Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
Bias-Free Language
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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.
Figure 1. Branch Office Cisco Unifed IP Phones Connected to a Remote Central Cisco Unified Communications Manage Operating in SRST Mode
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:
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.
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.
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.
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.
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.
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.
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.
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:
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:
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.
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.
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.
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:
Router(config-sip-ua)# exit
Exits SIP configuration mode.
Step 13
voice service{ voip}
Example:
Router (config)# voice service voip
Enters voice-service configuration mode to specify a voice encapsulation type.
voip — Specifies Voice over IP (VoIP) parameters.
Step 14
sip
Example:
Router(config-voi-serv)# sip
Enters SIP configuration mode.
Step 15
no call service stop
Example:
Router(config-serv-sip)# call service stop
Activates SIP call service.
Step 16
exit
Example:
Router(config-serv-sip)# exit
Exits SIP configuration mode.
Step 17
voice register global
Example:
Router(config)# voice register global
Enters voice register global configuration mode to set parameters for all supported SIP phones in Cisco Unified CME.
Step 18
default mode
Example:
Router(config-register-global)# default mode
Enables mode for provisioning SIP phones in Unified SRST. The default mode is Unified SRST itself.
Step 19
max-dnmax-directory-numbers
Example:
Router(config-register-global)# max-dn 50
Limits number of directory numbers to be supported by this router.
Maximum number is platform and version-specific. Type ? for value.
Step 20
max-poolmax-voice-register-pools
Example:
Router(config-register-global)# max-pool 40
Sets maximum number of SIP phones to be supported by the Unified SRST router.
Step 21
exit
Example:
Router(config-register-global)# exit
Exits voice register global configuration mode.
Step 22
voice register poolpool-tag
Example:
Router(config)# voice register pool 1
Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.
Step 23
id{ networkaddressmaskmask| ip address maskmask| macaddress}
Example:
Router(config-register-pool)# id network
2001:420:54FF:13::901:0/117
Router(config-register-pool)# id network
10.64.88.0 mask 255.255.255.0
Explicitly identifies a locally available individual SIP phone to support a degree of authentication.
Step 24
end
Example:
Router(config)# end
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:
ipv6 unicast-routing
voice service voip
sip
no anat
call service stop
exit
exit
sip-ua
protocol mode dual-stack
exit
voice service voip
sip
no call service stop
exit
voice register global
default mode
max-dn 50
max-pool 40
exit
voice register pool 1
id network 2001:420:54FF:13::901:0/117
end
The following example provides interface configuration for IPv6 supported on Unified SRST:
configure terminal
interface GigabitEthernet0/0/1
ip address 10.64.86.229 255.255.255.0
negotiation auto
ipv6 address 2001:420:54FF:13::312:82/119
ipv6 enable
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:
Router# show sip service
SIP service is shut
under voice service voip, sip submode
The following example displays output when SIP call service is active with the nocall service stop CLI command:
Router# show sip-ua service
SIP Service is up
under voice service voip, sip submode
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.
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.
During Cisco Unified CM fallback, Cisco Unified SRST considers the Cisco VG248 to be a group of Cisco Unified IP phones. Cisco
Unified SRST counts each of the 48 ports on the Cisco VG248 as a separate Cisco Unified IP phone. Support for Cisco VG248
Version 1.2(1) and higher versions is available as of Cisco SRST Version 2.1. For more information, see Cisco VG248 Analog Phone Gateway Data Sheet and Cisco VG248 Analog Phone Gateway Version 1.2(1) Release Notes.
For IPv6 Support on Unified SRST, all the legacy IP Phones and Voice Gateways must be converted or reconfigured to IPv4-Only
SIP signaling from SCCP signaling, if applicable.
Platform and Memory Support
Platform and Memory Support
For the most up-to-date information about Platform and Memory Support, see Compatibility Information.
Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that are supported on specific platforms. To get updated information regarding
platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Availability of Cisco IOS Software Images
Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those
platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information
about platform support and availability of software images for each Cisco IOS software release, see the online release notes
or, if supported, Cisco Feature Navigator.
Cisco SRST 3.2 and later versions support all PRI and BRI switches including the following:
basic-1tr6
basic-5ess
basic-dms100
basic-net3
basic-ni
basic-ntt NTT switch type for Japan
basic-ts013
primary-4ess Lucent 4ESS switch type for the United States
primary-5ess Lucent 5ESS switch type for the United States
primary-dms100 Northern Telecom DMS-100 switch type for the United States
primary-net5 NET5 switch type for the United Kingdom, Europe, Asia, and Australia
primary-ni National ISDN switch type for the United States
primary-ntt NTT switch type for Japan
primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)
Multicast Music On Hold
Multicast Music On Hold
For Unified SRST 3.0 and later versions, you can configure the MOH audio stream as a multicast source. A Unified SRST router
that is configured for multicast MOH also transmits the audio stream on the physical IP interfaces of the specified router
to permit access to the stream by external devices. Certain IP phones do not support multicast MOH because they do not support
IP multicast. You can disable multicast MOH to individual phones that do not support multicast. Callers hear a repeating tone
when they are placed on hold.
Multicast MOH on Unified SRST is supported for both SIP and SCCP phones. Support is offered for G.711 and G.729 codecs with
multicast MOH on Unified SRST. Multicast MOH is supported on Cisco Integrated Services Router Generation 2 (ISR G2) and the
Cisco 4000 Series Integrated Services Routers.
For SIP phones to play the Multicast MOH, you need to configure the CLI command mohenable-g711filename (for example, mohenable-g711flash:en_bacd_music_on_hold.au or mohg729 flash:SampleAudioSource.g729.wav). For SCCP phones to play Multicast MOH, you need to configure the CLI command multicast mohip-addressportport-number [routeip-address-list] (for example, multicast moh239.1.1.1port2000), apart from the CLI command mohfilename. If both the CLI commands are not configured, SCCP phones will only play tone on hold.
If you specify a file with this command and later want to use a different file, you must disable use of the first file with
the no moh command before configuring the second file.
Router(config-cm-fallback)# multicast moh 239.1.1.1
port 2000
Specifies that this audio stream is to be used for multicast and also for MOH.
Note
This command is required to use MOH for internal calls and it must be configured after MOH is enabled with the moh command.
ip-address—Destination IP address for multicast.
port port-number—Media port for multicast. Range is 2000 to 65535. We recommend port 2000 because it is already used for normal
RTP media transmissions between IP phones and the router.
Note
Valid port numbers for multicast include even numbers that range from 16384 to 32767. (The system reserves odd values.)
route—(Optional) List of explicit router interfaces for the IP multicast packets.
ip-address-list—(Optional) List of up to four explicit routes for multicast MOH. The default is that the MOH multicast stream
is automatically output on the interfaces that correspond to the address that was configured with the ip source-address command.
Note
For MOH on internal calls, packet flow must be enabled to the subnet on which the phones are located.
Step 6
exit
Example:
Router(config-cm-fallback)# exit
Exits call-manager-fallback configuration mode.
Where to Go Next
The next chapters of this book describe how to configure Cisco Unified SIP SRST. As shown in the following table, each chapter
takes you through tasks in the order in which they need to be performed. The first task for configuring Cisco Unified SRST
is to ensure that the basic software and hardware in your system are configured correctly for Cisco Unified SRST.
Task
Where Task Is Described
7. Setting up a Cisco Unified SRST system to communicate with your network
Setting Up the Network
8. Configuring Version 4.1 features
Cisco Unified SIP SRST 4.1
9. Setting up the basic Cisco Unified SRST phone configuration using SCCP
Setting Up Cisco Unified IP Phones using SCCP
10. Providing a backup to an external SIP call control (IP-PBX) by supplying basic registrar services
Setting Up Cisco Unified IP Phones using SIP
11. Configuring incoming and outgoing calls
Configuring Call Handling
12. Configuring optional security for SRST
Configuring Secure SRST for SCCP and SIP
13. Setting up voicemail
Integrating Voicemail with Cisco Unified SRST
14. Setting up video parameters
Setting Video Parameters
15. Monitoring and maintaining Cisco Unified Survivable Remote Site Telephony (SRST)
No new or modified MIBs are supported by this feature, and support for existing MIBs has not been modified by this feature.
To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the
following URL:
http://www.cisco.com/go/mibs
RFCs
RFC
Title
RFC 2246
The Transport Layer Security (TLS) Protocol Version 1.0
RFC 2543
SIP: Session Initiation Protocol
RFC 3261
SIP: Session Initiation Protocol
RFC 3711
The Secure Real-Time Transport Protocol (SRTP)
Technical Assistance
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Link
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http://www.cisco.com/techsupport
Obtaining Documentation, Obtaining Support, and Security Guidelines
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also recommended aliases and general Cisco documents, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at
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