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Configuration SIP SurvivabilityAvaya Secure Router 2330/4134

10.3NN47263-510, 02.01

October 2010


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Contents

Chapter 1: New in this release.................................................................................................7Features............................................................................................................................................................7

SSM digest authentication........................................................................................................................7

Chapter 2: Introduction.............................................................................................................9Navigation.........................................................................................................................................................9

Chapter 3: SIP Survivability fundamentals...........................................................................11Deployment scenarios.....................................................................................................................................11

Normal mode...................................................................................................................................................13

SIP REGISTER handling........................................................................................................................14

SIP Proxy functionality............................................................................................................................14

Call Admission Control functionality.......................................................................................................14

Survivable mode.............................................................................................................................................15

SIP REGISTER handling........................................................................................................................16

SIP Proxy functionality............................................................................................................................17

Persistence across reboots....................................................................................................................17NTML..............................................................................................................................................................17

SSM feature licensing.....................................................................................................................................18

SSM digest authentication..............................................................................................................................18

Supported SIP servers and endpoints............................................................................................................18

SIP server interoperability with SSM...............................................................................................................19

Avaya Aura.............................................................................................................................................19

Avaya CS 1000.......................................................................................................................................21

Avaya CS 2100.......................................................................................................................................23

Avaya SCS.............................................................................................................................................24

Chapter 4: SSM basic configuration......................................................................................25Prerequisites for SSM basic configuration......................................................................................................25

SSM basic configuration tasks........................................................................................................................25SSM operating in survivable mode only.................................................................................................25

SSM operating in normal or survivable mode.........................................................................................26

SSM basic configuration tasks navigation.......................................................................................................27

Binding an IP address for SSM.......................................................................................................................27

Enabling SSM.................................................................................................................................................28

Configuring the domain...................................................................................................................................29

Configuring keepalives for SIP survivability....................................................................................................29

Configuring the call admission control on an interface....................................................................................30

Configuring an exclusion pool for call admission control................................................................................31

Configuring the default gateway......................................................................................................................32

Chapter 5: SSM configuration examples...............................................................................33

Basic SSM configuration example..................................................................................................................33SSM configuration example with existing LAN infrastructure..........................................................................36

Chapter 6: SSM additional configuration..............................................................................39Navigation.......................................................................................................................................................39

Loading and storing the dial plan....................................................................................................................40

Configuring the SIP transport and ports..........................................................................................................41

Configuration SIP Survivability October 2010 3


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Configuring user provisioning..........................................................................................................................41

Configuring digest authentication....................................................................................................................42

Configuring digest authentication user credentials.........................................................................................43

Configuring the SIP registrar for SIP survivability...........................................................................................44

Configuring the DNS timeout..........................................................................................................................45

Configuring service rejection...........................................................................................................................45

Configuring the SIP timer values.....................................................................................................................46Configuring session timers..............................................................................................................................47

Configuring the SIP header values.................................................................................................................48

Configuring SSM user licensing......................................................................................................................49

Configuring SSM congestion control...............................................................................................................50

Displaying the SIP survivability CAC configuration.........................................................................................52

Displaying the SIP survivability feature configuration.....................................................................................52

Displaying the SIP survivability dial plan filenames........................................................................................52

Displaying the SIP survivability licensed user capacity...................................................................................52

Displaying the SIP survivability protocol header configuration........................................................................53

Displaying the SIP survivability registered users............................................................................................53

Displaying the SIP survivability registrar parameters......................................................................................53

Displaying the SIP survivability session timer configuration...........................................................................53

Displaying the SSM server configuration........................................................................................................54Displaying the SIP survivability feature status................................................................................................54

Displaying the survivability SIP server statistics.............................................................................................54

Displaying the SIP survivability subscribers....................................................................................................55

Flushing the SSM database............................................................................................................................55

Clearing the SIP survivability feature statistics...............................................................................................55

Debug procedures...........................................................................................................................................55

Debugging with SIP message dump trace.............................................................................................56

Debugging with module trace.................................................................................................................56

Debugging with error trace.....................................................................................................................57

Chapter 7: Number Translation Markup Language..............................................................59Terminology and XML basics..........................................................................................................................59

Translation language.......................................................................................................................................60NTML interpreter....................................................................................................................................60

Nodes in the NTML language.................................................................................................................61

Translation node.....................................................................................................................................62

Switch nodes..........................................................................................................................................63

Action nodes...........................................................................................................................................68

Route nodes...........................................................................................................................................73

Exit node.................................................................................................................................................74

Chapter 8: SSM configuration example with NTML.............................................................75Basic SSM configuration example with NTML................................................................................................75

Chapter 9: Additional NTML examples..................................................................................77

Navigation.......................................................................................................................................................77NTML with default route..................................................................................................................................77

Route selection based on number in Req-uri header......................................................................................77

Exact match routing rule.........................................................................................................................78

Routing rule based on number of digits..................................................................................................78

Routing rule based on prefix and variable length...................................................................................79

Routing rule based on prefix and fixed length........................................................................................79



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Routing rule based on range of numbers...............................................................................................79

Routing rule based on more than one criterion......................................................................................80

Route selection based on number in From header.........................................................................................81

Route selection based on number in To header..............................................................................................81

Route selection based on user-name in Req-uri header.................................................................................82

Route selection based on domain in Req-uri header......................................................................................82

Route Selection based on previous hop address............................................................................................83Translation of Request URI number................................................................................................................83

Inserting numbers...................................................................................................................................84

Dropping numbers..................................................................................................................................84

Replacing numbers.................................................................................................................................84

Translation based on mixed criterion......................................................................................................85



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Chapter 1: New in this release

The following section details what is new inAvaya Secure Router 2330/4134 Configuration SIP

Survivability(NN47263-510) for Release 10.3.

Features

See the following sections for information about supported features:

SSM digest authentication

SSM supports digest authentication for subscribers in backup mode. For more information,

see the following sections:

SSM digest authenticationon page 18

Configuring digest authenticationon page 42

Configuring digest authentication user credentialson page 43



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New in this release



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Chapter 2: Introduction

This document provides information you need to configure the SIP Survivability feature.

Navigation

SIP Survivability fundamentalson page 11

SSM basic configurationon page 25

SSM configuration exampleson page 33

SSM additional configurationon page 39

Number Translation Markup Languageon page 59

SSM configuration example with NTMLon page 75

Additional NTML exampleson page 77



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Introduction



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Chapter 3: SIP Survivability fundamentals

In a centralized SIP server architecture, the remote branches make use of the call processing resourcesavailable at a central location, generally located at the corporate headquarters. The SIP survivabilityfeature enhances the feature set of the Avaya Secure Router 4134 (SR4134) and Avaya Secure Router2330 (SR2330) by providing business continuity to the branch office in the event of a SIP server failureor of a WAN connection outage to the corporate headquarters. With this feature, employees at the branchoffice can continue to use SIP phones to place and receive intra-site calls and make calls over the PSTN,including 911 calls, after a SIP server or WAN connection failure.

The SIP survivability module (SSM) is a software-only subsystem on the SR2330/4134 that provides SIPsurvivability capabilities. The SSM operates as a SIP Back to Back User Agent (B2BUA) that can backup a central SIP server by providing basic call services to connected endpoints at the branch if the WANconnection to the central SIP server fails.

SSM is a licensable feature. A 25-user SSM license is included for free with the installed PVIM card onthe SR2330/4134.

Deployment scenarios

You can deploy SIP Survivability in a Remote Branch Office network deployment.



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Figure 1: Remote Branch Office network

A Remote Branch Office (RBO) makes use of the VoIP infrastructure deployed at a centralcorporate location. During normal operation when the WAN is up and running, all calls (intra-RBO, inter-RBO, PSTN) are routed through the central infrastructure deployed at theheadquarters. In Normal mode, the SSM acts as a proxy server, forwarding outgoing calls tothe SIP server for call routing, and forwarding incoming calls to the branch SIP endpoints.

When the central SIP server or WAN connection goes down, the Secure Router enterssurvivable mode and the SSM handles intra-RBO and PSTN calls. Inter-RBO VoIP calls cannotbe handled because the central infrastructure is not reachable. However, Inter-RBO calls canstill be made if they are routable using PSTN trunks.

The following figure shows a variation on the RBO network example in which the Secure Routeris installed in an existing branch LAN and therefore does not require routing capabilities. Inthis case, all SIP Phones are connected on the existing LAN infrastructure and the SecureRouter connects to the LAN through a single Ethernet connection. The SIP Media Gatewayprovides connections to FXS phones and to PSTN trunks, and the SSM provides survivabilitycapabilities for the SIP Phones.

SIP Survivability fundamentals



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Figure 2: SSM with existing LAN infrastructure in Remote Branch Office network

The SR2330/4134 SSM operates in two modes Normal mode and Survivable mode.

In normal mode, the SSM functions as an outbound proxy and proxies all SIP messagesinitiated from the SIP phones and the SIP Media Gateway to the SIP Server located in thehead office.

In survivable mode, the SSM supports SIP server functionality to provide basic call features tothe SIP endpoints at the branch, and also supports local registrar functionality to storeregistrations. Survivable mode is also called backup mode as the SSM functions as a backupserver to the central SIP server in this mode.

The SSM can monitor the availability of the central SIP server in both modes using a keepalivemechanism and this information can be used to switch between the modes. SSM can be forcedto remain in survivable mode by not configuring this keepalive mechanism.

The SSM modes are explained in detail in the following sections.

Normal mode

For the proper functioning of SSM in Normal mode, you must configure the outbound proxy onthe SIP Phones to point to the SSM. In this case, when the SIP Phones send SIP messagesto the central SIP server, they forward the requests through the SSM.

Normal mode



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SIP REGISTER handling

In normal mode, the SSM proxies REGISTER messages (including new registrations, re-

registrations, and de-registrations) received from SIP endpoints (SIP Phones or SR2330/4134

FXS ports) to the pre-configured central SIP server. SSM modifies the Contact headers in theREGISTER messages to point to SSM bind IP address before forwarding the REGISTER to

the central SIP server to ensure that incoming calls are routed through the SSM.

On receiving a successful response (200OK) to a proxied registration message, the SSM

stores registered contacts locally in an in-memory database along with the expiration time seen

in the response to a proxied registration message. It also adds the subscriber information,

namely the registered subscribers AOR, into its information database. This feature allows for

addition of subscriber data dynamically, without having a need to pre-provision subscriber

information.

The SSM updates the registration expiry interval if re-registration is successful and removes

the registration entries from its local database if the de-registration is successful. The SSMRegistrar keeps track of the registration expiry and removes the entries from its database once

they expire.

The SSM ensures that the number of contacts for a given AOR never exceeds the configured

max-contactsvalue for the AOR. If a new registration request for an AOR exceeds the

maximum allowed contacts for that AOR, SSM rejects the request with a 4xx response.

SIP Proxy functionality

The SSM behaves as an outbound proxy and forwards all calls to the central SIP server. SSM

supports proxying of all types of SIP messages REGISTER, INVITE, REFER, SUBSCRIBE/

NOTIFY, UPDATE, INFO, PRACK as well as all types of SIP responses 1xx (with or without

100 rel where xx is not 00), 2xx, 3xx, 4xx, 5xx, 6xx. Both UDP and TCP transport mechanisms

are supported for SIP messages.

The SSM uses administratively configured rules for proxying outbound calls based on dialed

digit patterns using a dial plan written in Avaya NTML (Number Translation Markup Language).

The NTML rules are configured in a Normal Mode NTML file located on attached flash media

which is uploaded using CLI for SSM processing.

The SSM routes incoming calls from the central SIP server to the destination SIP endpoints

by retrieving the contact from the registration database.

Call Admission Control functionality

For the proper functioning of SSM in Normal mode, you must enable call admission control

(CAC).




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CAC is required on the IP WAN interface because, unlike trunks on circuit-switched networks,

packet-switched networks have no hard physical limit on the number of calls that can exist

on a link. If you do not configure CAC on the WAN link, an unlimited number of calls threatens to

consume all the link bandwidth, causing degradation of voice quality for all calls. With CAC

enabled, you can limit the number of simultaneous calls on the link to avoid bandwidth

overutilization on the WAN link.

SSM allows you to configure the max-call counter associated with the WAN link. This count

need to be calculated based on the WAN link bandwidth of the WAN router and not based

on the bandwidth of the link between SR2330/4134 and the router. When SR2330/4134 is also

used as a WAN router, the directly connected WAN link bandwidth is used for the max-call

count calculation.

On receiving an outbound or inbound call that would use WAN bandwidth for media, the SSM

verifies whether the max-call limit allows for the call to be admitted. If the max-call limit is

reached, SSM rejects the inbound call with response 503. To identify whether the call will use

WAN bandwidth, an additional configuration of exclude-pool is needed. An exclude-pool

identifies the IP address range of the SIP endpoints that will use SSM and are local to the

branch office. CAC processing will then exclude the incoming streams from these endpointsfrom the max-call limit count. When SR2330/4134 is also used as WAN router, the exclude-

pool configuration is not required.

SSM supports PSTN fallback on CAC failure. To enable the fallback feature, you must configure

the default-gatewaycommand. In this case, if the max-call limit is reached, the SSM

forwards the outbound call to the configured default gateway for routing. You must also

configure the specified default gateway as the bind IP of the SIP Media Gateway on the same

SR2330/4134.

In normal mode, the SSM also monitors the configured CAC interfaces and, if all configured

CAC interfaces are down, it switches to survivable mode. This feature is useful for

SR2330/4134 with WAN routing, when the directly connected WAN links are configured asCAC interfaces.

With the CAC feature, only static bandwidth monitoring is supported. With static bandwidth

monitoring, SSM tracks the number of simultaneous calls established through the WAN link

and does not allow it to exceed the configured maximum value. This method assumes that all

calls have the same bandwidth requirement. Take this operation into consideration when

configuring the max-call limit, especially if you expect bandwidth requirements to vary for each

call.

Survivable mode

In survivable mode, the SSM routes the intra-branch calls to the appropriate endpoint

according to the in-memory registration database. Calls for PSTN interfaces are routed to the

SIP Media Gateway. The Gateway can route the intra-branch POTS calls to the FXS endpoints.

Survivable mode



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Inter-branch calls are routed through the Gateway to the PSTN trunks, provided that they are

dialed as PSTN numbers.

Figure 3: Survivable mode

SSM periodically polls the central server using SIP OPTIONS requests, if configured. While in

Normal mode, when the server is not reachable (either because the server is down or the WAN

connectivity is down), SSM transitions to Survivable mode. When the server becomes

reachable, it reverts to Normal mode. If the periodic polling is not configured, SSM remains in

Survivable mode always. The SSM can also monitor WAN link status. This feature is useful

for SR2330/4134 with WAN routing enabled. In this scenario in Survivable mode, SSM moves

from "WAN Down" state to "WAN Up" state after recovery of the WAN link following a failure,

and then transitions to Normal mode when it gets a response to SIP OPTIONS requests.

SIP REGISTER handlingIn survivable mode, the SSM functions as a SIP Registrar and terminates REGISTER requests

by sending a 200 OK response after successful validation. SSM also provides optional

authentication of SIP endpoints. If authentication is enabled, you must provision the

subscribers by configuring SIP username and password. If authentication is disabled, a

configurable option is provided to allow all REGISTER requests or to restrict the registration

to provisioned users. In addition to dynamically populating subscribers at SSM, a manual




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provisioning of subscribers is also supported. When responding to the REGISTER request with

a successful response, SSM reduces the expiration interval to a small configured value (for

example 120 s) to ensure that registrations are refreshed frequently and are migrated back to

the Central server when it is available (that is, when SSM returns to Normal mode).

Similar to normal mode, SSM stores registered contacts locally in an in-memory database and

removes the entries from its database once they expire.

SSM refreshes the registration expiry timer on receiving a registration refresh and removes

the registration entry from its database on receiving a de-registration request.

SIP Proxy functionality

The SSM routes local intra-site calls in survivable mode using the in-memory database lookup.

It uses administratively configured rules for proxying incoming calls based on dial digit patterns

using a dial plan written in Avaya NTML. The NTML rules are specified through a survivable

mode NTML file uploaded using CLI.

SSM supports proxying of SIP messages required for support of endpoint hosted

supplementary services Call hold/resume, call transfer, call forward, call waiting, 3-way

conference, Do Not Disturb.

Persistence across reboots

In normal mode and in survivable mode, the in-memory database is periodically backed up in

flash file ssm.db. If the SR2330/4134 reboots, SSM reloads ssm.db to its memory and uses it

for call routing.

NTML

The Avaya Number Translation Markup Language (NTML) is an Extensible Markup Language(XML) based, proprietary scripting language used by SSM to define URI transformation rulesand specialized routing rules. You can use NTML scripts to modify the URI, for example, toinsert a string at any position within the user or host part of a URI, to remove a substring from aURI, or to replace the URI with an entirely new URI. NTML also allows decisions to be madebased on the value or pattern of the dialed URI or the caller or original called URI. It also allows

a set of next-hop routes to be specified for a given outgoing request to force specific routingrules.

SSM supports the configuration of NTML scripts for each of the modes: normal and survivable.

NTML



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SSM feature licensing

Software licensing limits the number of SSM users allowable on the SR2330/4134. If you boot

up the SR2330/4134 with the PVIM module only, in addition to the eight supported DSP

channels, the SR2330/4134 supports a maximum of 25 SSM users.

To operate the SR2330/4134 with additional SSM users, you must obtain a license key from

Avaya support. License keys can expand the maximum SSM user capacity to support 100

users.

The license upgrade procedure is similar to the DSP channel upgrade procedure. For more

information, seeAvaya Secure Router 2330/4134 Configuration SIP Media Gateway

(NN47263-508).

SSM digest authentication

SSM supports digest authentication for subscribers in backup mode. If authentication is

enabled, you must provision the subscribers by configuring SIP username and password using

subscriber-credentialscommand. If authentication is disabled, use the registrar

policycommand to allow all register requests, or to restrict the registration to provisioned

users. Digest authentication is skipped for all requests received from gateway (trusted hosts)

in backup mode.

Supported SIP servers and endpoints

SSM supports the following SIP servers and endpoints.

SIP servers SIP endpoints

Avaya Aura 5.2, 6.0 Avaya 96xx SIP Deskphone Series , one-X

Communicator

Avaya SES/CM 5.2.1 Avaya 96xx SIP Deskphone Series

Avaya CS 1000 6.0, 7.0 Avaya 1120E/1140E SIP DeskphonesAvaya CS 2100 Avaya 1120E/1140E SIP Deskphones,

Avaya Multimedia client, LG Nortel (Now LG

Ericsson) 6812, Polycom IP330




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SIP servers SIP endpoints

Avaya SCS 3.0 Avaya 12xx Series SIP Deskphones, Avaya

3456 UC client, LG Nortel (Now LG Ericsson)

6812, 8820, Polycom IP330

Nortel CS 2000 (Now part of Genband) Avaya 1120E/1140E SIP Deskphones,

Avaya Multimedia client, LG Nortel (Now LGEricsson) 6812, Polycom IP330

Broadsoft BroadWorks 14.0 SP1 LG Nortel (Now LG Ericsson) 6812, Polycom

IP330

Sylantro 4.1.1 (Now part of Broadsoft) LG Nortel (Now LG Ericsson) 6812, Polycom

IP330

SIP server interoperability with SSMThe mode of operation of SSM must be configured based on the central SIP server. Also the

SR2330/4134 gateway subsystem can be configured to communicate with the central SIP

server directly instead of using SSM as the outbound proxy. The following sections describe

the deployment model of SSM and gateway subsystems with each of the Avaya SIP servers.

Avaya Auraon page 19

Avaya CS 1000on page 21

Avaya CS 2100on page 23

Avaya SCSon page 24

Avaya Aura

Avaya Aura is the core communications platform for supporting midsize to large companies,

providing the foundation for all of Avayas Unified Communications and Customer Service

Solutions. The core components of Avaya Aura include: Avaya Aura Communication Manager

(CM), Avaya Aura Session Manager (SM) and Avaya Aura System Manager. Avaya Aura

Session Manager enables distributed SIP-based system solutions featuring multi-vendor

integration, centralized dial plans and user profiles, centralized SIP trunks, easily administered

"on-net" call routing, and greatly enhanced SIP scalability and security. Avaya Aura

Communication Manager provides robust PBX features, high reliability and scalability, andmulti-protocol support. Communication Manager can be deployed in one of two roles: it can

be deployed as a PBX or as a Feature Server. When deployed as a feature server,

Communication Manager can support features for up to 7000 SIP endpoints that register to

Session Manager.

The SR2330/4134 is positioned as the SIP survivable gateway solution for Avaya Aura.

SIP server interoperability with SSM



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In this deployment, the SR2330/4134 SSM module is always kept in Survivable mode by not

configuring server polling or the keepalive mechanism. When the Avaya Aura Session

Manager is reachable from the branch office, "simultaneous dual registration capable" SIP

endpoints register with the SM and the SSM separately and direct the calls to the SM. The

"alternate registration capable" SIP endpoints register with SM only and direct the calls to the

SM. These SIP endpoints monitor the reachability of the SM. If the SM is not reachable, they

move to survivable mode.

The following figure shows the interoperation of the SR2330/4134 with the Avaya Aura when

the Session Manager is reachable from the branch office.

Figure 4: Avaya Aura: Normal operation

When Avaya Aura Session Manager is not reachable from the branch office, "alternate

registration capable" SIP endpoints register with the SSM and direct the calls to the SSM. In

the case of "simultaneous dual registration capable" SIP endpoints, the registration step with

SSM is not required as these SIP endpoints were already registered when the session manager

was still reachable from the branch office which reduces the delay in switch over.

The following figure shows the operation of the SR2330/4134 when Avaya Aura Session

Manager is not reachable. SSM functions as a backup server to the central SIP server and

provides basic call features to the SIP endpoints including SIP phones, analog phones

connected to the gateway subsystem, and other devices.




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Figure 5: SR2330/4134 operation when central SIP server is not reachable

Configuration

Refer to the following documents for configuration and interoperability details:1. Avaya Solution & Interoperability Test Lab Application Note for Avaya Aura 6.0:

Configuring a Survivable SIP Gateway Solution using the Avaya Secure RouterSR2330 10.2.1, Avaya AuraSession Manager 6.0, Avaya AuraCommunicationManager 6.0, and Avaya Modular Messaging 5.2 Issue 1.0(http://support.avaya.com/css/P8/documents/100091900 )

2. Avaya Solution & Interoperability Test Lab Application Note for Avaya Aura 5.2:

Configuring a Survivable SIP Gateway Solution using the Avaya Secure Router,Avaya AuraSession Manager 5.2, Avaya AuraCommunication Manager 5.2,and Avaya Modular Messaging 5.2 in a Distributed Trunking Configuration Issue0.1(http://support.avaya.com/css/P8/documents/100090275 )

Avaya CS 1000

The Avaya Communication Server 1000 (CS 1000), which enables a simple evolutionary path

to the award-winning Avaya Aura unified communications solution, is a full-featured, highly-

scalable IP communications system that meets the needs of enterprises from small to large.

The core components of CS 1000 include SIP Line Gateway (SLG), SIP Signaling Gateway

(SSG) and SIP Proxy. SLG integrates SIP end points in the Communication Server 1000

system and extends business telephony features to SIP clients. The Network Routing Service

(NRS) application provides routing services to SIP-compliant devices. NRS is offered in two

versions: a SIP Redirect Server NRS and a SIP Proxy NRS (SPS). SSG is a softwarecomponent configured on virtual loops. They bridge CS 1000 call processing features and the

SIP network.

The SR2330/4134 is positioned as the SIP survivable gateway solution for Avaya CS 1000.


http://support.avaya.com/css/P8/documents/100091900http://support.avaya.com/css/P8/documents/100091900http://support.avaya.com/css/P8/documents/100090275http://support.avaya.com/css/P8/documents/100090275http://support.avaya.com/css/P8/documents/100091900http://support.avaya.com/css/P8/documents/100091900


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Normal mode

The SR2330/4134 SIP Gateway monitors the reachability of the currently active SPS usingSIP OPTIONS requests and registers with the currently active one. If the currently active SPS isnot reachable, the SIP Gateway switches to the other SPS as the currently active SPS.

SSM also monitors the reachability of the associated SIP Line Gateway (SLG) using SIP

OPTIONS requests. If the SLG is not reachable, the SR enters survivable mode.

In normal mode, the SR2330/4134 SSM proxies all Registration and INVITE requests from SIPUser Agents (SIP phone/softphone) to the SLG the Contact header is changed to includethe SSM IP address and port.

The following figure shows the interoperation of the SR2330/4134 with the Avaya CS 1000 innormal mode.

Figure 6: Avaya CS 1000: Normal mode

Survivable mode

The survivable mode operation with the CS 1000 is similar to the survivable mode operationwith the Avaya Aura (seeAvaya Auraon page 19 for details).

Configuration

Refer to the following document for configuration and interoperability details:




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Secure Router 2330/4134 as Communication Server 1000 Survivable SIP Branch Solution -Quick Start Configuration Guide(http://support.avaya.com/css/P8/documents/100099413 ).

Avaya CS 2100

Avaya Communication Server 2100 is an enterprise converged solution designed specifically

to address the demanding business and operational requirements of todays large enterprises.

The Communication Server 2100 platform is a best of all worlds integrated unified

communications solution that combines Avayas leading enterprise features and applications

with the exceptional scalability, reliability and networking usually only found in carrier solutions.

Combining these strengths with the capabilities of Avaya Aura architecture provides customers

an unmatched solution for consolidating dial plans, network access and applications into single

data center reducing network and operational cost.

The SR2330/4134 is positioned as the SIP survivable gateway solution for Avaya CS 2100.

SSM monitors the reachability of the call server using SIP OPTIONS request. If the call server is

not reachable, the SR2330/4134 enters survivable mode. In normal mode, SR2330/4134 SSM

proxies all Registration and INVITE requests from SIP end points and the SIP gateway to CS

2100.

Normal mode

The following figure shows the interoperation of the SR2330/4134 with the CS 2100 in normalmode.


http://support.avaya.com/css/P8/documents/100099413http://support.avaya.com/css/P8/documents/100099413


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Figure 7: Avaya CS 2100: Normal mode

Survivable mode

The survivable mode operation with the CS 2100 is similar to the survivable mode operation

with the Avaya Aura (seeAvaya Auraon page 19 for details).

Avaya SCS

The Avaya Software Communication System enables small and medium-sized businesses to

deliver powerful unified communications applications to improve employee productivity,

collaboration, and mobilityall while reducing communications costs. Functions include

unified messaging, meet me conferencing, video conferencing, presence, IM, advanced

softphones, personal auto attendant, find me/follow me capabilities, application integration with

Microsoft and IBM, and more. Employees can control and customize their apps with just a few

easy clicks.The Avaya SCS along with SR2330/4134 can be deployed as stand-alone, distributed and

centralized solutions for small and medium businesses.

When used in a centralized model, SR2330/4134 operation with SCS is similar to

SR2330/4134 operation with the CS 2100 (seeAvaya CS 2100on page 23 for details).




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Chapter 4: SSM basic configuration

This chapter describes the basic SSM configuration. The required configuration steps are listed below.

Prerequisites for SSM basic configuration

Internal PVIM module must be installed

If more than 25 users are required, SSM license must be purchased and installed

SSM basic configuration tasks

SSM operating in survivable mode only

The following steps describe the configuration tasks required for deployments in which SSM

operates only in survivable mode. This deployment is feasible if the SIP endpoints have the

capability to perform dual registration - either registering simultaneously with two servers or

registering with two servers one after another based on the server availability. In this case, the

SIP endpoints have the capability to monitor the availability of the servers.

1. Configure the SIP Media Gateway. SeeAvaya Secure Router 2330/4134

Configuration SIP Media Gateway(NN47263-508).

2. If the SSM and the SIP Media Gateway need to be bound to the same IP interface for

SIP traffic, you must specify separate ports to bind for the SSM and the SIP Media

Gateway. Avaya recommends to use the SIP default port 5060 for SSM. For the SIP

Media Gateway, change the port value of the bound interface from 5060 to a

different port (for example port 5070).

3. Configure the SSM:

Bind the IP interface for SIP traffic. If you are using the same IP interface as

the SIP Media Gateway bound interface, Avaya recommends to use well-

known SIP port 5060 for SSM and use a different port for the gateway (for

example port 5070).

Enable SSM (this step is required to before the remaining steps are allowed).



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Specify the domain name for the central SIP server.

Use thessm default-gatewaycommand to point the SSM to the SIP Media

Gateway IP interface (specifying the non-default port). This ensures that the

SSM routes the call to the gateway if the SSM is not able to find contacts for

the calling party in its local database..

4. Configure the SIP Media Gateway to point to the central call server using the sip-

ua sip-servercommand.

5. Configure the SIP Phones to point to the central call server.

SSM operating in normal or survivable mode

The following steps describe the configuration tasks required for deployments in which SSM

operates in normal mode or in survivable mode. In this deployment SSM monitors the

reachability of the call server using SIP OPTIONS requests. The SIP endpoints and SIP

gateway configure SSM as the outbound proxy. If the call server is not reachable, the SSMenters the survivable mode.

1. Configure the SIP Media Gateway. SeeAvaya Secure Router 2330/4134

Configuration SIP Media Gateway(NN47263-508).

2. If the SSM and the SIP Media Gateway need to be bound to the same IP interface for

SIP traffic, you must specify separate ports to bind for the SSM and the SIP Media

Gateway. Avaya recommends to use the SIP default port 5060 for SSM. For the SIP

Media Gateway, change the port value of the bound interface from 5060 to a

different port (for example port 5070).

3. Configure the SSM:

Bind the IP interface for SIP traffic. If you are using the same IP interface as

the SIP Media Gateway bound interface, Avaya recommends to use well-

known SIP port 5060 for SSM and use a different port for the gateway (for

example port 5070).

Enable SSM (this step is required to before the remaining steps are allowed).

Specify the domain name for the central SIP server.

Configure keepalives (using the keepalive-servercommand) to monitor

the central SIP server.

Configure Call Admission Control on the interface connecting to the SIP server

to limit the amount of voice traffic allowed on the WAN link. Configure exclude-pool to identify the branch SIP end-points. exclude-pool configuration is not

required when SR2330/4134 also functions as a WAN router.

Use thessm default-gatewaycommand to point the SSM to the SIP Media

Gateway IP interface (specifying the non-default port). This ensures that the

SSM routes the call to the gateway under the following conditions: in normal

mode, if the SSM is not able to find a registered contact for the caller for the

SSM basic configuration



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calls coming from the server; and in survivable mode, if the SSM is not able to

find contacts for the calling party in its local database. SSM also forwards the

call to the gateway when CAC validation fails for calls originated from the

branch.

4. Configure the outbound proxy on the SIP Media Gateway to point to the SSM IP

interface at port 5060 using the sip-ua outbound-proxycommand. This allowsthe SSM to serve as the proxy server for the Media Gateway.

5. Configure the outbound proxy on the SIP Phones to point to the SSM IP interface

at port 5060 also.

SSM basic configuration tasks navigation

Binding an IP address for SSMon page 27

Enabling SSMon page 28

Configuring the domainon page 29

Configuring keepalives for SIP survivabilityon page 29

Configuring the call admission control on an interfaceon page 30

Configuring the default gatewayon page 32

Binding an IP address for SSM

Use this procedure to bind the IP address for SSM. This IP address needs to be validatedagainst an existing IP interface address. Signaling and media uses this address as a sourceaddress. The application listens to this address for SIP signaling.

The configuration does not take effect until enableis executed. Before changing or removingan IP address or shutting down or deleting the interface, you must first disable SSM (ssmno enable).

By default, no IP address is configured.

Procedure steps

1. To enter configuration mode, enterconfigure terminal

2. To select VoIP service configuration, enter

voice service voip

3. To select SSM configuration, enter:

SSM basic configuration tasks navigation



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ssm

4. To bind an IP address for SSM, enter:

[no] bind ip ipv4:

Table 1: Variable definitions

Variable Value

[no] Removes the specified IP address

binding.

Specifies the IP address to bind for SSM.

Enabling SSM

Use this procedure to enable or disable the SIP survivability functionality. By default,survivability is disabled.

Prerequisites

You cannot enable SIP survivability until you bind an IP address for SSM using thebind ipcommand.

Procedure steps

1. To enter configuration mode, enter

configure terminal


voice service voip

3. To select SSM configuration, enter

ssm

4. To enable or disable SIP survivability, enter

[no] enable


Variable Value

[no] Disables the SIP survivability server.




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Configuring the domain

Use this procedure to configure the managed domain under which the SR2330/4134 isoperational. By default, no domain is configured.

Procedure steps


configure terminal


voice service voip


ssm

4. To select SIP server configuration, enter

sip-server

5. To configure the domain, enter

[no] domain


Variable Value

[no] Removes the specified domain.

Specifies the domain name as either ipv4:or dns:.

Configuring keepalives for SIP survivability

Use this procedure to configure the address of the central SIP server to which keepalivemessages are sent. It also specifies the interval at which OPTIONS messages are sent to thecentral server along with the number of retries before taking a decision on the change fromnormal mode to survivable mode. By default, keepalives are disabled.

Procedure steps


configure terminal


Configuring the domain



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voice service voip


ssm


sip-server

5. To configure keepalives, enter

[no] keepalive-server {dns:[:port-num] |ipv4:[:port-num]}

[interval ] [retries ]

[transport {tcp | udp }]


Variable Value

[no] Removes the specified configuration.

dns:[:port-

num]

Specifies the DNS host name of the central SIP

server.

ipv4:[:port-num] Specifies the IP address of the central SIPserver.

interval Specifies the interval in seconds at which tosend OPTIONS messages to the keepalive

server. Default value is 60.

retries Specifies the number of retries before changingfrom normal to survivable mode, and vice versa.

Default value is 1.

transport {tcp | udp } Specifies the transport protocol. Default valueis udp.

Configuring the call admission control on an interface

Configure call admission control to limit the number of calls that can exist simultaneously onan interface. There are a maximum of five interfaces that can be configured. By default, CAC is

disabled.

Procedure steps





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configure terminal


voice service voip


ssm

4. To configure the maximum number of calls on an interface, enter

[no] cac max-calls


Variable Value

[no] Deletes the specified value.

Specifies the interface to configure.

Specifies the maximum number of calls allowed

on the specified interface.

Configuring an exclusion pool for call admission control

With CAC, you can configure an exclusion pool that identifies the IP address range of the SIPendpoints that use SSM. This configuration is required only if the SIP endpoints and the Serverare connected to the same interface of the SR2330/4134.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure the exclusion pool for CAC, enter

[no] cac exclude-pool Table 6: Variable definitions

Variable Value

[no] Deletes the specified value.

Configuring an exclusion pool for call admission control



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Variable Value

Specifies the network assigned to the SIPPhones.

Specifies the network mask.

Configuring the default gateway

Use this procedure to configure the default gateway to be used by SIP survivability. By default,there is no gateway configured; however the default transport is UDP, and the default port is5060.

The default gateway IP address must be the same as the bind IP address used by the SIPMedia Gateway.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure the default gateway, enter

[no] default-gateway { ipv4:[:port-num] } [transport{tcp | udp }]


Variable Value

[no] Removes the specified default gateway.

ipv4:[:port-num] Specifies the IP address and optional portnumber for the default gateway.

[transport {tcp | udp} ] Specifies the transport used for the defaultgateway. Default value is udp.




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Chapter 5: SSM configuration examples

This chapter contains the following SSM configuration examples.

Basic SSM configuration exampleon page 33

SSM configuration example with existing LAN infrastructureon page 36

Basic SSM configuration example

The following figure shows a sample SSM network.



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Figure 8: Basic SSM configuration example

The following steps show the configuration required for the SIP Media Gateway and the SSM toprovide SIP survivability in the network shown in the preceding figure.

Prerequisites

Secure Router must be running minimum Release 10.2 software

Internal PVM (for Avaya Secure Router 4134) or PVIM (for Avaya Secure Router 2330)module must be installed

If more than 25 users are required, SSM license must be purchased and installed

Procedure steps

1. Configure the Ethernet interface for connection to the SIP server:

configure terminalinterface ethernet 0/1

SSM configuration examples



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ip address 20.20.20.1 255.255.255.0exit ethernet

2. Configure the Ethernet interface for SIP phone connectivity:

interface ethernet 0/2ip address 10.10.10.1 255.255.255.0exit ethernet

3. Configure a default route to the other end of the WAN:

ip route 0.0.0.0 0.0.0.0 20.20.20.2

4. Configure the SIP Media Gateway to listen on port 5070:

voice service voipsipbind all ipv4:10.10.10.1:5070exit sipexit voip

5. To configure the SIP Survivability Module, bind the IP interface for SIP traffic usingdefault port 5060:

voice service voip

ssmbind ip ipv4:10.10.10.1

6. Enable SSM:

enable

7. Specify the SSM domain name for the central SIP server:

sip-serverdomain dns:avaya.net

8. Enable SSM keepalives to monitor the central SIP server:

keepalive-server ipv4:30.30.30.1:5060transport udpexit sip-server

9. Configure SSM Call Admission Control on the WAN interface connecting to the SIPserver:

cacmax-calls ethernet0/1 50exit cac

10. Point the SSM to the SIP Media Gateway IP interface as the default gateway(specifying the non-default port):

default-gateway ipv4:10.10.10.1:5070exit ssmexit voip

11. Configure the outbound proxy on the SIP Media Gateway to point to the SSM:

sip-uaoutbound-proxy ipv4:10.10.10.1:5060

12. Configure the SIP Server for the SIP Media Gateway:

sip-server dns:avaya.netexit sip-ua

Basic SSM configuration example



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13. Configure the FXS voice port for the analog phone:

voice-port 2/1signal loop-startstation number 4087541000no shutdownexit voice-port

14. Configure the POTS dial peer for the analog phone:dial-peer voice pots 1destination-pattern 4087541000port 2/1forward-digits allno shutdownexit pots

15. Configure the outbound proxy on the SIP Phones to point to the SSM IP interfaceat port 5060 also.

SSM configuration example with existing LAN infrastructureThe following figure shows the basic SSM network configuration as shown in Basic SSMconfiguration example with NTMLon page 75, but in this case the Secure Router is operatingwith SSM and SIP Media Gateway features enabled, but without routing enabled. The initialconfiguration is similar to the previous example, however here, the bound IP must be the IPaddress of the Ethernet interface that connects to the branch LAN. Also, you must create anexclusion pool for CAC, so that incoming calls from the SIP phones on the Secure RouterEthernet connection are not counted toward the max-call limit.




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Procedure steps

1. Configure the Ethernet interface for connection to the SIP server:configure terminalinterface ethernet 0/1ip address 20.20.20.1 255.255.255.0exit ethernet

2. Configure a default route to the branch router:

ip route 0.0.0.0 0.0.0.0 20.20.20.2

3. Configure the SIP Media Gateway to listen on port 5070:

voice service voipsipbind all ipv4:20.20.20.1:5070exit sip

exit voip

4. To configure the SIP Survivability Module, bind the IP interface for SIP traffic usingdefault port 5060:

voice service voipssmbind ip ipv4:20.20.20.1

SSM configuration example with existing LAN infrastructure



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5. Enable SSM:

enable

6. Specify the SSM domain name for the central SIP server:

sip-serverdomain dns:avaya.net

7. Enable SSM keepalives to monitor the central SIP server:

keepalive-server ipv4:30.30.30.1:5060transport udpexit sip-server

8. Configure SSM Call Admission Control on the WAN interface connecting to the SIPserver:

cacmax-calls ethernet0/1 50exit cac

9. Configure the CAC exclusion pool:

exclude-pool 40.40.40.0 255.255.255.0exit cac

10. Point the SSM to the SIP Media Gateway IP interface as the default gateway(specifying the non-default port):

default-gateway ipv4:20.20.20.1:5070exit ssmexit voip

11. Configure the outbound proxy on the SIP Media Gateway to point to the SSM:

sip-uaoutbound-proxy ipv4:20.20.20.1:5060

12. Configure the SIP Server for the SIP Media Gateway:

sip-server dns:avaya.netexit sip-ua

13. Configure the FXS voice port for the analog phone:

voice-port 2/1signal loop-startstation number 4087541000no shutdownexit voice-port

14. Configure the POTS dial peer for the analog phone:

dial-peer voice pots 1destination-pattern 4087541000port 2/1forward-digits all

no shutdownexit pots

15. Configure the outbound proxy on the SIP Phones to point to the SSM IP interfaceat port 5060 also.




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Chapter 6: SSM additional configuration

This chapter provides additional optional procedures for SSM configuration.

Navigation

Loading and storing the dial planon page 40

Configuring the SIP transport and portson page 41

Configuring user provisioningon page 41

Configuring the SIP registrar for SIP survivabilityon page 44

Configuring the DNS timeouton page 45

Configuring service rejectionon page 45

Configuring the SIP timer valueson page 46

Configuring session timerson page 47

Configuring the SIP header valueson page 48

Configuring SSM user licensingon page 49

Configuring SSM congestion controlon page 50

Displaying the SIP survivability CAC configurationon page 52

Displaying the SIP survivability feature configurationon page 52

Displaying the SIP survivability dial plan filenameson page 52

Displaying the SIP survivability licensed user capacityon page 52

Displaying the SIP survivability protocol header configurationon page 53

Displaying the SIP survivability registered userson page 53

Displaying the SIP survivability registrar parameterson page 53

Displaying the SIP survivability session timer configurationon page 53

Displaying the SSM server configurationon page 54

Displaying the SIP survivability feature statuson page 54

Displaying the survivability SIP server statisticson page 54

Displaying the SIP survivability subscriberson page 55



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Flushing the SSM databaseon page 55

Clearing the SIP survivability feature statisticson page 55

Debug procedureson page 55

Loading and storing the dial plan

Use this procedure to load and store the preconfigured dial plan. By default, no dial plan isloaded or stored.

Procedure steps

1. To enter configuration mode, enter:

configure terminal

2. To select VoIP service configuration, enter:

voice service voip


ssm

4. To load the dial plan, enter:

[no] dialplan load {normal | survivable}

If the load command is entered again, it overwrites the currently loaded dial-plan

5. To store the dial plan, enter: .

dialplan store {normal | survivable} You must load a dial plan before you can perform a store operation.


Variable Value

load {normal |

survivable}

Loads either the normal or survivable mode dial plan

from the specified path. If the full path is not

specified, then the system searches for the specified

filename in cf0 (internal flash memory). The no form

of the command unloads the dial plan.

store {normal |

survivable}

Stores either the normal or the survivable mode dial

plan to the specified path. The no form of thecommand is not supported.

SSM additional configuration



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Configuring the SIP transport and ports

Use this procedure to configure the port and the associated transport protocol on which theSSM accepts SIP requests. By default, SSM listens on both the transport protocols: UDP andTCP. Each port is validated against an existing port used in the local SIP IP port. If the twoports are the same and have the same IP, a warning message is displayed requesting you toreconfigure the SIP IP port.

The configuration does not take effect until ssm enableis executed. Once SSM is enabled, tomodify the transport configuration, you must first enter the no enablecommand under thessmtree.

The default port value for TCP and UDP is 5060.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure SIP transport parameters, enter:

[no] bind transport


Variable Value

[no] Restores binding parameters to the default values.

Specifies the transport protocol: udp, or tcp.

Specifies the port to bind. Range is 1024-65535.

Configuring user provisioningBy default SSM learns the subscriber details from SIP REGISTRATIONS. It also provides astatic method to provision subscriber information. The subscriber information used by SSMincludes user name, domain, alias and identity.

Configuring the SIP transport and ports



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Use this procedure to configure subscriber details. It is mainly used to configure aliasinformation.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure user provisioning for SIP survivability, enter

[no] provisioning subscriber alias calling-line-id


Variable Value


Specifies the user name of the subscriber.

Specifies the domain name of the subscriber informat of ipv4: or dns:

Specifies the alias name.

Specifies the calling line ID.

Configuring digest authentication

Use this procedure to configure digest authentication for subscribers in backup mode.

Procedure steps

1. To enter the configuration mode, enter:

configure terminal2. To select VoIP service configuration, enter:

voice service voip





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ssm

4. To configure authentication options, enter:

[no] digest-auth enable


Variable Value

enable Enables digest authentication.

[no] Disables digest authentication.

Configuring digest authentication user credentials

Use this procedure to configure user credentials for digest authentication.

Procedure steps

1. To enter the configuration mode, enter:

configure terminal


voice service voip


ssm

4. To configure subscriber credentials for authentication, enter:[no] provisioning subscriber-credentials


Variable Value

[no] Removes the user credentials configuration.

Specifies the user name of the subscriber.

Specifies the domain name of the subscriber

in format of ipv4:or dns:.

Specifies the user name for digest

authentication.

Specifies the password for digest

authentication.

Configuring digest authentication user credentials



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Configuring the SIP registrar for SIP survivability

Use this procedure to configure the SIP registrar for SIP survivability.

Procedure steps


configure terminal


voice service voip


ssm

4. To select SIP registrar configuration, enter:

registrar

5. To configure the range within which the expires header value is accepted by theSSM registrar, enter:

[no] expires {max|min|default}

6. To configure the maximum contacts that can be registered for a single AOR, enter:

[no] max-contacts

7. To configure registration policy options in backup mode, enter:

[no] policy {all | provisioned-users}


Variable Value

[no] Removes the specified registrar configuration.

expires {max|min|default} Specifies the range within which the expires header

value is accepted by the survivable SIP registrar. The

no form of the command sets the values for

maximum, minimum and default to their default

values (180, 180, and 180, respectively).

max-contacts Specifies the maximum contacts that can be

registered for a single AOR. The no form of thecommand sets the value to the default: 5.




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Variable Value

policy {all | provisioned-users Configures registration policy in backup mode.

allallows all registrations.

provisioned-usersallows registration from

provisioned users only.

Default value is all.

Configuring the DNS timeout

Use this procedure to configure the DNS lookup timeout in milliseconds after which DNSlookups attempted by the proxy must timeout. By default, the DNS timeout is 86400milliseconds.

Procedure steps


configure terminal


voice service voip


ssm


sip-server

5. To configure the DNS timeout, enter .

[no] dns-timeout


Variable Value

[no] Restores the DNS timeout to the default value:86400 milliseconds.

Configuring service rejection

Use this procedure to configure the SIP server to reject SIP service messages in the survivablemode. By default, the SIP server rejects SIP service messages in survivable mode.

Configuring the DNS timeout



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Procedure steps


configure terminal


voice service voip


ssm


sip-server

5. To configure service rejection, enter

[no] reject-services-in-survivable-mode


Variable Value

[no] Activates the forwarding of service related toSIP messages.

Configuring the SIP timer values

Use this procedure to configure the SIP timer values within the SIP server.

Procedure steps


configure terminal


voice service voip


ssm


sip-server

5. To configure SIP timer values, enter

[no] timer {T1|T2|B|C|D|F|H|I|J|K}




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Variable Value

[no] Resets the specified timer to the default values:

T1: 500 ms

T2: 4000 ms

B: 32000 ms

C: 180000 ms

D: 32000 ms

F: 32000 ms

H: 32000 ms

I: 5000 ms

J: 32000 ms

K: 5000 ms

T1, T2, B, C, F, H, I, J, KSpecifies values between 1-2147483647.

C Specifies a value between 180000-2147483647.

D Specifies a value between 32000-2147483647.

Configuring session timers

Use this procedure to configure session timer values in seconds. By default, session timervalues are max: 3600, min: 90, and default: 1800. Be sure to configure the min value to beless than the default value, and the default value to be less than the max value, otherwise theconfiguration is rejected.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure the session timer, enter:

Configuring session timers



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[no] sessiontimer session-timer {max|min|default} ]

5. To configure the session timer period validation, enter:

[no] sessiontimer range-validation


Variable Value

[no] Sets the specified parameter to the default value.

range-validation Turns on the session timer period validationrequested by endpoints. The no form of the

command resets the parameter to the default

value, that is, to not validate the session timer

range.

session-timer {max|min|

default}

Specifies the session timer to configure.

Specifies a value in seconds for the sessiontimer. Range is 904294967295.

Configuring the SIP header values

Use this procedure to configure the SIP header values.

Procedure steps


configure terminal


voice service voip


ssm

4. To select SIP protocol header configuration, enter:

protocol-header

5. To configure the retry-after interval, enter:

[no] retry-after-interval

6. To configure the organization header, enter:

[no] organization-header

7. To configure the server header, enter:




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[no] server-header


Variable Value


retry-after-interval Specifies the retry after interval value (in

seconds) to be inserted into the retry after SIP

header. The no form of this command sets the

interval to the default value (300 seconds).

Use 0 to omit the retry after SIP header.

organization-header Specifies the organization name to be inserted as

organization header into the messages

generated by the SIP server. The no form of the

command removes the organization header.

There is no default value.

server-header Specifies the string to be used in the server

header in responses generated by the SIP server.The no form of this command removes the server

header. There is no default value.

Configuring SSM user licensing

Use this procedure to load an SSM license key onto the SR2330/4134 to support additionalSSM users.

Procedure steps


configure terminal

2. To specify the number of users the license key supports, enter:

system license ssm_users

The system then prompts you for the license key.

3. When prompted, enter the license key. If the key is valid, the system prompts youto reboot.

4. After the reboot is completed, to verify the updated number of supported SSM users,enter:

show ssm licensed-users

Configuring SSM user licensing



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Configuring SSM congestion control

SSM implements a mechanism to perform SIP message load control and throttling based onits memory consumption. SSM monitors the memory it utilizes and takes appropriate actionsbased on the percentage of memory consumption. There are three threshold levels defined.

Associated with each level are two trigger values: up and down.

Threshold 1: The percentage of memory consumption after which SSM starts rejectingall new requests which are not part of an existing dialogue with "503 Service Unavailable"responses. However, all requests within the dialogue are processed normally.

Threshold 2: The percentage of memory consumption after which SSM starts droppingall new requests which are not part of an existing dialogue. While in this state, SSM doesnot send any response to these requests. However, all requests within the dialogue areprocessed normally.

Threshold 3: Defines the percentage of memory consumption after which SSM stopsreceiving SIP messages from all network interfaces. In this state, SSM only sendsmessages out to the network and does not accept any messages from the network.

Use the following procedure to configure SSM congestion control.

Procedure steps


configure terminal


voice service voip


ssm

4. To configure congestion threshold levels, enter:

[no] congestion-thresholds memory

5. To verify the configured values, enter:

show ssm congestion-thresholds


Variable Value

[no] Resets the threshold levels to the default values:

up-threshold1: 75

up-threshold2: 85

SSM a

nn47263-510 02.01 config sip survivability

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