools for SIP IP Phones.

For more details about IPv6 deployment, see IPv6 Deployment Guide for Cisco Collaboration Systems Release 12.0.

Feature Support for IPv6 in Unified SRST SIP IP Phones

The basic feature supported for a IPv6 WAN down scenario is:

Basic SIP Line (IPv4 or IPv6) to SIP Line calls (IPv4 or IPv6) when Unified SRST is in dual-stack no anat mode.

The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:

  • Hold/Resume

  • Call Forward

  • Call Transfer

  • Three-way Conference (with BIB conferencing only)

  • Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features

  • Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough

Restrictions

The following are the known restrictions for IPv6 support on Unified SRST:

  • SIP Trunks are not supported on Unified SRST for IPv6 deployment. PSTN calls are supported only through T1/E1 trunks.

  • SCCP IP Phones are not supported in a deployment of IPv6 for Unified SRST.

  • SIP Phones can be either in IPv4 only or IPv6 only mode (no anat).

  • Trancoding and Transrating are not supported.

  • H.323 trunks are not supported.

  • Secure SIP lines or trunks are not supported.

  • IPv6 on Unified SRST is not supported on the Cisco IOS platform. The support is restricted to Cisco IOS XE platform with Cisco IOS Release 16.6.1 or later versions.

Configure IPv6 Pools for SIP IP Phones

Before you begin

  • Unified SRST 12.0 or a later version.

  • IPv6 option only appears if protocol mode is dual-stack configured under sip-ua configuration mode or IPv6.

  • Cisco Unified SRST License must be configured for the gateway to function as a Unified SRST gateway to support IPv6 functionality. For more information on licenses, see Licensing.

  • Cisco Unified Communications Manager (Unified Communications Manager) is provisioned with the IPv6 address of Unified SRST. For information on configuration of Unified SRST on Unified Communications Manager, see Survivable Remote Site Telephony Configurationin Cisco Unified Communications Manager Administration Guide.

SUMMARY STEPS

  1. enable
  2. configure terminal
  3. ipv6 unicast-routing
  4. voice service voip
  5. sip
  6. no anat
  7. call service stop
  8. exit
  9. exit
  10. sip-ua
  11. protocol modeipv4ipv6dual-stackpreferenceipv4ipv6
  12. exit
  13. voice servicevoip
  14. sip
  15. no call service stop
  16. exit
  17. voice register global
  18. default mode
  19. max-dn
  20. max-pool
  21. exit
  22. voice register pool
  23. idnetworkmaskip address maskmac
  24. end

DETAILED STEPS

 Command or ActionPurpose
Step 1

enable

Example:

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

Example:

Enters global configuration mode.

Step 3

ipv6 unicast-routing

Example:

Enables the forwarding of IPv6 unicast datagrams.

Step 4

voice service voip

Example:

Enters voice-service configuration mode to specify a voice encapsulation type.

  • voip — Specifies Voice over IP (VoIP) parameters.

Step 5

sip

Example:

Enters SIP configuration mode.

Step 6

no anat

Example:

Disables Alternative Network Address Types (ANAT) on a SIP trunk.

Step 7

call service stop

Example:

Shuts down SIP call service.

Step 8

exit

Example:

Exits SIP configuration mode.

Step 9

exit

Example:

Exits voice service voip configuration mode.

Step 10

sip-ua

Example:

Enters SIP user-agent configuration mode.

Step 11

protocol modeipv4ipv6dual-stackpreferenceipv4ipv6

Example:

Allows phones to interact with phones on IPv6 voice gateways. You can configure phones for IPv4 addresses, IPv6 address es, or for a dual-stack mode.

  • ipv4—Allows you to set the protocol mode as an IPv4 address.

  • ipv6—Allows you to set the protocol mode as an IPv6 address.

  • dual-stack—Allows you to set the protocol mode for both IPv4 and IPv6 addresses.

  • preference—Allows you to choose a preferred IP address family if protocol mode is dual-stack.

Step 12

exit

Example:

Exits SIP configuration mode.

Step 13

voice servicevoip

Example:

Enters voice-service configuration mode to specify a voice encapsulation type.

  • voip — Specifies Voice over IP (VoIP) parameters.

Step 14

sip

Example:

Enters SIP configuration mode.

Step 15

no call service stop

Example:

Activates SIP call service.

Step 16

exit

Example:

Exits SIP configuration mode.

Step 17

voice register global

Example:

Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME.

Step 18

default mode

Example:

Enables mode for provisioning SIP phones in Unified SRST. The default mode is Unified SRST itself.

Step 19

max-dn

Example:

Limits number of directory numbers to be supported by this router.

Maximum number is platform and version-specific. Type ? for value.

Step 20

max-pool

Example:

Sets maximum number of SIP phones to be supported by the Unified SRST router.

Step 21

exit

Example:

Exits voice register global configuration mode.

Step 22

voice register pool

Example:

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

Step 23

idnetworkmaskip address maskmac

Example:

Explicitly identifies a locally available individual SIP phone to support a degree of authentication.

Step 24

end

Example:

Exits to privileged EXEC mode.

Examples for Configuring IPv6 Pools for SIP IP Phones

The following example provides configuration of IPv6 pools for SIP IP Phones:

The following example provides interface configuration for IPv6 supported on Unified SRST:

The following example provides IP route configuration for IPv6 supported on Unified SRST:

The following example displays output when SIP call service is shut down with the call service stop CLI command:

The following example displays output when SIP call service is active with the nocall service stop CLI command:

Support for Cisco Unified IP Phones and Platforms

Support for Cisco Unified IP Phones and Platforms

The following sections provide information about Cisco Feature Navigator and the histories of Cisco Unified IP Phone, platform, and Cisco Unified CM support from Cisco SRST Version 1.0 to the present version of Cisco Unified SRST.

Unified SRST is supported on Cisco 1100 Series Integrated Services Router (ISR) platforms with Cisco IOS XE Amsterdam 17.3.2 and later releases.

Unified SRST is supported on Cisco 4000 ( 4321, 4331, 4351, 4431, 4451, and 4461) ISR Series Platforms with Cisco IOS XE Amsterdam 17.3.2 and later releases:

Unified SRST is supported on Cisco Catalyst 8000 Series Edge Platforms as following:

  • Cisco Catalyst 8300 Series Edge Platforms—Cisco IOS XE Amsterdam 17.3.2 and later releases

  • Cisco Catalyst 8200 Series Edge Platforms—Cisco IOS XE Bengaluru 17.4.1a and later releases

  • Cisco Catalyst 8200L Series Edge Platforms—Cisco IOS XE Bengaluru 17.5.1a and later releases

Finding Cisco IOS Software Releases That Support Cisco Unified SRST


Note

With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones regardless of whether these are SIP or SCCP


To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

See Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix for related compatibility information.

Cisco Unified IP Phone Support

For the most up-to-date information about Cisco Unified IP Phone support, see Compatibility Information.

For ATAs that are registered to a Cisco Unified SRST system to participate in FAX calls, they must have their ConnectMode parameter set to use the "standard payload type 0/8" as the RTP payload type in FAX passthrough mode. For ATAs used with Cisco Unified SRST 4.0 and higher versions, this is done by setting bit 2 of the ConnectMode parameter to 1 on the ATA. For more information, see the Parameters and Defaults chapter in Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator's Guide for SCCP.

Sours: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/srst_overview.html

Cisco Unified Communications Manager Express System Administrator Guide

This feature enables routers to provide call-handling support for Cisco Unified IP phones if they lose connection to remote primary, secondary, or tertiary Cisco Unified Communications Manager installations or if the WAN connection is down. When Cisco Unified SRST functionality is provided by Cisco Unified CME, provisioning of phones is automatic and most Cisco Unified CME features are available to the phones during periods of fallback, including hunt-groups, call park and access to Cisco Unity voice messaging services using SCCP protocol. The benefit is that Cisco Unified Communications Manager users will gain access to more features during fallback without any additional licensing costs.

This feature offers a limited telephony feature set during fallback mode. Customers who require the following features should continue to use Cisco Unified SRST, because these features are not supported with SRST fallback support using Cisco Unified CME.

  • More than 240 phones during fallback service

  • Cisco VG 248 Analog Phone Gateway support

  • Secure voice fallback during SRST fallback service

  • Simple, one-time configuration for SRST fallback service

Cisco Unified Communications Manager supports Cisco Unified IP phones at remote sites attached to Cisco Integrated Services Routers across the WAN. This new feature combines the many features available in Cisco Unified CME with the ability to automatically detect IP phone configurations that is available in Cisco Unified SRST to provide seamless call handling when communication with the Cisco Unified Communications Manager is interrupted.

When the system automatically detects a failure, Cisco Unified SRST uses Simple Network Auto Provisioning (SNAP) technology to auto-configure a branch office router to provide call processing for the Cisco Unified IP phones that are registered with the router. When the WAN link or connection to the primary Cisco Unified Communications Manager is restored, call handling returns to the primary Cisco Unified Communications Manager.

A limited number of phone features are automatically detected at the time that call processing falls back to Cisco Unified CME in SRST Fallback Mode, and an advantage of SRST fallback support using Cisco Unified CME is that you can choose to prebuild a Cisco Unified CME configuration that contains a number of extensions (ephone-dns) with additional features that you want them to have for some or all of your extensions. The configurations will contain ephone-dn configurations but will not identify which phones (which MAC addresses) will be associated with which ephone-dns (extension numbers).

By copying and pasting a prebuilt configuration onto Cisco Unified CME routers at several locations, you can use the same overall configuration for sites that are identically laid out. For example, if you have a number of retail stores, each with five to ten checkout registers, you can use the same overall configuration in each store. You might use a range of extensions from 1101 to 1110. Stores with fewer than ten registers will simply not use some of the ephone-dn entries you provide in the configuration. Stores with more extensions than you have prebuilt will use the auto-provisioning feature to populate their extra phones. The only configuration variations from store to store will be the specific MAC addresses of the individual phones, which are added to the configurations at the time of fallback.

When a phone registers for SRST service with a Cisco Unified CME router and the router discovers that the phone was configured with a specific extension number, the router searches for an existing prebuilt ephone-dn with that extension number and then assigns that ephone-dn number to the phone. If there is no prebuilt ephone-dn with that extension number, the Cisco Unified CME system automatically creates one. In this way, extensions without prebuilt configurations are automatically populated with extension numbers and features as the numbers and features are “learned” by the Cisco Unified CME router in SRST mode when the phone registers to the router after a WAN link fails.

The SRST fallback support using Cisco Unified CME feature is able to interrogate phones to learn their MAC addresses and the extension-to-ephone relationships associated with each phone. This information is used to dynamically create and execute the Cisco Unified CME button command for each phone and automatically provision each phone with the extensions and features you want it to have.

The following sequence describes how Cisco Unified CME provides SRST services for Cisco Unified Communications Manager phones when they lose connectivity with the Cisco Unified Communications Manager and fall back to the Cisco Unified CME router in SRST mode:

Before Fallback

  1. Phones are configured as usual in Cisco Unified Communications Manager.

  2. The IP address of the Cisco Unified CME router is registered as the SRST reference on the Cisco Unified Communications Manager device pool.

  3. SRST mode is enabled on the Cisco Unified CME router.

  4. (Optional) Ephone-dns and features are prebuilt on the Cisco Unified CME router.

During Fallback

  1. Phones that are enabled for fallback register to the default Cisco Unified CME router that has SRST mode enabled. Each display-enabled IP phone displays the message that has been defined using the system message command under telephony-service configuration mode. By default, this message is .”

  2. While the fallback phones are registering, the router in SRST mode initiates an interrogation of the phones in order to learn their phone and extension configurations. The following information is acquired or “learned” by the router:
    • MAC address

    • Number of lines or buttons

    • Ephone-dn-to-button relationship

    • Speed-dial numbers

  3. The option defined with the srst mode auto-provision command determines whether Cisco Unified CME adds the learned phone and extension information to its running configuration. If the information is added, it appears in the output when you use the show running-config command and is saved to NVRAM when you use the write command.

    • Use the srst mode auto-provision none command to enable the Cisco Unified CME router to provide SRST fallback services for Cisco Unified Communications Manager.

    • If you use the srst mode auto-provisiondn or srst mode auto-provision all commands, the Cisco Unified CME router includes the phone configuration it learns from Cisco Unified Communications Manager in its running configuration. If you then save the configuration, the fallback phones are treated as locally configured phones on the Cisco Unified CME-SRST router which could adversely impact the fallback behavior of those phones.

  4. While in fallback mode, Cisco Unified IP phones periodically attempt to reestablish a connection with Cisco Unified Communications Manager every 120 seconds (default). To manually reestablish a connection to Cisco Unified Communications Manager you can reboot the Cisco Unified IP phone.

  5. When a connection is reestablished with Cisco Unified Communications Manager, Cisco Unified IP phones automatically cancel their registration with the Cisco Unified CME router in SRST mode. However, if a WAN link is unstable, Cisco Unified IP phones can bounce between Cisco Unified Communications Manager and the Cisco Unified CME router in SRST mode.

An IP phone connected to the Cisco Unified CME-SRST router over a WAN reconnects itself to Cisco Unified Communications Manager as soon as it can establish a connection to Cisco Unified Communications Manager over the WAN link. However, if the WAN link is unstable, the IP phone switches back and forth between Cisco Unified CME-SRST and Cisco Unified Communications Manager, causing temporary loss of phone service (no dial tone). These reconnect attempts, known as WAN link flapping issues, continue until the IP phone successfully reconnects itself back to Cisco Unified Communications Manager.

WAN link disruptions can be classified into two types: infrequent random outages that occur on an otherwise stable WAN, and sporadic, frequent disruptions that last a few minutes.

To resolve WAN-link flapping issues between Cisco Unified Communications Manager and SRST, Cisco Unified Communications Manager provides an enterprise parameter and a setting in the Device Pool Configuration window called Connection Monitor Duration. (Depending on system requirements, the administrator decides which parameter to use.) The value of the parameter is delivered to the IP phone in the XML configuration file.

  • Use the enterprise parameter to change the connection duration monitor value for all IP phones in the Cisco Unified Communications Manager cluster. The default for the enterprise parameter is 120 seconds.

  • Use the Device Pool Configuration window to change the connection duration monitor value for all IP phones in a specific device pool.

A Cisco Unified IP phone will not reestablish a connection with the primary Cisco Unified Communications Manager at the central office if it is engaged in an active call.

After the First Fallback

Additional features can be set up, such as ephone hunt groups, which can contain learned extensions and prebuilt extensions. The complete core set of Cisco Unified CME phone features is available to the IP phones and extensions, whether they are learned or configured.

Figure 51-1 shows a branch office with several Cisco Unified IP phones connected to a Cisco Unified CME router in SRST fallback mode. The router provides connections to both a WAN link and the PSTN. The Cisco Unified IP phones connect to their primary Cisco Unified Communications Manager at the central office via this WAN link. Cisco Unified CME provides SRST services for the phones when connectivity over the WAN link is interrupted.

Sours: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmeadm/cmesrst.html
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Force Cisco IP Phones to Fall into SRST Mode

Force Cisco IP Phones to Fall into SRST Mode

byAvinash KarnaniinCUCM

Are you planning to do SRST Testing to ensure that Cisco IP Phones falls back into SRST Mode when there is a WAN Link failure? It is not recommended to plug out the WAN Link or Turn off Cisco Unified Communications Manager servers or services. The best recommended practice is to do this testing is to apply ACL (Access Control List) on the WAN interface of the router.

What needs to be blocked?

Communication Protocol

  • SCCP > Port Number 2000 (TCP)
  • Secure SCCP >Port Number 2443 (TCP)
  • SIP > Port Number 5060 (TCP/UDP),
  • Secure SIP >Port Number 5061 (TCP/UDP)

Real Time Protocol (RTP)

  • Standard RTP: Port Numbers between 16384-32767 (UDP)

So you have to block Communication Protocol as well as Real Time Protocol.

What commands to be applied?

Access your WAN Router and configure the following ACL commands.

ip access-list extended SRST-ACL    —->>>> Extended ACL Name

deny tcp any any eq 5060     —->>>> Used by SIP
deny udp any any eq 5060    —->>>> Used by SIP
deny tcp any any eq 5061    —->>>> Used by Secure SIP
deny udp any any eq 5061    —->>>> Used by Secure SIP
deny tcp any any eq 2000    —->>>> Used by SCCP

deny tcp any any eq 2443    —->>>> Used by Secure SCCP
deny udp any any range 16384 32767    —->>>> Used by RTP
permit ip any any    —->>>> Allowing all other traffic except the above

Now apply the above ACL Name on the WAN Interface to block the services

interface 1/1—->>>> Replace 1/1 by your WAN interface card identification
ip access-group SRST-ACL in     —->>>> Apply the Extended ACL that was created in the above steps. Until and unless this command is applied, ACL is not effective.

Now Cisco IP Phones should fall back Cisco Unified Communication Manager to SRST Mode in few seconds .

Once the testing is completed, remove the ACL entries that was created above by placing “no” command. Following is an example –

no ip access-list extended SRST-ACL

interface 1/1

no ip access-group SRST-ACL in

That’s all !

Hope this helps!

Published by Team UC Collabing

Avinash Karnani

I am working in an IT company and having 10+ years of experience into Cisco IP Telephony and Contact Center. I have worked on products like CUCM, CUC, UCCX, CME/CUE, IM&P, Voice Gateways, VG224, Gatekeepers, Attendant Console, Expressway, Mediasense, Asterisk etc. I am not an expert but i keep exploring whenever and wherever i can and share whatever i know. You can visit my LinkedIn profile by clicking on the icon below.

“Everyone you will ever meet knows something you don’t.” ― Bill Nye

Tagged with:CUCM, Fallback, SRST

Sours: https://www.uccollabing.com/force-cisco-ip-phones-to-fall-into-srst-mode/
CUCM features: what is Mgcp Fallback and SRST

This page details the process to configure SRST on Cisco IOS gateways (SIP, SCCP, and MGCP*) this is a must in a multisite implementation.

Process[]

Step 1: Configure SRST configuration on IOS gateway

Step 2: Configure Call manager device pool's SRST reference

Step 3: Configure dial peers on IOS gateway (optional - kinda)

Detailed Steps[]

Step 1: Enter Call-manager-fallback configuration mode to activate SRST

Router# Config T
Router(config)#Call-manager-fallback

Step 2: Define the IP address and port to which the SRST service binds

Router(config-cm-fallback)# ip source-address ip-address [portport-number] [any-match| strict-match]

Step 3: Define the maximum number of directory numbers (DN) to support

Router(config-cm-fallback)# Max-dnnumber-of-maximum-directory-numbers

Step 4: Define the maximum number of ip phones to support

Router(config-cm-fallback)# Max-ephonesmax-phones

Step 5: define the maximum number of numbers allowed per phone type

Router(config-cm-fallback)# Limit-dnphone-type

Step 6: define the phone keepalive interval

Router(config-cm-fallback)# KeepaliveSeconds

Example Configuration

Router# Config T

Router(config)#Call-manager-fallback

Router(config-cm-fallback)# ip source-address 10.10.1.21 port 2000

Router(config-cm-fallback)# Max-dn 150

Router(config-cm-fallback)# Max-ephones 120

Router(config-cm-fallback)# Limit-dn 7841 100

Router(config-cm-fallback)# keepalive 30

Configure Cucm reference[]

Step 1: log into your CUCM Publisher then go to System > SRST

Step 2: Click on Add New

Step 3: Enter a Name, Port (2000 is the Default), IP address, SIP Port, Click Save

Step 4: Go to System > Device Pool

Step 5: click on the device pool related to the site that you are working on.

Step 6: Locate SRST in the list of settings, Select the reference that you just added to the system.

Configure Dial Peers on IOS gateway.

this is a complex subject please refer to the Configure IOS Dial-peers article

Sours: https://telephonynetworking.fandom.com/wiki/Configure_SRST_(Fallback)

Srst mode cisco

What is SRST mode?

SRST. Survivable Remote Site Telephony (SRST) is a Cisco Unified Communications Manager (CUCM) call processing backup mechanism that allows Cisco IP phones to register to a Cisco router.

Click to see full answer.


Also question is, how does Cisco SRST work?

SRST Function of Switchover SignalingWhen the WAN link fails, the IP Phones lose contact with the central CUCM but then register with the local Cisco Unified SRST gateway. The Cisco Unified SRST gateway detects newly registered IP Phones, queries these IP Phones for their configuration, and then autoconfigures itself.

Also, what is Call Manager fallback? MGCP fallback enables a gateway to act as local call control when the CUCM server to which the remote site phones and gateway register is offline or WAN connectivity is lost (in which case, Cisco Unified SRST kicks in and offers call control functionality).

Similarly, you may ask, how long is the default keepalive period for Srst in Cisco IOS?

The default keepalive period is 30 seconds. If the phone has an active standby connection established with a Cisco Unified SRST router, the fallback process takes 10 to 20 seconds after connection with Cisco Unified CM is lost.

What is preservation mode Cisco?

Call preservation (aka call survivability) is a feature that allows active calls using a gateway to be preserved during a CUCM outage. Each gateway and trunk configuration device discussed in this post has a device pool with a Call Manager group configuration.

Sours: https://askinglot.com/what-is-srst-mode
SRST

Implementing Cisco Unified Communications Manager, Part 2 (CCNP Voice): Examining Remote-Site Redundancy Options

Basic Cisco Unified SRST Usage

Cisco Unified SRST provides CUCM with fallback support for Cisco Unified IP Phones that are attached to a Cisco router on a local network.

Cisco Unified SRST enables routers to provide basic call-handling support for Cisco Unified IP Phones when they lose connection to remote primary, secondary, and tertiary CUCM servers or when the WAN connection is down.

Cisco Unified SIP SRST Usage

Cisco Unified SIP SRST provides backup to an external SIP proxy server by providing basic registrar and redirect server services earlier than Cisco Unified SIP SRST version 3.4 or B2BUA for Cisco Unified SIP SRST version 3.4 and higher services.

A SIP phone uses these services when it is unable to communicate with its primary SIP proxy of CUCM in the event of a WAN connection outage.

Cisco Unified SIP SRST can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across SIP networks in the same way that SCCP phones do.

Cisco Unified SIP SRST supports the following call combinations: SIP phone to SIP phone, SIP phone to PSTN or router voice port, SIP phone to SCCP phone, and SIP phone to WAN VoIP using SIP.

SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These servers usually are located in the core of a VoIP network. If SIP phones located at remote sites at the edge of the VoIP network lose connectivity to the network core (because of a WAN outage), they may be unable to make or receive calls. Cisco Unified SIP SRST functionality on a SIP PSTN gateway provides service reliability for SIP-based IP Phones in the event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP Phones to continue making and receiving calls to and from the PSTN. They also can continue making and receiving calls to and from other SIP IP Phones by using the dial peers configured on the router.

When the IP WAN is up, the SIP phone registers with the SIP proxy server and establishes a connection to the B2BUA SIP registrar (B2BUA router). But, any calls from the SIP phone go to the SIP proxy server through the WAN and out to the PSTN.

When the IP WAN fails or the SIP proxy server goes down, the call from the SIP phone cannot get to the SIP proxy server. Instead, it goes through the B2BUA router out to the PSTN.

Cisco Unified SRST does not support enhanced features, such as Presence or Cisco Extension Mobility. Message Waiting Indicator (MWI) is also not supported in fallback mode.

CUCME in SRST Mode Usage

CUCME in SRST mode enables routers to provide basic call-handling support for Cisco Unified IP Phones if they lose connection to remote primary, secondary, and tertiary CUCM installations or if the WAN connection is down.

When Cisco Unified SRST functionality is provided by CUCM Express, you can use automatic provisioning of phones like you do with standard Cisco Unified SRST. However, because of the wide feature support of CUCM Express, more features can be used compared to the standard Cisco Unified SRST.

Examples of features that are provided only by CUCM Express in SRST mode are Call Park, Presence, Cisco Extension Mobility, and access to Cisco Unity Voice Messaging services using SCCP.

These features, however, cannot be configured automatically when a phone falls back to SRST mode. If a certain feature is applicable to all phones or directory numbers (DN), the configuration can be applied by a corresponding template. If features have to be enabled on a per-phone (or per-DN) basis, they have to be statically configured.

Phones that do not require unique feature configuration can be configured automatically so that only those phones that require individual configuration have to be statically configured in CUCM Express.

Cisco Unified SRST Operation

Figure 5-2 illustrates the function of Cisco Unified SRST.

CUCM supports Cisco Unified IP Phones at remote sites attached to Cisco multiservice routers across the WAN. The remote-site IP Phones register with CUCM. Keepalive messages are exchanged between IP Phones and the central CUCM server across the WAN. CUCM at the main site handles the call processing for the branch IP Phones. Note that Cisco IP Phones cannot register SCCP or SIP through the PSTN to CUCM even if the PSTN is functional because SCCP and SIP must run over an IP network.

SRST Function of Switchover Signaling

When Cisco Unified IP Phones lose contact with CUCM, as shown in Figure 5-3, because of any kind of IP WAN failure, they register with the local Cisco Unified SRST router to sustain the call-processing capability that is necessary to place and receive calls.

Figure 5-3

Figure 5-3 Cisco Unified SRST Function of Switchover Signaling

Cisco Unified SRST configuration provides the IP Phones with the alternative call control destination of the Cisco Unified SRST gateway.

When the WAN link fails, the IP Phones lose contact with the central CUCM but then register with the local Cisco Unified SRST gateway.

The Cisco Unified SRST gateway detects newly registered IP Phones, queries these IP Phones for their configuration, and then autoconfigures itself. The Cisco Unified SRST gateway uses SNAP technology to autoconfigure the branch office router to provide call processing for Cisco Unified IP Phones that are registered with the router.

SRST Function of the Call Flow After Switchover

Cisco Unified SRST ensures that Cisco Unified IP Phones offer continuous service by providing call-handling support directly from the Cisco Unified SRST router using a basic set of call-handling features, as shown in Figure 5-4.

Figure 5-4

Figure 5-4 SRST Function of the Call Flow After Switchover

The Cisco Unified SRST gateway uses the local PSTN breakout with configured dial peers. Cisco Unified SRST features such as call preservation, autoprovisioning, and failover are supported.

During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP Phones display a message informing users that the phone is operating in CUCM fallback mode. This message can be adjusted.

While in CUCM fallback mode, Cisco Unified IP Phones continue sending keepalive messages in an attempt to reestablish a connection with the CUCM server at the main site.

SRST Function of Switchback

Figure 5-5 shows Cisco Unified IP Phones attempting to reestablish a connection over the WAN link with CUCM at the main site periodically when they are registered with a Cisco Unified SRST gateway.

Cisco IP Phones, by default, wait up to 120 seconds before attempting to reestablish a connection to a remote CUCM.

When the WAN link or connection to the primary CUCM is restored, the Cisco Unified IP Phones reregister with their primary CUCM. Three switchback methods are available on the Cisco IOS router: immediate switchback, graceful switchback (after all outgoing calls on the gateway are completed), or switchback after a configured delay. When switchback is completed, call handling reverts to the primary CUCM, and SRST returns to standby mode. The phones then return to their full functionality provided by CUCM.

SRST Timing

Typically, a phone takes three times the keepalive period to discover that its connection to CUCM has failed. The default keepalive period is 30 seconds, as shown in Figure 5-6.

If the IP Phone has an active standby connection established with a Cisco Unified SRST router, the fallback process takes 10 to 20 seconds after the connection with CUCM is lost. An active standby connection to a Cisco Unified SRST router exists only if the phone has a single CUCM in its Cisco Unified CM group. Otherwise, the phone activates a standby connection to its secondary CUCM.

If a Cisco Unified IP Phone has multiple CUCM systems in its Cisco Unified CM group, the phone progresses through its list before attempting to connect with its local Cisco Unified SRST router. Therefore, the time that passes before the Cisco Unified IP Phone eventually establishes a connection with the Cisco Unified SRST router increases with each attempt to connect to a CUCM. Assuming that each attempt to connect to a CUCM takes about 1 minute, the Cisco Unified IP Phone in question could remain offline for 3 minutes or more following a WAN link failure. You can decrease this time by setting the keepalive timer to a smaller value. You can configure the keepalive timer using the CUCM service parameter Station Keepalive Interval.

While in SRST mode, Cisco Unified IP Phones periodically attempt to reestablish a connection with CUCM at the main site. The default time that Cisco Unified IP Phones wait before attempting to reestablish a connection to CUCM is 120 seconds.

MGCP Fallback Operation

MGCP gateway fallback, as shown in Figure 5-7, is a feature that improves the reliability of MGCP remote site networks. A WAN link connects the MGCP gateway at a remote site to the Cisco Communications Manager at a main site, which is the MGCP call agent. If the WAN link fails, the fallback feature keeps the gateway working as an H.323 or SIP gateway and rehomes to the MGCP call agent when the WAN link is active again. MGCP gateway fallback works in conjunction with the SRST feature.

Figure 5-7

Figure 5-7 MGCP Gateway Fallback in a Normal Situation

Cisco IOS gateways can maintain links to up to two backup CUCM servers in addition to a primary CUCM. This redundancy enables a voice gateway to switch over to a backup server if the gateway loses communication with the primary server. The secondary backup server takes control of the devices that are registered with the primary CUCM. The tertiary backup takes control of the registered devices if both the primary and secondary backup CUCM systems fail. The gateway preserves existing connections during a switchover to a backup CUCM server.

When the primary CUCM server becomes available again, control reverts to that server. Reverting to the primary server can occur in several ways: immediately, after a configurable amount of time, or only when all connected sessions are released.

MGCP Gateway Fallback During Switchover

The MGCP gateway performs a switchover to its default technology of H.323 or SIP, as shown in Figure 5-8, when the keepalives between CUCM and the Cisco MGCP gateway are missing.

Figure 5-8

Figure 5-8 MGCP Gateway Fallback During Switchover

The MGCP gateway fallback feature provides the following functionality:

  • MGCP gateway fallback support: All active MGCP analog, E1 CAS, and T1 CAS calls are maintained during the fallback transition. Callers are unaware of the fallback transition, and the active MGCP calls are cleared only when the callers hang up. Active MGCP PRI backhaul calls are released during fallback. Any transient MGCP calls that are not in the connected state are cleared at the onset of the fallback transition and must be attempted again later.
  • Basic connection services in fallback mode: Basic connection services are provided for IP telephony traffic that passes through the gateway. When the local MGCP gateway transitions into fallback mode, the default H.323 or SIP session application assumes responsibility for handling new calls. Only basic two-party voice calls are supported during the fallback period. When a user completes (hangs up) an active MGCP call, the MGCP application handles the on-hook event and clears all call resources.

MGCP Gateway Fallback During Switchback

The MGCP-gateway-fallback feature provides the rehome functionality to switch back to MGCP mode. As shown in Figure 5-9, the switchback or rehome mechanism is triggered by the reestablishment of the TCP connection between CUCM and the Cisco MGCP gateway.

Figure 5-9

Figure 5-9 MGCP Gateway Fallback During Switchback

Rehome function in gateway-fallback mode detects the restoration of a WAN TCP connection to any CUCM server. When the fallback mode is in effect, the affected MGCP gateway repeatedly tries to open a TCP connection to a CUCM server that is included in the prioritized list of call agents. This process continues until a CUCM server in the prioritized list responds. The TCP open request from the MGCP gateway is honored, and the gateway reverts to MGCP mode. The gateway sends a RestartInProgress (RSIP) message to begin registration with the responding CUCM.

All currently active calls that are initiated and set up during the fallback period are maintained by the default H.323 session application, except ISDN T1 and E1 PRI calls. Transient calls are released. After rehome occurs, the new CUCM assumes responsibility for controlling new IP telephony activity.

MGCP Gateway Fallback Process

The MGCP gateway maintains a remote connection to a centralized CUCM cluster, as shown in Figure 5-10, by sending MGCP keepalive messages to the CUCM server at 15-second intervals.

If the active CUCM server fails to acknowledge receipt of the keepalive message within 30 seconds, the gateway attempts to switch over to the next available CUCM server.

If none of the CUCM servers responds, the gateway switches into fallback mode and reverts to the default H.323 or SIP session application for basic call control.

The gateway processes calls on its own using H.323 until one of the CUCM connections is restored. The same occurs if SIP is used instead of H.323 on the gateway.

Sours: https://www.ciscopress.com/articles/article.asp?p=1744068&seqNum=4

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Survivable Remote Site Telephony

SRST

Survivable Remote Site Telephony (SRST) is a Cisco Unified Communications Manager (CUCM) call processing backup mechanism that allows Cisco IP phones to register to a Cisco router. A Cisco router in SRST mode provides call setup and call teardown services similar to CUCM, but only when connectivity to the CUCM servers provided in the Cisco Call Manager group are not reachable. SRST technology is widely used in the centralized call processing model where branch sites do not have any local call processing capabilities. Cisco IP phones at branch locations register with CUCM over wide area network (WAN) links. All call setup and call teardown functions of the branch site Cisco phones are carried over the WAN links. Call setup and call teardown (call control) bandwidth requirements are benign (under 1kbps). Table 3-12 of the CUCM 7.x SRND recommends the amount of call control bandwidth dependent upon the number of users at the site. If WAN connectivity is lost between CUCM and the branch site, Cisco IP phones will re-register to their SRST reference provisioned in the device pool configuration of the Cisco IP phone. Cisco IP phones send a TCP keepalive to their primary CUCM server every 30 seconds by default. If 3 keepalives are missed, the Cisco IP phone will re-register with another CUCM server in their Call Manager group configuration and attempt to register with the SRST reference if no CUCM servers are available. The SRST failover process occurs faster if there is an active event on the Cisco IP phone device (e.g. Cisco IP phone was taken offhook). The SRST reference can be configured to the default gateway of the DHCP scope or an administratively defined IP address. In many deployments, the default gateway of the site is the only router(s) at the branch location. The default gateway of a branch location may be a virtual IP address shared between two Cisco routers using a virtual router redundancy protocol (HSRP, GLBP, or VRRP). As of this writing, the current version of SRST is 7.1. The SRST Administration guide maps SRST release versions to the Cisco IOS version required for the SRST version. In the next blog, we will continue our SRST conversation with the number of Cisco IP phones and directory numbers supported by different router models. Links SRST Administration Guide http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srstsa.html

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