Cisco SIP SRST V3.4: Cisco IOS SIP Survivable Remote Site Telephony Feature Overview

This chapter includes information about supported Cisco IP phones and platforms. It also includes information on Cisco IOS SIP SRST specifications, features, prerequisites, restrictions, and where to find additional reference documents.

Note For the most up-to-date information about Cisco IP phone support, the maximum number of Cisco IP phones, the maximum number of DNs or virtual voice ports, and memory requirements for Cisco SRST and Cisco SIP SRST, see the Cisco Survivable Remote Site Telephony (SRST) 3.4 Specifications for Cisco IOS Release 12.4(4)T at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/srst/srst34/srs34spc.htm

Cisco SIP SRST Description

This book describes Survivable Remote Site Telephony (SRST) functionality for Session Initiation Protocol (SIP) networks. Cisco SIP SRST provides backup to an external SIP proxy server 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 SIP SRST can support SIP phones with standard RFC 3261 feature support locally and across SIP WAN networks. With Cisco SIP SRST 3.4, SIP phones can place calls across SIP networks in the same way as SCCP phones.

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 SIP SRST functionality on a SIP PSTN gateway provides service reliability for SIP-based IP phones in the event of a WAN outage. Cisco 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.

Figure 1 shows that when the WAN is up, dual registration occurs. The phone registers with the SIP proxy server and the 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.

Figure 2 shows that when the WAN or SIP proxy server goes down, the call from the SIP phone cannot get to the SIP proxy server and instead goes through the B2BUA router out to the PSTN.

Support for Cisco IP Phones and Platforms

The following sections provide information about Cisco Feature Navigator and the histories of Cisco IP phone and platform support from Cisco SRST Version 3.0 to the present version.

Finding Cisco IOS Software Releases That Support Cisco SRST

The tables in this chapter list only the Cisco IOS software releases that first introduce new features to Cisco SRST. Other Cisco IOS software releases may subsequently inherit versions of Cisco SRST. To get a list of Cisco IOS software releases that support a particular version of Cisco SRST, use Cisco Feature Navigator.

Cisco Feature Navigator is a web-based tool that enables you to determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions found at this URL:

Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

Cisco IP Phone Support

For the most up-to-date information about Cisco IP phone support, see the Cisco Survivable Remote Site Telephony (SRST) 3.4 Specifications for Cisco IOS Software Release 12.4(4)T at the following URL:

Cisco 7940G and Cisco 7960G IP phones are fully supported if dual registration is enabled. Dual registration means that the SIP phone is capable of registering with the main SIP proxy and the Cisco SIP SRST device (redirect server or back-to-back user agent) at the same time. If this requirement is not met, the Cisco SIP SRST device may not be capable of routing incoming calls to the SIP phone until the SIP phone registers with the Cisco SIP SRST device. Cisco IP 7940G and 7960G phones beginning with phone load POS3-04-2-00.bin are capable of dual registration of the phone's primary phone line. Additional lines are not registered by the phone for Cisco SIP SRST. To enable dual registration for the primary line, you must set backup proxy information such as proxy_backup and proxy_backup_port in the SIP phone's configuration file. For configuration instructions, see the Cisco SIP IP Phone 7960 Administrator Guide, Version 5.1.

Cisco IP phones 7905G, Cisco 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 SIP SRST.

Platform and Memory Support

Prerequisites for Configuring Cisco SIP SRST

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An SRST feature license is required to enable the Cisco SIP SRST feature. Please contact your account representative if you have further questions.

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Note

When the WAN goes down, for each outgoing call the SIP phone continues to send the SIP proxy server up to seven Invite messages. If the Invite messages are not acknowledged, the SIP phone switches to Cisco SIP SRST to route the call. Thus, there may be a few seconds delay before SIP SRST takes over call processing from the SIP proxy server. If your network is designed to return an ICMP host unreachable indication to the phone in response to an outgoing SIP Invite message when the WAN is down, the phone responds by switching to the Cisco SIP SRST router more rapidly.

Dual registration is not supported on the Cisco IP phone 7905G, Cisco 7912G, or Cisco Analog Telephone Adaptor (ATA) series with a SIP image. Therefore auto registration to the SIP SRST router is not available.

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If the WAN is down, and you reboot your Cisco SIP SRST router, when the router reloads it will have no database of SIP phone registrations. The SIP phones will have to register again, which could take several minutes, because SIP phones do not use a keepalive functionality. To shorten the time before the phones re-register, the registration expiry can be adjusted with the registrar server command. The default expiry is 3600 seconds; an expiry of 600 seconds is recommended.

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For Cisco SIP SRST Version 3.0, SIP phones must support 300 or 302 Redirect messages.

Restrictions for Configuring Cisco SIP SRST

Table 3 provides a history of restrictions from Cisco SIP SRST Version 3.0 to the present version.

Table 3 History of Restrictions from Cisco SIP SRST Version 3.0 to the Present Version
Cisco SRST Version	Cisco IOS Release	Restrictions
Version 3.4 Version 3.2 Version 3.1 Version 3.0	12.4(4)T 12.3(11)T 12.3(7)T 12.2(15)ZJ 12.3(4)T	Not Supported •Music on hold (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 Phones 7912 and 7905G with a 1.0(2) build, the incoming call leg should be configured with the G.711 codec. Note Cisco IP phones 7905G, Cisco 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 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 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 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).

Where to Go Next

The next chapters of this book describe how to configure Cisco SIP SRST. As shown in Table 4, each chapter takes you through tasks in the order in which they need to be performed. The first task for configuring Cisco SRST is to ensure that the basic software and hardware in your system are configured correctly for Cisco SRST. For instructions, see the "Prerequisites for Configuring Cisco SIP SRST" section.

Table 4 Cisco SRST Configuration Sequence
Task	Where Task Is Described
1. If you are upgrading to Version 3.4 or using Cisco SIP SRST for the first time, this chapter describes procedures to get you started.	"Upgrading from Cisco IOS SIP SRST Version 3.0 to Version 3.4" chapter
2. This chapter describes how to provide a backup to an external SIP proxy server by providing basic registrar services.	"Configuring the SIP Registrar" chapter
3. This chapter describes basic Cisco SIP SRST and local SIP phone configurations that were introduced in Version 3.0.	"Configuring Cisco SIP SRST Version 3.0 Features Using Redirect Mode" chapter
4. This chapter describes global phone configurations and additional features, such as call forwarding, that were introduced in Version 3.0.	"Configuring Cisco SIP SRST Version 3.4 Features Using Back-to-Back User Agent Mode" chapter

Additional References

Related Topic	Documents
SRST commands and specifications	• Cisco IOS Survivable Remote Site Telephony Version 3.4 Command Reference • Cisco Survivable Remote Site Telephony (SRST) 3.4 Specifications for Cisco IOS Release 12.4(4)T
SRST administration	• Cisco IOS Survivable Remote Site Telephony Version 3.4 Administrator Guide
Cisco IP phones	• Cisco IP Phone 7902 Quick Start Guide • Cisco IP Phone 7902G Quick Start Guide • At a Glance Cisco IP Phone 7912G • Cisco IP Phone 7960 and 7940 Series User Guide • Cisco IP Phone 7970 Guide • Cisco SIP IP Phone 7960 Administrator Guide, Version 5.1
Cisco SIP functionality	• Cisco IOS SIP Configuration Guide
Command reference information for voice and telephony commands	• Cisco IOS Voice Command Reference • Cisco IOS Debug Command Reference
Standard preface	•Cisco IOS Voice Configuration Library Preface
Standard glossary	•Cisco IOS Voice Configuration Library Glossary

RFC	Title
RFC 2543	SIP: Session Initiation Protocol
RFC 3261	SIP: Session Initiation Protocol

Table Of Contents

Cisco IOS SIP SRST Feature Overview

Contents

Cisco SIP SRST Description

Support for Cisco IP Phones and Platforms

Finding Cisco IOS Software Releases That Support Cisco SRST

Cisco IP Phone Support

Platform and Memory Support

Prerequisites for Configuring Cisco SIP SRST

Restrictions for Configuring Cisco SIP SRST

Where to Go Next

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance