Nortel Networks NN10029-111 User Manual

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NN10029-111

Succession Multimedia Communications Portfolio

MCP SIP Application Module

Basics

Standard MCP 1.1 FP1 (02.02) April 2003

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Overview

How this chapter is organized

The SIP Application Module Overview contains the following subsections:

• “Overview” on page 4 — Functional desc ription — Network configuration — Interfaces —Protocols

• “Hardware” on page 11

• “Services and features” on page 11 — “Routing and Translation services” on page 12 — “Interworking services” on page 16 — “Service package enforcement” on page 17

— “Authentication services” on page 17 — “Network/Address Hiding service” on page 19 — “911 Notification support” on page 21 — “Instant Messaging” on page 22 — “Presence” on p age 22 — “Voicemail server intero perability and MWI” on page 22 — “Registration—static and dynamic” on page 24 — “Network address book” on page 25 — “Overload contro l ” on page 25 — “Reliability and fault tolerance” on page 26

• “OAM&P strategy” on page 28

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4 Overview

Overview

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The SIP Application Module is a service execution engine that provides the following functionality:

• core signaling functionality enabling communication among SIP clients

• SIP proxy server

• Back-to-Back User Agent

• SIP Registration

• CPL interpretation

• Location server

• optional Presence subscripti on and notification (For more information on the Presence feature, see the MCP SIP Presence Basics document.)

The SIP Applicatio n Module handles SIP sessions an d applications and provides the core services that enable communication between SIP clients. The SIP Application Module is housed on the SIP Application Server.

Functional description

The SIP Application Module includes the following components:

• Back-to-Back User Agent (BBUA)/Proxy Server Although the BBUA and Proxy Server are basically two differen t

logical entities within the same physical server, they both act as clients and servers. The SIP Application Module decides on a call-by-call basis wh ether to process the req uest as a pure Proxy or BBUA.

The Proxy Server processes SIP requests and responses, re writes headers, modifies requ est- U RIs (U niver sal Resource Indicator), performs locati on look-up, and forwards reques ts to SIP clients or other servers in the network.

The SIP Application Module provides a fully session-stated proxy; in other words, the SIP Application Module main tains a call state for the entire session.

The BBUA extends the proxy function to perform advanced functions such as

— originating new calls — tearing down existing calls — modifying messages

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— changing IP addresses in the contact header so that the SIP

— modifying the Sessi o n De scription Protocol (SDP) using va l ue s

— providing advanced screening capabilities The architecture of a BBUA service consists of two user agent

clients linked back-to-back through a proprietary interface. The BBUA is guaranteed to be on the signaling path of all future

requests and responses because it is an endpoint relative to the SIP network component s. Th i s is i m po rtant for services such as billi ng , which need to be aware of all events that take place on a session. The BBUA in the netwo rk also provi des a barrie r for cli ents th at are not fully SIP compliant and entry and exit points for traffic travelling to and from the public network, including agents behind an enterprise firewall. See Figure 1, “Back-to-Back User Agent service.”

Figure 1 Back-to-Back User Agent servic e

Overview 5

Application Module remains on the signaling path

supplied by the RTP Media Portal to control media endpoints

userA

User Agent Client

userB

Internal Protocol

Routing in a S IP network is based on the same hop- by-hop principle as routing e-ma il within the Internet. Th e next hop for a S IP request is determined by a pr oxy using th e domai n or the host p art of a SIP URL (user@domain). The terminating proxy determines whether the domain sent in the SIP URL is one of the domains managed by the SIP proxy. Otherwise, the SIP request is forwarded to another Proxy based on the location lookup performed by the SIPApplication Module. The SIPApplication Module supports routing using t able l ookup i n the SIP dat abase o r using th e Domain Name Server (DNS) to find a route.

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• Redirect Server The SIP Application Module decides whether to proxy or redirect the

call separately for each individual request. This decisio n is made based on subscriber service logic. If the decision is to redirect the request, a 302 Resp onse message is returned with a list of alternate locations.

• Registration Server The Registration Server performs registration on messages it

receives from clients. The Registration Server stores information in the database.

• Location Server The Location Server performs location lookup services using

domain and user information stored in the database. The SIP Application Module integrates the above logical servers,

which are all defined in SIP Draft RFC 2543 (see note for specific reference), in to a single server wi th the enhanced se rvices provided by the Back-to-Back User Agent.

Note: J. Rosenberg et al, SIP: Session Initiati on Protocol, Internet Draft draft-ietf-sip-rfc2543-bis09.txt, IETF, Feb 27, 2002.

Network configuration

The SIPApplication Module is configured with two network cards to allow for a network configuration that has a private side and a public side. Figure 2, “Example of network configuration,” shows the SIPApplication Module and RTP Media Portal with public ports and ports that are internal to the priv ate network. This network co nfiguration provides security by placing all the components in a private network and exposing only the public signaling and ports to the public network.

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Figure 2 Example of ne twork configuration

SIP Application

BigMart.com

Module

BiggerMart.com

SIP Audio Servers

Overview 7

Management Module

PSTN Gateways

Accounting Modules

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8 Overview

Interfaces

The SIP Application Module interfaces with numerous other components. See Figure 3, “Network interfaces.”

Figure 3 Network interfaces

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Private network

PSTN or PBX

Access client

Public network

Access client

Legend

Signaling Media stream

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Protocols

The SIPApplication Module uses various protocols to support SIP clients, including the Management Module, RTP Media Portal, Database Module, and the PSTN Gateways. The protocols use an IP backbone to connect the components. These interfaces are shown in Figure 3, “Network interfaces.”

Figure 4 Protocols

Overview 9

OMI

PCP

SIP

SQL

SIP

Access client

DTP

Private network

SIP

Public network

SIP

MGCP+

SIP

Access client

SIP

PSTN or PBX

Legend

Signaling Media stream

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Table 1, “SIPApplication Module protocols,” gives details about these interfaces.

Table 1 SIP Application Module protocols

Protocol Functional component

SQL Structured Query Language

Interface to the Database Module

SIP Session Initiation Protocol

Interface to the

•SIPclients

• SIP Audio Server

• IP Client Manager (IPCM)

• SIP PRI Gateway

• Web Client Manager

OMI Open Management Interface

Interface to t he SIP Management Module

MGCP+ Media Gateway Control Protocol

Interface to the RTP Media Portal

PCP Perfect Channel Protocol for logs and a larms going to

the Management Server DTP Data Tran sport Protocol Note: The external interfac es use an IP network to int erconnect the

components listed in this table.

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Hardware

Refer to Table 2, “Minimum hardware requirements,” for the l ist of required hardware.

Table 2 Minimum hardware requirements

Sun Netra t 1400(DC) /1405 (AC) Description

4-440 Mhz Ultra Sparc II CPUs 4 GB RAM 2-36 GB Ultra SCSI disk drives 1-32X Internal CDROM drive (bootable) 24 GB 4 mm internal tape drive 1 Quad Fast Ethernet PCI card 1 PCI UltraSCSI card

Overview 11

Services and features

The SIP Application Module performs the following services:

• Routing and Translations Services — Call Transfer — Local termination — Foreign termination —Redirect — Telephony Routing — SIP Aliases — Multiple Route Termination/SIP Forking feature — Call Processing Language (CPL)

• Interworking services — Discriminator service

AC (t 1405)/DC (t 1400) power supplies

— Bearer Path Control — Privacy Control service

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Routing and Translation services

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• Service package enforcement

• Authentication services

• Converged PC service

• Network/Address Hiding

• 911 Notification support

• Instant Messaging

• Presence

• Voicemail server interope rability and MWI (message waiting indication) notification

• Registration

• Network address book

• Overload control

• Reliability and fault tolerance

Foreign termination

If an incoming req uest specifies a do main that is not ser ved by (in other words, is not local to) the SIP Application Module, the SIP Application Module tries to route that request to the appropriate server for that domain.

The first step in this process is to query the DNS SRV, if one is configured in the system, in ord er to obtain the IP address of the server associated with the foreign domain.

Note: A DNS SRV extends the basic functionality provided by a traditional domain name server (DNS). It allows a protocol field to be the query fo r a particular domain and uses that protocol field to provide the correct IP addr ess of the server for the specifie d protocol. For example, clients may query the server with a domain name of nortelnetworks.com and protocol field of sip. The DNS SRV would then respond with the IP address of the SIP server for that domain (which may dif fer from, fo r example, t he H.323 serve r). This all ows a domain to have different servers for different protocols.

If this query fails to find the IP address or if a DNS SRV is not configured, the SI P Appl icatio n Mod ule atte mpt s to lo ok up t he for eig n domain in the database to see if an IP address has been provisioned for this foreign domain (see the SIP Provisioning Client User Guide for

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details). If this step also fails, the SIP Application Module attempts a general DNS A-record lookup to route the request.

Note: The DNS A-recor d is the traditiona l response given by a DNS. It translates a domain name into an IP address.

If any of these steps succeed, the SIP Application Module routes the request. If all these methods fa il, the SIP Application Module reject s the request.

Call Transfer service

The SIP Application Module handles the transfer on behalf of clients that do not support the call transfer service.

The SIP Applicatio n Module supports unattended Call T ransfer throug h the Refer mechanism. Unattended Transfer (or Blind Transfer) refers to cases where the transferor redirect s the transferee to the tr ansfer target without first con fer ri ng wi th th e tr a nsfe r target. The transfer or receives a Notify message, however, indicating whether the transfer was successful. If it was, the transferor releases the original call. If it was not, the transferor is reconnected to the transferee.

Overview 13

Local termination

The SIPApplicat ion Module fir st determines w hether the incom ing SIP request terminates to a client in a domain managed by the SIPApplication Module. The SIP Application Mod ule performs local routing lookup through the Location Server, which is part of its internal software.

Telephony routing

When the SIP Application Module receives an incoming call, it looks up the callee in the database. If the callee is not in the database but the domain is served and the user portion of t he URL is a T elep hony routing number, the Telephony routing number is sent through the Telephony routing software within the Location Server.

The Te leph ony r outin g s oftware must perfor m digi t tra nslat ion to fin d a gateway to terminate a call to. These t ables are located in the Database Module. You can provision them through the Provisioning Client. For more information, refer to the SIP Provisioning Client User Guide and the MCP Database Module Basics document.

The Telephony routing service allows the SIP Application Module to

• provide unique dial plans for each subdomain

• provide routes to gateways or to other domains

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These routes include routes for private digit dial plans, routes to gateways, and telephony-style routing between SIP domains. Multiple lists can reuse the same routes in a route list.

• assign class of service (COS) COS is basically used to block particular types of calls, such as

international dialing or long-distance dialing. For example, telephones in an office lobby can be restricted to local and emergency calls only in a domain.

Figure 5, “Relationship between Telephony routin g st ag es,” shows th e relationship betwee n the tele phony ro uting sta ges provi ded by the S IP Application M odule. If the C OS val ue of the subscri ber and subdom ain route do not match, then the SIP Ap plication Module checks the pa rent for routes with the same or higher COS value.

Figure 5 Relationship between Telephony routing stages

No subscriber

Foreign domain routing Do database lookup

Failed

*If the DNS A record fails, a 404 Error response is sent back to the originator.

DNS A record*

Route lists The T elephony routing service is an enhancement to the Location Server on the SIP Application Module. This enhanced Location Server function has the ability to translate PSTN numbers into

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URL addresses specifying an appropriate gateway . It supports the use of digit translation and digit manipulation.

A route list is assigned a single COS. The route list provides the following additional options that can restrict incoming sessions from using the domain’s telephony resources:

• allow/block all incoming sessions from other domains

• allow/block all incoming sessions from other subdomains

• redirect session to the orig inator’s domain. This option can be used to redirect an incom ing reque st from ano ther domai n that is r outing to a restricted route list.

Route lists consist of

• private telepho ny routes, which a re used for privat e telephony-styl e digit dial plans

• gateway routes, which provide access to the gateways

• SIP telephon y routes, which point to oth er SIP Application Mo dules, and SIP domains and subdomains for interdomain routing using telephony-style dial plans

Overview 15

SIP Aliases

Alias URLs can be used to refer to a SIP client in the network. For example, a user “sip: u serA@domainX. com” can also be ref erred to by an alias of “sip:41037@domainX.com”.

If an incoming request specifies the “sip:41037@domainX.com” alias in a Request-URI, the alias takes precede nce over gateway ro uting translations, and r ou ting information pert aini ng to use rA i s r etri eve d. I f an alias of “sip:410 37@domain X.com” is not confi gured, then g ateway routing translations are performed to find out if a terminating gateway exists.

Multiple Route Termination

If a single SIP user is registered at more than one device (PSTN or SIP), forking is used to terminate a session simultaneously or sequentially to multiple devices.

The SIP Application Module interfaces with the SIP database to determine the use r routing preference, the r outes available, and ro uting options for a particular user. The user defines these options through the SIP Personal Agent. For additional information on the SIP Personal Agent, refer to the on-product help and SIP Personal Agent Getting

Started Guide.

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Call Processing Language

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With simultaneous ringi n g, the call terminates to multiple routes at the same time. The first terminating route to answer is accepted and the rest of the routes are released.

With sequent ial ringi ng, the call tries to terminate to on ly one of several routes at a time. Route advancement occurs whenever an error response is received, a provisionable No Answer timer expires, or a redirect response is received.

The SIPApplication Module supports the use of the Call Processing Language (CPL), based on the IETF CPL draft, draft-ietf-iptel-cpl.txt. SIP clients can ch ange the behavior of a session using a CPL script that contains general directives for routing a request.

For example, subscribers can include CPL scripts in the body of registration requests that contain instructions for location lookups and call screening, a pr ocess that is actually done th rough the Call Manager in the Personal Age nt. Third-p arty client s can also up load script s using the Registration mechanism. The Registration function of the SIPApplication Module stores the request. When the SIPApplication Module is queried for routing information for a subscriber who has valid data stored in the database, the software returns the script along with the routing information. The SIPApplication Module applies the CPL script to the returned routes and can eliminate or alter the routes based on the CPL script.

CPL scripts do not support the following:

• Remove location

• Mail option

• Log option

Interworking services

Discriminator service

The SIPApplication Module screens requests bound for devices that are not fully SIP compli ant, for example, the Communication Server for Enterprise (CS E) 2000. These comp onents canno t process all types o f signaling and certain media change requests. Therefore, the SIPApplication Module either performs the requested operation or rejects the request and responds with an error response.

The Discriminator service works with various gateways and SIP clients using provisioning facilities implemented by the SIP Application Module. As gateways or SIP clients with limited SIP capability are added to the network, this service can be configured to support these

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devices. Information for each component is stored in .xml format to provide flexibility when describing the capabilities o f the component.

Bearer Path Control

The SIPApplication Module uses the RTP Media Portal to control media streams originating from and terminating to non-compliant SIP devices if they d o not support media negotiations. The ex ception to this occurs when the o rigin at ing a nd ter minat in g p art ie s are both th e sam e device type. If both gateways are CSE 2000s, for example, the SIP Application Module does not use the RTP Media Portal.

Privacy Control service

The SIP Application Module supports Privacy Control based on draft-ietf-sip-privacy. This draft defines a mechanism that allows clients to supply a network server with their private user information while at the same time instructing the server no t to pass that information outside the boundaries of the truste d network. The information is passed in a Remote-Party-ID header with the privacy indica tor set to “full.” The SIP Application Module removes this header any time it forwards the message out over a public network interface.

Overview 17

Service package enforcement

A service package is made up of a user’s enabled network services, such as audio conferencing, and subscriber profile. The service provider defines the available service packages for the domain. The domain provisioner can then assign a specific service package to a subscriber.

Authentication services

The SIP Application Module performs user authenticat i on when the server receives an incoming SIP request. The SIP Applicatio n Mo dule supports the challenge-based Digest method for SIP Client-to-Proxy authentication. In Digest authentication, the SIP Application Module challenges a client when a SIP request is received. The SIP Client re-sends a SIP requ est with a valid password an d user name att ached. The request types to be authenticated are configurable.

Note: Only US ASCII is supported for user names.

The software performs authentication using the password of the subscriber originating the call. Only subscribers from a local domain actually have a password stored in the database to authenticate against. If a subscriber from a foreign domain (refer to t he no te below for definitions of these types of domains) places a call and authentication is required for a known foreign domain, the

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Converged PC service

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authentication fails since the database does not have the subscriber's information. As a result, the call is blocked.

Administrators can configure whether they want a call from an unknown foreign domain authenticated or not. System administrators can also specify foreign proxie s in the Nodal Auth fiel d of the Authenticati on t ab. In this way, no requests originating from those proxies are failed because of authentication.

Note: The following definitions apply:

• Local Domain: Local domains are provisioned for and serviced by a particular SIP Application Module. Su bscribers for a particular system belong to local domains. Local domains are provisioned through the Provisioning Client.

• Foreign Domai n: A foreign domain is a domain that is eithe r provisioned as foreign for this SIP Application Module or not provisioned at all for this specific system. It b asically represents a domain that is not served.

The Converged PC service allows end users to use their PCs for the multimedia portion of their communications while using their existing telephony system for voice. The service uses the simring feature on an existing telephony system to send mirrore d calls to the SIP Application Module through the SIP PRI Gateway . This allows the SIP Application Module to presen t a ca ll wi ndow o n the end user 's PC w hen t he use r's desktop phone rings.

If both parties in a call are Converged users, they will each get a call window from which they can initiate multimedia sessions such as Instant Messaging and collaborative applications between each other.

Some benefits of providing multimedia services using the Converged service are:

• End users can keep using their existing telephone and its capabilities.

• There is no need to replace an existi ng telephony switch to add multimedia capabilities.

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The Converged service adds the following cap abilities to the end user's telephony service:

• the ability to manually redirect incoming calls to another party from the PC

• the ability to set up automated enhanced routing and screening of incoming calls based on time of day or based on the calling party's identity

• a call log of all incoming calls

• the ability to send instant messages to the party on the other end of a call

• the ability to start collaborative applications such as shared whiteboard, file transfer, and clipboard transfer with the party on the other end of the call

• the ability to receive a picture ID of the party on the other end of the call

Network/Address Hiding service

The SIP Application Module uses SIP and the Session Description Protocol (SDP) to coordinate the establishment of multimedia sessions for signaling and media, respectively. These protocols embed IP information in their messaging. While Networ k Address Translation (NAT) devices change port and address informati on in the IP packet header , most are not currently SIP or SDP aware. IP addresses in these messages are theref ore sent out unchange d through the NA T. If the SIP Application Mo du le were to for w ar d the se messa ge s on uncha nged, sensitive IP inform ation w ould be gi ven to untr usted clien ts. In or der to remedy this, the SIP Application Module sanitizes the messages before forwarding them.

Overview 19

For IP information in the SIP headers, the SIP Application Module either removes the header (for example, Via headers) or replaces the IP address with the address of the SIP Application Server (for example, Contact head er). A media portal is necessary in order to re place the IP information in the SDP headers. The SIP Application Module queries the Media Port al (u si ng M GC P+) for a new IP and port combi n ati o n to replace the IP and port put th ere by th e client . This ef fe ctively an chors the media stream at the Media Portal.

Clients therefore see the SIP Application Module as their signaling endpoint and th e Media Port al as their R TP media en dpoint. They have no knowledge, and therefore no IP information, about the other client they are in a session with.

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The RTP Media Portal handles Network Hiding for the media stream. For information on the R TP Media P ortal, re fer to the MCP RTP Media Portal Basics document.

Note: The SIPApplication Module cannot map SDP information without an RTP Media Portal. It only performs address mapping for SIP header fields. Therefor e, SDP p asses thr ough unto uched. If the server must map SDP address information, then you need an RTP Media Portal.

The SIPApplication Module is configured to use an RTP Media Portal to originate and terminate media streams (RTP/RTCP). The SIPApplicati on Mod ul e uses exte nd ed Me di a Gatew ay Co ntr o l Protocol (MGCP+) to allocate and release resources on the RTP Media Portal for each session as needed.

Enterprise Clients

The SIPApplication software uses the RTP Media Portal to hide sensitive IP address information about SIP clients behind a firewall in an Enterprise netw ork. The exception to th is occurs when the orig inator and terminator of the request are both part of the same network. This status is de ter min ed b y ch ecki ng th e do mains in the From head er a nd Request-URI of the SIP Invites. If both SIP clients belong to the same Enterprise netw or k, th e S IP Ap pl i cation Module does not u se th e RTP Media Portal. Adm inistrators can overri de this behavior by provision ing the AlwaysUseMediaPortal domain parameter in the Provisioning Client (for more information about this parameter, see the SIP Provisioning Client User Guide). See Figure 6, “RTP Media Portal interworking with Enterprise or foreign clients.”

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Overview 21

Figure 6 RTP Media Portal interworking with Enterprise or foreign clients

Public Domain

SIP Client

SIP

RTP/RTCP

911 Notification support

The SIP Application Module supports Instant Message notifications to a specified On-Sit e Notification (OSN) location whenever a user makes a call to an emergency number such as 911. The software provides this service using the same mechanis m that allows us ers to push web pages and/or email links back to the originator of a call. In order to do this, administrators set up (at the Personal Agent) an emergency subscriber for each OSN location and a private telephony route to map the emergency number to this subs criber. Since telephony routes are only unique w ithin a sub doma in, you canno t have mo re t han one OSN location for each subdomain.

Enterprise Domain

SIP Client

SIP

MGCP+

RTP/RTCP

RTP/RTCP Media Portal

For each new emergency subscriber that the administrator creates, there must be both

• an emergency numbe r to route to the Public Safety An swering Point (PSAP)

• a SIP subscriber assigned t o the OSN loca tion that is to rece ive the notification.

Each OSN location must have a specific subscriber assig ned, such as sip:guarddeskA@nortelnetworks.com.

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Instant Messaging

Presence

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For more information a nd the procedure for setting up In stant Message notifications to emergency numbers, see the SIP Provisioning Client

User Guide.

Instant Messages are routed in parallel only to a subscriber' s dynamically registe r ed r ou tes ( see “Reg i str ati on —static and dynamic” on page 24). This is in contrast to session initiation requests, whi ch are subject to CPL routing logic. Upon receipt of an instant message, a client may respond back to the address supplied in the Contact header. This ensures that the response is sent back to the same client device that originally sent the message .

When a user initially reg isters, by default th eir presence st atus is set to “on-line” in the SIP re gistration message . Users subscribe to watch the status of other use rs, and to coordinate the status of their own devices. This information is maintained in an in-memory table on the SIP Application Mo dule (Presen ce software). The informatio n that is stor ed in this table includes:

• the user to be watched

• the party reque sting the subscription

• the correlation informatio n identifying that particular subscription request

• contact information regarding where to send the notifications that are generate d as a result of the subscription being active

When a user chan ges their presence (for example, to Busy), a registration message is automatically sent to the SIP Application Module.

The SIP Application Module then checks its in-memory table to see what their previous prese nce state was. If the update causes a m aterial change in their presence state, the SIP Application Module looks up which users need to be notified of the change (also in memory). This is done by sending a Notify message to each user at every contact contained in the table. For more information, refer to the MCP SIP Presence Basics document.

Voicemail server interoperability and MWI

In order to accomplish voicemail server interoperability and MWI (message waiting indication) notification, the SIP Application Module

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transmits the following information over a data link to a voicemail server:

• the called number (terminating party's telephone number)

• the calling numb er

• the type of call forwarding (for example, due to a busy line, an unanswered call)

This feature also provides an interface to pure IP solutions that use a SIP-enabled voicemail server. In this case, SIP messages provide the context data for each call needed by the voicemail server to record a voicemail message. Thus, a SIP-enabled voicemail server accepts Invites for calls routed to voicemail and sen ds Notify messages for MWI information. The softwar e uses Real -T im e Transport Protoc ol (R TP) to carry the voice media.

There are two co nfigurations throu gh which the SIP Ap plication Module supports voicemai l:

• A pure IP, third-party, SIP-enabled voicemail server that uses RTP to establish the voice path from the subscriber to the voicemail server while SIP provides the setup and MWI information.

Overview 23

• A legacy voicemail server that uses a SIP/PSTN gateway to establish the voice path from the subscriber to the PSTN-based voicemail server. The Simplified Message Desk Interface (SMDI) protocol provides the setup information. The platform uses any voicemail server that suppor ts the SMDI protocol. There are two supported physical connections: a line-based gateway and a PRI/T1-based gateway.

Using either of the above configurations, there are three primary scenarios that this feature co nsiders:

• MESSAGE DEPOSIT: An incoming call for a subscriber gets rout ed to voicemail because the called subscriber is unavailable, busy, or has all calls forwarded to voi cem ai l.

• MESSAGE NOTIFICATION: The voicemail server sends an MWI status update to the SIP Application Module for a particular subscriber. The SIP Application Module then sends a message to the client(s) to update its MWI display.

Note: Clients do not store the MWI state. Only the Presence Module stores the state. When a client registers with the proxy

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and has messages waiting, the system sends a Notify to the client.

• MESSAGE RETRIEVAL: A subscriber calls the voicemail server for message retrieval. The subscriber is then connected to the voicemail server and accesses the mailbox to retrieve messages.

When you provision the voicemail server, specify which SIP Application Module is the host (see the Configuration chapter in this docume nt for details). Only the SIP Application Module that is hosting a particular voicemail server attempts to establish an SMDI connection with that voicemail server.

Note: SMDI is used in certain voicemail configurations to allow the voicemail server to send Message Waiting Indication information to the SIP Application Module. Also when connected to a lines-based voicemail server, the SIP Application Module sends an SMDI message to the voicemail server when a call is being routed to voicemail for messag e deposit. The SMDI information includes which mailbox the message should be deposited in. Also, the voicemail server periodically sends an SMDI heartbeat message to the SIP Application Mo dule. The SIP Application Mod ule must respond to this message to let the voicemail se rver know that the SMDI link is still up.

Registration—static and dynamic

Registration can take two forms:

•Static Users or administrators can perform st atic registra tions. With st atic

registration, the user can obtain a presence when not logged into the network. The user can obt ain a presen ce and an accoun t in one of the following ways:

— Using the SIP Provisioning Client, the administrator can add a

user account and assign a static route.

— When users have accounts, they can add contact information,

such as PSTN numbers or cell phone numbers, to their routing information.

•Dynamic Once a user logs in, re-registration is automatic with the SIP

Multimedia PC Client, the SIP Multimedia Web Client, and the IPCM. The IPCM takes car e of this re-reg istr ation aut omati cal ly for the i2004. Dynamic registration is automated and behind the scenes.

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Network address book

Client Address Book information is stored in the network so that it can be accessed from all clients. The information is downloaded in bulk whenever a client comes on line (either through a Simple Object Access Protocol [SOAP] int erface or direct database access depend ing on the client).

In order to receive updates to the Address Book after the initial download, the c lient subscribe s to the Address Book event package and updates it as needed. Whenever an update is made through the Personal Agent or one of the clients, a Notify message is sent to the client indicating which entries have changed. The client can then incrementall y update their view of the info rmation (aga in either thro ugh a SOAP interface or direct database access depending on the client).

A List of Buddie s is incorporate d as part of the Address Book. Each subscriber must create th eir own personal Address Book an d designate their own Buddies. For each of these specified entries, the client automatically subscribes to their pre sence eve nt packa ge. Thi s allows them to monitor and update the network presence of each Buddy (for example, online or offline).

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Overload control

Overload Control monitors the Incoming Protocol Message Queue Length. If this queue l ength crosses a configurable threshold value, the system performs Session Blocking, allowing no new incoming request s to process. The system does, however, continue to process requests for an established session. For rejected requests, the system sends a “503 Service Unavailable” response with a Retry-After header, which specifies the amount of time a client should wait before retrying the request.

Note that multiple thresholds may be crossed simultaneously. If this occurs, the appropriat e actions are invoked and are not cleared until al l aspects of the system have crossed below the assigned threshold value.

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Reliability and fault tolerance

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The SIP Application Module provides reliability and fault tolerance through multiple SIP Application Mo dules deployed in an N+M active-standby configuration.

Note: The supported active/standby configurations include:

• a 1+1 configuration (one active plus one standby server), which is the most basic reliable configuration

• an N+M configuration of up to four servers (the sum of N plus M should not exceed 4)

— a 2+1 (2 active and one standby) — 2+2 configuration — 3+1 configuration

To accomplish this, all the servers in a reliability group are configured with the same set of NSDs. This gives the standby server the information it needs in case an active server fails. Each server in the group transmits messages indicating its current state. Other servers respond with their curren t states, including th e NSD activated on the m.

An initializing server configures itself with one of any inactive NSDs. If all NSDs are active, the initializing server becomes the standby. This prevents confl i cts where more than on e server is activating simultaneously.

Before activating, the server determines whether it is isolated from critical network resources defined through provisioning. If any of the resources cannot be reached, the server cannot activa te and raises an alarm. The alarm clears when the resources become available.

When there are two or more active servers, the g roup is called a cluster . You can configure b oth the N+M strate gy and the cluster at the Transport Management tab in step 22 in the Configuration chapter.

When one of the active SIP Application Modules fails, the passive Module takes over the IP address. The passive Module has now become active and assumes the responsibilities of the failed Module. When this occurs, any sessions already in the active state remain up. This means that ca lls that ha ve already bee n establ ished contin ue and the parties maintain voice path. Any future requests during that session, however , fai l (for examp le, Hold, Retr ieve, and W eb Pu shes) since th e session information is no longer available. Any sessions that were not in the active state bef ore the failove r ar e lost. The ori ginating clien ts o f

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these sessions either receive no indication or continue to hear an alerting tone for an indefinite period of time.

Manual failover

There are two recommended procedures for manually initiating the fail-over of an active instance to a Standby node: the initiation of discrete LOCK and UNLOCK actions, or the initiation of a restart.

Lock/Unlock If you want to force a fail-over in order to perform maintenance on the "failed" se rver, then request a LOCK from th e Management Console. The LOCK forc es the component into a disabled operational state, where it remains until you request an UNLOCK from the Management Console. You can perform any maintenance on the "failed" server while it is LOCKed. Once maintenance is complete, the server can be UNLOCKed from the Management Console, which causes an automatic restart and brings the server back into service.

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Restart If you want to simply force an immediate manual fail-over, then you can request a Restart from the Management Console.

WARNING

The N+M reliability strategy provides a highly available service environment. The fail-over mechanisms enable an instance of the S IP Application Module to survive failure condition(s) by migrating to a standby server where it can resume the processing of new sessions.

In such a highly available service environment the failed instance loses all knowledge of sessions started before the fail-over event. Therefore, the stability of these pre-existing sessions cannot b e guaranteed. For examp l e:

Sessions involving SIP clients will survive until the clients encounter a "no response" or "unknown call" response to a request on their active se ssion. At that point the clients will release the session and its associated media resources.

For more information, see the Configuration chapter in this document.

OAM&P strategy

The Management Module manages the OAM&P functions for the SIP Application Module. For additional information, refer to the MCP

Management Modu le Basics and the MCP System Management Console Basics documents.

Sessions involving the MCP SIP PRI Gateway will survive until there is no response to the SIP PRI Gateway-generated SIP "ping" to the SIP Application Module(s) handling the active sessions on the gateway. If there is no response to the SIP "ping" then the gateway will tear down the associated call and recovers its resources.

Also, sessions involving the MCP RTP Media Portal will not survive a manual fail-over because intentionally LOCKing the SIP Application Module initiates the automatic recovery of all resources (including RTP Media Portal resources) associated with in-progress sessions.

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Upgrades

For information on upgrading from one full relea se to another, refer to the Installation and Commissioning document you receive wi th the upgrade.

Updating the SIP Application Module software

Administrators can update the software version of the SIP Application Module using the System Management Console. The update can be either an up- or down-version of the software.

Updating the software affects the operation of the component’s hosted services during the procedure. This process automatically fills the service property fields of the updated component with the configured values from the previous version.

The update introduces new functionality across many components without affecting network stability. If a server update fails, you have a choice to roll back or not. For more information on the update procedure, refer to the MCP System Manageme nt Co nso le Basi cs document.

at the System Management Console

1 A load can be either up-versioned or down-versioned. In either

case, updating a load from one version to another results in stopping and deleting the previously added version, adding the new version and auto-launching the new version. Therefore, there is no need to manually LOCK and UNLOCK the service. The steps involved in an update are described below.

From the System Management Console, under the

Components folder , sele ct the name configured at deployment, AppSvr in the example shown in Figure 1.

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Figure 1 Updating the Application Module from the menu tree

You can also launch the update from the pull-down Configuration menu, as shown.

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Upgrades 31

Figure 2 Updating the SIP Application Module from the pull-down menu

2 Select the Update command. The foll owing window appea rs. Figure 3 The update window, retrieving the load list

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3 Because only versions not currently deployed appear in the

loadlist following the Add->Component step in the update procedure, multiple versions may not appear for the update operation. You can only do an update from one version to another. Therefore, the window only shows loads that have the same name as the lo ad being updated (see Figur e 4, “Load list for updating”).

Figure 4 Load list for updating

4 Select the version you want to update. Click on th e Apply

button.

5 The configuration window appears, showi ng the ta bs. Modify all

the configuration values you need to modify. Then click on the Apply button. The window that appears shows the progress of the update (see Figure 5, “Progress of update”). Each configured manage d object (MO) app ears as being successf ully added onto the managed element (ME).

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Figure 5 Progress of update

6 Once the update has complete d, th e fo llo w ing w ind ow ap pe ar s.

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Figure 6 Successful update dialog box

OAM&P strategy

The Management Module manages the OAM&P functions for the SIP Application Module. For additional information, refer to the MCP

Management Module Basics and the MCP System Management Console Basics documents.

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Fault management

The Management Module manages the faults for the SIP Application Module. For additi onal information on the Mana gement Module, refer to the MCP Manageme nt Mo du le Basics and the MCP System Management Console Basics documents.

How this chapter is organized

This chapter is organized as follow s:

• Alarm clearing procedures

• Recovery procedures

Alarm clearing procedures

Procedure 1 Clearing the SLE701 (SLEE Health Monitor) alarm

at the alarm browser

1 The SIP Application Module raises this warning alarm under one

of two conditions:

• The number of ap plication contexts (AC) available for use are inadequate for the level of traffic (in which case the administrator needs to back off the traf fic or call the next level of support).

The SIP Application Module raises this alarm when AC pool use reaches or exceeds 80%. The alarm clears when use drops below 80%.

• There is an error condition that is caus ing ACs to be consumed at a higher than normal rate. This could be due to a myriad of things; for example, the system might be consuming RetrieveSubscriber ACs at a high rate because the database is overloaded.

Severity is MAJOR. The SLEE is a service processing environment. An AC is a unit of work within that processing

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Procedure 2 Clearing the SMDI101 alarm

at the alarm browser

1 This alarm is raised when the Simplified Message Desk

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framework. For example, when you register your phone, a number of AC instance s are invoked to process the regi stration.

Interface (SMDI) telnet session between the SIP Application Module and a terminal server is lost. The SIP Application Module uses the SMDI pr otocol to communicate information between itself and a voicemail server. If the connection goes down,

• Message W aiting Indica tion notification to subscribers stops.

• Calls routed to the voicemail server are not sent to the appropriate mailbox.

However, depending upon the voicemail server’s capabilities, the calls may be answered by a default mailbox and the originator can enter the desired mailbox number in which to leave a message. In the same way , users may be able to retrieve their voicemails (for example, they get routed to the default mailbox and are prompted to enter their mailbox). Again, this functionality depends upon the voicemail server being used.

The SIP Application Module repeatedly tries to re-establish the telnet session to the terminal server. If the alarm does not go away in a few minutes, then the terminal server needs to be checked and possibly re-booted. Al so, admi nis tra tor s shou ld check the voicemail server to make sure it is running correctly. If problems persist, contact your next level of support.

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Configuration

How this chapter is organized

This chapter describes the pro cedures for configur ation required a t the service provider premises. This chapter is organized as follows:

• “Overview” on page 38

• “Adding a component” on page 40

• “Configuring the SIP Application Mo dule tabs” on page 44 — Applicatio n Ser v er — Long Call Service — Presence — Presence Location Service — Authentication —Media Portal — Locate User Svc

— Database Base — Data Synchronization — Prov Sync Service — Overload Controls — Forward Locati on Service — Local Accounting Manager — In Memory Database — Location Service — Registration — Server Properties — Server Subscription — SIP TCF Base

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Overview

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— Svc Pkg Enforcement Service — SipFwdAdapter — Transport Management

• “Additional SIP TCF Base tab configuration information” on page 90

• “OAM&P strategy” on page 92

CAUTION

Before making any changes to the base configuratio n, consult your next level of support . .

Nortel Networks performs the initial installation and commissioning. Once the install ation and commissioning are comp leted, you can begin to make your system fully operational. The following list identifies some general tasks:

• provision and complete translations to enable voice and trunk services

• configure any additional services, applications, and features that Nortel Networks i s not contracted to perform

• complete the installation of clients or add client software for all management interfaces

The SIP Application Module is configured using the System Management Console. For more information, refer to the MCP

Management Modu le Basics and the MCP System Management Console Basics documents. This chap ter describes the configurable

parameters affecting operation of the SIP Application Module and the procedures for config uration requi red at the service provi der premises.

Deployment from the System Management Console results in the installation of all SIP Application Module-specific software and configuration data on the host machine, and starts the software processes. Undeployment stops the software processes and remove s all related software and configuration data. When the deployment is complete, the SIP Application Module should be unlocked, enabled, and available to provide servic e.

Before a SIP Application Mod ule can be de pl oye d, the server must have been configured at the System Management Console. This server

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represents the physical hardware on which the SIP Application Module resides. Once the servers are configured, the SIP Applicatio n Module can then be deployed.

The SIP Application Module depends on various components that require configuration during the deployment process. In general, most of the SIP Application Module’s configuration items can be left with their default values; however, administrators should familiari ze themselves with the available options.

Administrators can also find h elp te xt with de scriptions an d accept able ranges by holding the cursor over the field name as shown in Figure 1, “Displaying help text.”

Note: In all tabs, the fields with asterisks (*) require an entry. The grayed-out fields are for information only and cannot be changed. Change all occurrences of the IP address “0.0.0.0” to the proper IP address for your situation.

Figure 1 Displaying help text

Configuration 39

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Adding a component

This procedure assumes that the serve r on whi ch the SIP Application Module will be depl oyed has already been configured. For example, Figure 3, “Adding a component,” shows the SIP Application Module being deployed onto the previously configured server. For the procedure for adding a server, refer to the MCP System Management

Console Basics.

Procedure 1 Adding a component

at the System Management Console

1 T o add the SIP Application Module, navigate to and right click on

the Components item in the Management Console tree structure.

Figure 2 Navigating to the Components file

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2 Select Add->Component as shown.

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Figure 3 Adding a component

3 Select the SIP Applica ti on Mo du l e so ftw are l oa d yo u w a nt fr om

the load list that appe ars (see Fig ure 4, “Load list”) and click on the Apply button. There may or may not be multiple software loads to choose from.

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Figure 4 Load list

4 The Configuration window (shown in Fi gure 5, “Configurati on

window (top half)”) appears. Once the configuration window appears, enter a label with a maximum of six characters in the Service Component Name field at the bottom. This name must be unique among the components. The following figure shows

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an example with the name SIPApp entered in the Service Component Name field.

Figure 5 Conf iguration window (top half)

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ATTENTION

DO NOT click on Apply until you have FINISHED filling in the fields that you need.

Note that there are a number of different tabs in the SIP Application Module configuration window representing the configurable serv ic es th at th e SI P Ap pl i cati o n M odu le r eq ui res. The following sections describe each tab in detail and provide guidance on how to configure the SIP Ap plication Module . Many of the fields a re alread y fille d with def ault valu es. Admin istrators can leave most of the filled-in fields with their default values. Only a few fields need customization.

Note: The parameters with asterisks (*) are mandatory. The grayed-out fields are for information only and cannot be changed.

Make any required modifications to any of the tabs. When you have COMPLETED all the tabs, click on the Apply button.

6 After you click the Apply button, the Management Module

begins the deployment and installation of the SIP Application

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Module. The Adding Services Progress dialog box appears as shown in Figure 6, “Adding Services Progress dialog box.”

Figure 6 Adding Services Progress dialog box

Configuration 43

If the deployment is successful, an “Add successful” box appears, as shown in Figure 7, “The Add successful dialog box.”

Figure 7 The Add successful dialog box

If the deployment is not successful, re-examine the configuration tabs and verify that al l 0.0.0.0 IP addresse s have been replaced with the correct IP addr ess. V erify oth er non-defau lt param eters for accuracy. The SIP Application Module and all of its services must be unlocked and enabled. There must be no alarms.

After deployment and installation, the Management Module configures services according to values entered in the configuration tabs during step 4.

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Configuring the SIP Application Module tabs

The following sections describe the configuration tabs in detail. The tables following the figures describe the fields shown in the figures.

Note: These tabs do NOT have to be completed in this particular order. The following order is only for example.

Procedure 2 Completing the tab fields

at the System Management Console

1 Click on the Application Server tab. The Application Server tab

allows the SIP Application Module to set high-level data, such as the title of the server instance, the managed domains, and the private IP address of the server.

Figure 8 Completing the Application Server tab fields

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Table 3 Application Server tab field descriptions (Sheet 1 of 2)

Field Value Description

Show GUI Type=checkbox

This is a read-only field.

Default=unchecked

Title Type=string

Range=0-64 characters

This field contains the title of this Server instance.

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Table 3 Application Server tab field descriptions (Sheet 2 of 2)

Field Value Description

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Opaque Headers

Private IP Address

Type=string Range=0-2048 characters Default=to+from+call-id+via+ cseq+content-type+contentlength+contact+record-route+ route+proxy- re qu ir e+rseq

Type=valid IP address Range=0-4096 numbers or blank Default=0.0.0.0

This field contains a “+” delimited list of headers that should not be passed through the server.

This field contains the private IP address of the server.

Note: Do not leave this field b lank or the software will not deploy.

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2 Click on the Long Call Service tab. The Long Call Service tab

allows the service provider to set the length of time between endpoint audit s. The Lo ng Call Servi ce detect s abandoned calls and releases the resources used by such calls.

Figure 9 Completing the Long Call Service tab fields

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Table 4 Lo ng Call Service tab field descriptions

Field Value Description

Duration Type: Integer

Range: 0–Max_Integer Default: 60 minutes

This field shows the length of ti me in minutes between endpoint audits and is used to detect abandoned calls. A value of zero deactivates it. The recommended value is 10 (minutes). If the SIP Application Module detects an abandoned call at the endpoint audit, it drops the resources for that leg.

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3 Click on the Presence tab. This tab allows the service provider

to configure con text and exp iration inf ormation for the Presence service.

Figure 10 Completing the Presence tab fields

Configuration 47

Table 5 Presence tab field descriptions

Field Value Description

maximumNumber OfIdleContexts

Type=integer Range=1-10000 Default=50

This field indicates the maximum number of idle contexts at any time. This should not exceed the maximum number of contexts.

initialNumberOfContexts Type=integer

Range=1-10000 Default=1

This field indicates the initial number of contexts to create. This should not exceed the maximum number of contexts.

maximumNumberOf Contexts

Type=integer Range=1-15000

This field indicates the maximum number of contexts to create.

Default=15000

maximum Expires Type=integer

Range=60-86400 Default=3700

This read-only field contains the maximum allowable expiration value for a presence subscription request, in seconds.

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4 Click on the Presence Location Service tab. This tab allows the

service provider to configure the use of off-board Location Servers for routing.

Figure 11 Completing the Presence Location Service tab fields

Table 6 Presence Location Service tab field descriptions (Sheet 1 of 2)

Field Value Description

Maximum Number of Idle Contexts

Type=integer Range=1-100 numbers Default=10

This read-only field contains the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

Initial Number of Contexts

Type=integer Range=1-100 numbers Default=1

This read-only field contains the initial number of contexts to create. It should not exceed the maximum number of contexts.

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Table 6 Presence Location Service tab field descriptions (Sheet 2 of 2)

Field Value Description

Maximum Number Of Contexts

Type=integer Range=1-10000 numbers Default=5000

Use DNS Type=checkbox

Default=unchecked

Use Location Server

DNS Provider URL

Type=checkbox Default=unchecked

Type=string Range=1-1024 numbers Default=dns://0.0.0.0

DNS Default Transport

Type=string Range=udp, tcp Default=udp

Location Server URL

Location Server Timeout

Type=string Default=sip://0.0.0.0:5060:udp

Type=string Default=3600

This read-only field contains the maximum number of contexts to create.

Turns DNS server f unctionality on and off.

Turns Location Server functionality on and off.

This field indicates th e address of the DNS Server format >dns://0.0.0.0

Transport type used to communicate with the DNS server.

This is the address of the Location Server.

This field is not used.

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5 Click on the Authentication tab. The Authenticat ion tab enables

or disables authentication for requests and sets additional authentication information.

Figure 12 Completing the Authentication tab fields

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Table 7 A uthentication tab field descriptions (Sheet 1 of 2)

Field Value Description

Methods to Authorize

Realm Type=string

Type=string Default=register

Range=0-256 charac ter s

This field indicates which SIP methods to authenticate.

This field indicates the string that is displayed to the user to indicate what realm they need to supply a password for.

Private Key Type=string

Range=0-256 charac ter s

An extra key used to uniquely generate authenticati on challenges.

Default=MCP

Nonce Interval

Type=integer Range=10000- 600000 milliseconds Default=600

The software uses this field to determine how long to wait (in milliseconds) for a response to a challenge with a specific nonce value before generating a new nonce value.

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Table 7 A uthentication tab field descriptions (Sheet 2 of 2)

Field Value Description

Configuration 51

Authorized SIP Nodes

Type=IP address Range=0-2000 numbers

Nodal Auth Type=checkbox

Default=unchecked

This field contains a + -delimited list IP addresses. Use the SIP PRI Gateway and SIP Audio Server addresses.

When this field is checked, the SIP Application Mod ule redirects reques ts. When unchecked, this field authenti cate s requests and o nly the SIP PRI Gateway and SIP Audio Server listed in the previous field can send INVITE messages to the SIP Application Mo dule without au thentication. Nortel Networks recommends that you do not change this field.

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6 Click on the Media Portal tab. This tab allows the service

provider to set port and firewall in formation pertaining to the Media Portal.

Figure 13 Completing the Media Portal tab fields

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Table 8 Media Portal tab field descriptions

Field Value Description

Initial Capacity

Type=integer Range=113-16384

This field is not used.

Default=113

Fire Wall Type=checkbox

This field is not used.

Default=unchecked

MGCP Port Type=integer

Range=1025-65535 Default=3903

This field indicates the UDP Communica tions port number where the Me dia Portal sends and receives MGCP+ messages.

7 Click on the Database Base tab.

General proper tie s fo r the SIP Application Modu le’s connection to the database ar e define d in the Dat aba se Base t ab. Se e the MCP Database Module Basics document for more information and field descriptions. Modifications to the Database Base require that th e Database Base be locked. A lock of the Database base releases all SIP Application Mo dule resources associated with the Database Base. When released, these

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resources are removed from the SIP Application Module’s local cache. When the Dat abase Base is unlocked, all SIP Application Module resources must be reall o cate d cau si ng a re-r ead of the resources from the data base. This tab also cont ain s connecti on information for the database.

Figure 14 Completing the Database Base tab fields

Configuration 53

Note: See the MCP Database Module Basics document for field descriptions.

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8 Click on the Locate User Svc tab. This tab allows a dministrators

to configure the use of off-board Location Servers for routing.

Figure 15 Completing the Locate User Svc tab fields

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Table 9 Lo cate User Svc tab field descriptions (Sheet 1 of 2)

Field Value Description

DNS SRV Default Transport

DNS SRV URL Type=string

Type=string Range=UDP, TCP Default=UDP

Range=1-64 numbers

This field indica tes the transport type used to communicate with the DNS SRV server.

This field indicates the address of the DNS SRV server.

Default=dns://0.0.0.0

Use DNS SRV Type=checkbox

Default=unchecked

If box is checked, then enter a URL in the DNS SRV URL field. Prefix that URL with dns://.

Location Server Transport

Type=string Default=UDP

This read-only field contains the transport type used to communica te with the Location Server.

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Table 9 Lo cate User Svc tab field descriptions (Sheet 2 of 2)

Field Value Description

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Location Server URL

Use Location Server

Range=1-64 numbers Default=sip://0.0.0.0:5065

Type=checkbox Default=unchecked

This field indicates the address of the Location Server.

This field is not used.

9 Click on the Data Synchronization tab. This tab allows the

service provider to set the context and expiration information relating to the synchroni zation of in-memory and pe rsistent data.

Figure 16 Completing the Data Synchronization tab fields

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Table 1 0 Data Synchronization tab field descriptions

Field Value Description

maximumNumberofIdleContexts Type=integer

Range=1-10000 numbers Default=100

initialNumberofContexts Type=integer

Range=1-10000 numbers Default=10

maximumNumberOfContexts Type=integer

Range=1-10000 numbers Default=1000

maximumExpires Type=integer

Range=60- 86400 seconds Default=3700

10 Click on the Prov Sync Service tab. The Prov Sync Service

parameter forwards provisioning modifications on user and device records to the SIP Applicati on Modul e wh en eve r modifications occur or when additions or deletions are made.

This is the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

This is the maximum number of con texts to create.

This is the maximum allowable expiration value for a DataSync subscription request, in seconds.

The Prov Sync Service tab allows the SIP Application Module to keep its configuration data updated with any changes that are made through th e Provisioning Client web page. This tab also allows service providers to set how often the SIP Application Module queries the database for provisioning changes.

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Modifications to this tab require that the Prov Sync Service be locked.

Figure 17 Completing the Prov Sync Service tab fields

Configuration 57

Table 11 Prov Sync Service tab field descriptions

Field Value Description

Sync Time Period

Type=integer Range=5-60000 seconds Default=10

This field indicat es how often (in seconds) the SIP Application Module queries the database for provi si oning changes.

Task Group Size

Type=integer Default=5

This is a read-only field.

11 Click on the Overload Controls tab. This tab allows the service

provider to set threshold alarm information and system resource collection intervals.

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Figure 18 Completing the Overload Controls tab fields

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Table 12 Overload Controls tab field descriptions (Sheet 1 of 2)

Field Value Description

Mem Polling Interval (sec)

Type=integer Range=a positive integer

This field indicates the number of seconds to wait in between checks on memory usage.

Default=3

Minor Alarm Threshold (%)

Type=integer Range=0-100 num bers Default=80

This field indicates the threshold at which Minor overload is encountered f or both CPU and memory.

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Table 12 Overload Controls tab field descriptions (Sheet 2 of 2)

Field Value Description

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Major Alarm Threshold (%)

Critical Alarm Threshold (%)

Call queue high threshold

Call queue low threshold

Other queue high threshold

Type=integer Range=0-100 num bers Default=85

Type=integer Range=0-100 num bers Default=90

Type=integer Range=1-500 num bers Default=25

Type=integer Range=0-500 num bers Default=5

Type=integer Range=1-500 num bers Default=15

This field indicates the threshold at which a Major overload is encountered f or both CPU and memory.

This field indicates the threshold at which a Critical overload is encountered f or both CPU and memory.

This field contains the number of elements in the queue that, if exceeded, causes the system to disallow new calls.

This field contains the number of elements in the queue that causes the system to allow new calls.

This field contains the number of elements in the queue that, if exceeded, causes the system to disallow other session types, such as registrations, instant messages, or subscriptions.

Other queue low threshold

Type=integer Range=0-500 num bers Default=2

This field contains the number of elements in the queue that causes the system to allow other session types.

Database queue high threshold

Type=integer Range=1-500 num bers Default=5

This field contains the number of elements in the queue that, if exceeded, causes a cluster overload.

Database queue lo w threshold

Protocols Monitored Type=string

Type=integer Range=0-500 num bers Default=0

Range=0-1024 characters

This field contains the number of elements in the queue that causes the cluster overload to clear.

Plus sign (+)-delimited list of protocols whose IO queues are monitored for excessive delays.

Default=sip

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12 Click on the Local Accounting Manager tab. For more

information on the Local Accounting Manager t ab fields, see th e MCP Accounting Module Basics document. This tab contains information pertaining to the Accounting Manager and billing records, including IP addresses, port s, file rotation size and time, and recording unit queue size.

Figure 19 Completing the Local Accounting Manager tab fields

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13 Click on the In Memory Database tab. This tab allows the service

provider to set information relati ng to local domains, event subscriptions, and the nonce used for authentication.

Figure 20 Completing the In Memory Database tab fields

Configuration 61

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Table 13 In Memory Database tab field descriptions (Sheet 1 of 2)

Field Value Description

IMDomainInfo optimal table limit

Type=integer Range=16-Max Int nu mb er s Default=64

IMDomainInfo table sync interval

Type=integer Range=300- 3600 numbers Default=600

IMUserInfo table limit Type=integer

Range=32768-MaxInt numbers Default=131073

IMUserInfo table sync interval

Type=integer Range=1800- 86400 numbers Default=1800

IMDomainSubBan table limit

Type=integer Range=16-Max Int nu mb er s Default=64

IMDomainSubBan table sync interval

Type=integer Range=300-86400 numbers Default=900

Set this to the number of domains and subdomains you expect to have.

Set this to however often you want the system to scan for changes to domain provis ioning.

Set this to the number of users you expect to host.

Set this to however often you want the system to scan for changes to the user’s provisioned presence information.

Set this to the number of domains and subdomains you expect to have.

Set this to however often (in seconds) the SIP Application Module checks the database for changes in the domain ban information.

IMWatchList optimal table size

Type=integer Range=32768-MaxInt numbers

Set this to the total number of user ban list entries yo u expect.

Default=32768

IMWatchList table sync interval

Type=integer Range=300-1800 numbers Default=900

Set this to however often (in seconds) the SIP Application Module checks the database for changes in the user ban information.

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Table 13 In Memory Database tab field descriptions (Sheet 2 of 2)

Field Value Description

Configuration 63

IMWatchers optimal table size

Type=integer Range=65536-MaxInt numbers Default=131073

IMWatchers table cleanup interval

Type=integer Range=300 to 86400 numbers Default=1800

IMAuthNonce table si ze Type=integer

Range=1-10 numbers Default=5

IMAuthNonce generation interval

Subscriber cache interval

Type=integer Default=60

Type=integer Range=60-3600 numbers Default=60

Set this to the number of subscriptions expected at any interval in time.

Set this to however often (in seconds) you want the system to clear out any expired event subscriptions.

Set how many nonces you want the system to keep for authentication.

Sets how often (in seconds) the system creates a new nonce. This is a read-only fi eld.

Set this to however long you want the system to cache, in memory, subscriber information from the database, in seconds.

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14 Click on the Location Service tab. This tab allows the service

provider to configure the use of off-board Location Servers for routing.

Figure 21 Completing the Location Service tab fields

SipApp

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Table 14 Location Service tab field descriptions (Sheet 1 of 2)

Field Value Description

maximumNumberOf IdleContexts

Type=integer Range=1-512 numbers Default=10

This is the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

initialNumberOf Contexts

Type=integer Range=1-512 numbers Default=10

This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

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Table 14 Location Service tab field descriptions (Sheet 2 of 2)

Field Value Description

Configuration 65

maximumNumberOf Contexts

Type=integer Range=1-512 numbers Default=100

UseDNS Type=checkbox

Default=unchecked

UseLocationServer Type=checkbox

Default=unchecked

DNSProviderURL Type=string

Range=0-1024 numbers Default=dns://0.0.0.0

DefaultTransport Type=string

Range=udp or tcp Default=udp

LocationServerURL Type=string

Range=0-1024 numbers Default=sip://0.0.0.0:5060:udp

LocationServerTimeout Type=integer

Default=3600

This is the maximum number of contexts to create.

Turns DNS server functionality on and off.

Turns Location server functionality on and off.

This is the address of t he DNS server.

Transport type used to communicate with th e DNS server.

This is the address of t he Location Server.

This is a read-only field.

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15 Click on the Forward Location Service tab. This tab allows the

service provider to configure the use of off-board Location Servers for routing.

Figure 22 Completing the Forward Location Service tab fields

Table 15 Forward Locat ion Service tab field descriptions (Sheet 1 of 2)

Field Value Description

Maximum Number of Idle Contexts

Type=integer Range=1-100 numbers Default=10

This is a read-only field. This is the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

Initial Number of Contexts

Type=integer Range=1-100 numbers Default=10

This is a read-only field. This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

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Table 15 Forward Locat ion Service tab field descriptions (Sheet 2 of 2)

Field Value Description

Maximum Number Of Contexts

Type=integer Range=1-100 numbers Default=150

Use DNS Type=checkbox

Default=unchecked

Use Location Server Type=checkbox

Default=unchecked

DNS Provider URL Type=string

Range=1-1024 numbers Default=dns://0.0.0.0

DNS Default Transport Type=string

Range=udp, tcp Default=udp

Location Server URL Type=string

Default=sip://0.0.0.0:50 60:udp

This is a read-only field. This is the maximum number of contexts to create.

Turns DNS server functionality on and off. Check th is field if you are using the DNS Svr se rv ice to resolve foreign domains.

Turns Location Server functionality on and off.

This field indicates the address of the DNS Server format >dns://0.0.0.0

Transport type used to communicate with the DNS server.

This is the address of the Location Server.

Location Server Timeout

Type=string Default=3600

This field is only used with an offboard Location Server. This timeout value tells the Application Server how long to wait for an answer from the Location Server. If it didn't receive one in this time, the request would be failed .

16 Click on the Registration tab. This tab contains registration

context information as well as the valid maximum expiration value for registrations.

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Figure 23 Completing the Registration tab fields

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Table 16 Registration tab field descriptions

Field Value Description

maximumNumberOf IdleContexts

Type=integer Range=1-10000 numbers Default=50

This is the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

initialNumberOf Contexts

Type=integer Range=1-10000 numbers Default=10

This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

maximumNumberOf Contexts

Type=integer Range=1-10000 numbers

This is the maximum number of contexts to create.

Default=5000

Valid Maximum Expires

Type=integer Range=60-86400 numbers Default=86400

This is the maximum allowable expiration value for a registration request, in seconds.

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17 Click on the Server Properties tab. This tab allows the service

provider to set the system properties for the server.

Figure 24 Completing the Server Properties tab fields

Configuration 69

Table 17 Server Properties tab field descriptions (Sheet 1 of 2)

Field Value Description

Label: Type=string

Range=1-80 characters

This is a label that identifies the public address for the server.

Default=Public_Service_Address

Value Type=string in the form of a valid

This field contains an IP address.

IP address (x.x.x.x)

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Table 17 Server Properties tab field descriptions (Sheet 2 of 2)

Field Value Description

Label: Type=string

Range=N/A Default=Private_Service_Address

Value: Type=string in the form of an IP

address (x.x.x.x)

Label: Type=string

Range=1-80 characters Default=blank

Value: Type=string

Range=1-80 characters Default=blank

Label: Type=string

Range=1-25 characters Default=blank

This is a label that ident ifies the privat e address for the Server. Only add an entry for a standalone configuration.

This field contains t he pr ivate mach ine logical IP address of the SIP Application Module assigned to the label just above this field.

This field contains a unique label to reference the value to the field directly below assigned below. Enter server.gateways. Only add an entry for redundant configurations.

This field cont ains the nodes that the software needs to check upon boot up of this SIP Application Module. This value is assigned to the label just above this field.

This field contains a unique label to reference the value. Enter server.blade.host.label for interworking with the RTP Media Portal. See Table 22, “Transport Management tab fi eld descriptions,” on page 79, which contains the actual value of private_static_address. In other words, this is wher e a numeric IP address is assigned to the label

private_static_address. The label server.blade.host.label has a value

which is another label (private_static_address) whose value in turn is define d in the Transport Management tab.

Value: Type=string

Range=1-25 characters

This field should contain the string private_static_address.

Default=blank

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18 Click on the Server Subscription tab. This tab contains a list of

the provisioning servers that the SIP Application Module can communicate with, as well as the context limits.

Figure 25 Completing the Server Subscription tab fields

0.0.0.0:5095:8000:http

Configuration 71

Table 18 Server Subscrip tion tab field descriptions (Sheet 1 of 2)

Field Value Description

maximumNumberOf IdleContexts

Type=integer Range=1-10000 numbers Default=1

This is the maximum number of idle con texts at any time. It should not exceed the maximum number of contexts.

initialNumberOf Contexts

Type=integer Range=1-10000 numbers Default=1

This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

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Table 18 Server Subscrip tion tab field descriptions (Sheet 2 of 2)

Field Value Description

maximumNumberOf Contexts

Type=integer Range=1-10000 numbers Default=10

provServerURL Type=string

Range=0-4096 characters Default=0.0.0.0:5095:8000:http

19 Click on the SIP TCF Base tab. The SIP TCF Base provides

support for the SIP protocol. The SIP Application Module is one of several components that use the SIP TCF Base. See “Additional SIP TCF Base tab configuration information” on page 90 for more information. The SIP TCF Base cont ains many parameters perta ining to the SIP Applicat ion Module’ s trans port configuration. It inclu des information regard in g the tran spor t IP addresses/ports, timers, number of redirects, and retransmission, among other items. Modifications to the SIP TCF Base tab require that the SIP TCF Base be locked.

This is the maximum number of contexts to create. The range is 1 to

10000. The comma-delimited list of

Provisioning Modules this server is to communicate with. Format is [IP ADDR]:[SIP PORT]:[HTTP PORT]:[PROTOCOL].

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Figure 26 Completing the SIP TCF Base tab fields

Configuration 73

Table 19 SIP TCF Base tab field descriptions (Sheet 1 of 2)

Field Value Description

Transport Config Type=string

Default=UDP=0.0.0.0:5 060:optional:name=sipL scConduit;TCP=0.0.0.0: 5060:optional:name=sip LscConduit

Specifies the transport, IP addresses, and ports. Includes both the public and private interfaces. If your system consists of only public IPs, do NOT duplicate the string. This field indicates the public IPs for the UDP and TCP portions only. Transports can appear more than once. Use this field only for a standalone system. Enter the machine logical IP address of the SIP Application Module. For redundant configurations, leave this field blank.

Retransmission Off

Type=checkbox Default=unchecked

This is a read-only field that controls SIP retransmissions.

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Table 19 SIP TCF Base tab field descriptions (Sheet 2 of 2)

Field Value Description

Invite Timer Type=integer

Range=1200003600000 number s Default=128000

Application Type Typ e= str ing

Range=callstate_server, stateful_server, stateless_server, user_agent Default=callstate_server

Time Transaction Type=checkbox

Default=checked

Add Defaults Type=checkbox

Default=checked Recommendation is to check the box.

Check Mandatory Headers

Type=checkbox Default=unchecked Recommendation is NOT to check the box.

This controls the maximum time in milliseconds to wait for an INVITE to receive a Final Response after receiving a provisional Response.

This is the type of SIP Server on the node.

This field specifies whether the SIP transactions should be timed. This field is read only.

Specifies whether to fill in missing mandatory headers with default values in the SDP message bodi es.

Controls wh ether the mandato ry SDP headers are checked f or prese nce in the SDP messages.

Maximum Number of Redirections

Initial Maximum Hop Value

Type=integer Range=3-10 numbers Default=5

Type=integer Range=5-50 numbers

Maximum number of redirections allowed before a re qu est i s dropped.

Maximum number of hops allowed before a request is dropped.

Default=20

20 Click on the Svc Pkg Enforcement Service tab. This tab allows

the service provider to toggle on and off the enforcement of audio conferencing and voicemail settings based on the user’s service packages.

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Figure 27 Completing the Svc Pkg Enforcement Service tab fields

Table 20 Svc Pkg Enforcement Service tab field descriptions

Field Value Description

Configuration 75

Enforce Type=checkbox

Default=unchecked

Turns on or off the server-based enforcement of Audio Conferencing and Voice Mail services based on users’ Service Package settings. If you are only using clients, it is not necessary to check the box since the clients will perform the enforcement. If the re are thir d-party cli ents, you may want to check the box so that the SIP Application Server will enforce the service package.

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21 Click on the SipFwdAdapter tab. This tab allows the service

provider to set the valid events that the SIP Application Module will process and determines whe ther or not the SIP Application Server will or will not forward messages to a forei gn serve r.

Figure 28 Completing the SipFwdAdapter tab fields

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Table 21 SipFwdAdapter tab field descriptions (Sheet 1 of 2)

Field Value Description

maximumNumberOf IdleContexts

Type=integer Range=1-10000 numbers Default=50

This is the maximum number of idle contexts at any time. It should not exceed the maximum number of contexts.

initialNumberOf Contexts

Type=integer Range=1-10000 numbers Default=10

This is the initial number of contexts to create. It should not exceed the maximum number of contexts.

maximumNumberOf Contexts

Type=integer Range=1-10000 numbers

This is the maximum number of contexts to create.

Default=5000

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Table 21 SipFwdAdapter tab field descriptions (Sheet 2 of 2)

Field Value Description

Configuration 77

forwardSubscribe Type=checkbox

Default=unchecked

forwardRegister Type=checkbox

Default=unchecked

allowedEvents Type=string

Range=0-512 ch aracters Default= presence, address-book, message-summary, service-package

22 Click on the Transport Management tab. This tab has a number

of subfields. The next series of screens and tables give information on wh at data to enter an d where to enter it . For more information on the fun ction of this t ab, see the secti on followin g, “Transport Management: Active-Hot Standby Server Heartbeat Mechanism” on page 87. This tab contains a large number of server and network service parameters that set protocols, transports, ports, and heartbeat information.

If box is checked, the system allows subscribe messages to be forwarded.

If box is checked, the system allows register messages to be forwarded.

This field indicates the valid event packages for the SIP Application Mo dule to process.

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Figure 29 Completing the Transport Management tab fields

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Configuration 79

Table 22 Transport Management tab field descriptions (Sheet 1 of 3)

Field Value Description

Service Name

Type=string Range=1-20 characters Default=BBUA

Server ID Type=integer

Range=1-4 number s Default=1

StandAlone Server

Type=checkbox Default=checked

Label Type=string

Range=1-15 characters Default=Public_Static_Address

Value Type=string in form of an IP

address

Label Type=string

Range=1-15 characters Default=Private_Static_Address

Value Type=string in the form of an IP

address

This field indicates the name of the service this Reliability Manager is supporting.

This field indicates the ID number for this server.

Is the server standalone or part of a reliable group? Check the box if the server is standalone.

This field contains a unique label that references the value.

This field contains the value to assign this label (IP address).

This is a unique label that references the value.

This field contains the value to assign this label (IP address).

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Table 22 Transport Management tab field descriptions (Sheet 2 of 3)

Field Value Description

Label Type=string

Range=1-15 alphanumeric characters Default=Service_Node_Name

This is the name by which the CS 2000 knows an MCP server. This name has to be assigned to the service instance in the N+M configuration so it cannot be the node name of the platform. In addition it must contain no special characters like "_" or "-". It is defined in one of two places when the node is deployed from the management server. If the server is running as part of an N+M cluster then each service instance is defined as a service parameter in each Network Service Description (NSD) in the Transport Management tab. Each NSD has to define a unique service name. This is done by adding a service name of "Service_Node_Name" in the label field and the desired node na me in the Value part.

This information must then be datafilled in the CS 2000 as the name and IP from the NSD.

If the system is not running the N+M then the service name needs to be added to the "Server Properties" tab. This information als o needs to be entered on the CS 2000.

Value Type=string in the form of an IP

address

This field contains the value to assign this label (IP address).

Range=<host name of node>

HeartBeat Port

Sending Interval

Type=integer Range=40001 Default=40001

Type=integer Range=50-2000 numbers

This is the port for all servers to use to send or receive reliability messages. This is a read- onl y fiel d .

This is the interval in milliseconds between reliability messages.

Default=250

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Configuration 81

Table 22 Transport Management tab field descriptions (Sheet 3 of 3)

Field Value Description

HeartBeat Timeout

Discovery Period

Active Pending Period

Type=integer Range=1-10 numbers Default=3

Type=integer Range=2-60 numbers Default=3

Type=integer Range=2-10 numbers Default=4

This is the number of seconds before a server is declared failed.

This is the number of seconds a server stays in Discovery Mode.

This is the number of seconds a server stays in Active Pending Mode.

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Figure 30 Transport Management tab subfields, cont’d

{

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Table 23 Transport Management tab subfield descriptions, cont’d (Sheet 1 of 2)

Field Value Description

A Type=checkbox

Default=unchecked

HB Address: HB Address Enabled

Type=checkbox Default=unchecked

Use these checkboxes when you want to delete a checkbox’s relevant section, indicated by a bracket in the figure ab ove .

Check this checkbox to indicate that the a ddress i s enabled.

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Configuration 83

Table 23 Transport Management tab subfield descriptions, cont’d (Sheet 2 of 2)

Field Value Description

HB Address: Remote Server Address

HB Address: Local Server Address

Type=string in the form of a valid IP address

Type=string Range=1-50 characters Default=Private_Static_Address, Public_ Static_Address

This is the reliable IP address for a group server.

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Figure 31 Transport Management tab subfields, cont’d

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{

Table 24 Transport Management tab subfield descriptions, cont’d (Sheet 1 of 2)

Field Value Description

B/C Type=checkbox

Default=unchecked

NSD Number Type=integer

Range=1-3 numbers Default=1

Use these checkboxes when you want to delete a checkbox’s relevant section, indicated by the brackets in the figure above.

This is the unique number for this Network S er v i ce Descriptor (NSD).

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Configuration 85

Table 24 Transport Management tab subfield descriptions, cont’d (Sheet 2 of 2)

Field Value Description

NSD Enabled Type=checkbox

Default=unchecked

Service Parameter: Label

Type=string Range=1-15 characters Default=Private_Service_Address, Public_Service_Address

Service Parameter: Value

Type=string in the form of a valid IP address Range=1-15 nu mbers

Check the box if this NSD is enabled.

This is a unique label that references the value.

This field contains the valu e to assign this label (IP address).

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Figure 32 Transport Management tab subfields, cont’d

{

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Configuration 87

Table 25 Transport Management tab subfield descriptions, continued

Field Value Description

D Type=checkbox

Default=unchecked

Interface Config ur ation: Transport Enabled

Interface Config ur ation: Protocol

Interface Config ur ation: Transport

Type=checkbox Default=checked

Type=string Default=SIP

Type=string Range=TCP or UDP Default=UDP

Interface Config ur ation: Address

Type=string Range=1-50 characters Default=Private_Service _Address, Public_ Service_Address

Interface Config ur ation: Port

Type=string Range=1024-65535 numbers Default=5060

Use this checkbox when you want to delete a checkbox’s relevant section, indicated by the bracket in the figure above.

Check the box if this Transport is enabled.

This is the protocol t his interface supports.

This is the transport for this interface.

This is the service IP address label.

This is the port for this interface. The rang e is 1024 to 65535.

Interface Config ur ation: Params

Type=string Default=name=sipLscCo

These are the optional parameters for this interface.

nduit:interface=qfe0

Note: Scroll down to repeat entries in these subfields as needed.

T ransport Management: Active-Hot Standby Server Heartbeat Mechanism

The Transport Manager (referred to as the reliability managed object) is responsible for the heartbe at communi cation and st ate maint enance between the Application Modules. This software is responsible for monitoring the health and communicating the status of a Network Service Descriptor (NSD) across servers. Status communication and heartbeating occurs across all provisioned interfaces.

The heartbeat data in the Transport Manager defines a server group. Each server in a group includes provisioning for the static public and private addr ess of every othe r server in that group. Each server in a

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group consisting of four servers has provisioning for three public static heartbeat addresses and three private static heartbeat addresses.

The only time whe n this condition m ay not hold is wh en a new serve r is being introduced into an existing g r oup that is currently providing service. In this case, the new server is provisioned with the information for all the other serve rs while the other servers ar e not yet updated with provisioning f or the new server.

The reliability manager is configured by provisioning of engineeringand network-rela te d parameters. The engineering parameters determine the timeout intervals and failure detection thresholds. The NSD parameters defi ne the visible network interfaces used by the reliable services. The reliability manager service defines all the NSD data for the set of servers.

Note that checkboxes exist withi n several of the configuration data areas to enable or disable use of the data. In some cases, not all items are provisioned. For these cases, do not select those checkboxes.

The provisioning also allows for the use of label/value pairs. Where noted, you can use an add ress label in place of an explict a ddress. This capability simplifies the configuration process where the same data are provisioned multiple times. For the Transport Manager configuration, the Module P arameter label/va lue p airs are valu es associa ted with the entire component (the SIP Application Module). The Service Parameters apply only to the service instance.

The default configuration includes, in the Server Parameter fields, Public_S tati c_Address and Pr ivate_S tat ic_Address labe ls. These refer to the fixed IP addresses of the server. In the Network Service Descriptor area, the Service Parameter label/value pairs define the Public_Service_Address and the Private_Service_Address. These labels refer to the service addresses for a particular service instance that could be enabled on any one of many servers in the service NSD group.

Active-standby server group configuration

Configuration of an active-standby server group occurs as part of the normal deployment process. When servers are deployed, the administrator is prompted for configuration data specific to that component. During deployment, the administrator will see a configuration tab called Transport Manager. There is one set of fields for the engineering parameter data, one set of fields for the Heartbeat Parameters, and multiple sets of fields for the NSD data.

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Each NSD bean describes the network interfaces and protocols to be managed by the service being deployed under normal operating conditions. Configuration of the servers in the active-standby group occurs independently. The system manager is not aware of any relationship betwee n the servers. The refore, t ake care to confi gure the server group so that the reliability service functions properly.

When using the reliability manager , the administrator must ensure that conflicts with other managed-objects do not occur. Configuration data for the reliability manager replaces similar configuration data that may have previously been found in the configuration data of other managed-objec ts. The reliability mana ger internally launc hes network services by communicating with other manage d-ob jects in the system (through the service registry). The reliability manager passes this data to the transport controller during system initialization and state transitions.

When provisioning for reliability , leave the SIP transport parameters in the SIP Configuration tab blank. A set of equivalent fields are provisioned in the Transport Management dialog box instead. All other provisioning is unaffected.

Configuration 89

The SIP Application Module, when running in reliable mode, requires public and private service addresses for each service instance (a service instance is a “ virtual” applicati on server that can exis t on one of any number of physical servers). These service addresses are what other clients and servers use to communicate with the application server instances (note that the Management Module is configured to use the static addresses of the previous section).

A 1+1 reliable SIP Application Module configuration (one active and one standby server with one servic e instance) needs seven addre sses on the public network and seven addresses on the private network (total for both se rvers).

Configuration of the NSD is what defines those SIP Applicatio n Module network services that require reliability. If there are two physical servers in a 1+1 configuration, there must be one active NSD. Each active NSD describes the SIP services to activate on an ac tive server. The servers in the group negotiate which NSDs each will activate. The server that finds all NSDs already activated automatically becomes a standby server.

Each enabled NSD must define a unique public and private service address and may define other instance specific properties. Note that the public and private ser vi c e ad dress tag values (their provisi on ed IP addresses) should be different from the provisioned static addresses.

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Additional SIP TCF Base tab configuration information

This section contai ns additional configuration information for the IP an d port properties under the SIP TCF Base tab.

TCF Config details

SIP server protocol, network, and ports are started based on the information in this parameter. The TCF Config syntax for this parameter follows:

You can define multiple transports by continuing this format with a semicolon separator.

The supported transports for SIP are UDP, TCP, and SSL. The format of the optional parameter in the configuration string is specific to each supported transport type and, for some transports, configuration information in the optional parameter is mandatory.

For TCF Config parameter values, refer to Table 26, “UDP/TCP/SSL Config values.”

Table 26 UDP/TCP/SSL Config values

Parameter Value Description

Host Address IP v4 address This field con tains the address of the

host on which you want the connection to open.

Host Port Integer This field contains the port on which

to open the conn ection. The standard port for SIP is 5050.

Optional hostaddr:<I Pv 4> The hostaddr valu e specifies the

primary public address of the SIP Applicatio n Mo du le.

TCF Config parameter example 1, UDP, TCP

Example

UDP=192.168.0.1:5060;TCP=192.168.0.1:5060

Note: Do not use the IP addresses from this example in your network.

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This example creates a gen eral SIP server for TCP and UD P and starts execution of two SI P server ports on ne twork address 192.168.0.1 . The first server uses UDP/IP transport listening on port 5060. The second server uses TCP/IP accepting connections on port 5060.

UDP=192.168.0.1:5060:hostaddr:47.249.32.64

This example creates a SIP server for UDP on all interfaces on port

5060. This must be used on all application servers that span

public/private netw or ks. This example creates an SSL server that can be used for secure

communications with an SSL client application.

Example

UDP=192.168.0.1:5060;TCP=192.168.0.1:5060;SSL=192 .168.0.1:7020

Note: Do not use the IP addresses from this example in your network.

Configuration 91

This example combines the five previous examples into one example that shows all transport services starting together on a single SIP server.

Note 1: Each transport specification is separated by a semicolon. Note 2: Be sure to avoid address and port conflicts, which can cause

service startup failure, and require re-configuration and server restart.

Retransmission Of f parameter

When Retransmission Off is false, the SIP server follows the retransmission policies id en ti fi ed by the RFC 2543 (see note for specific reference) specification for SIP.

Note: J. Rosenberg et al, SIP: Session Initiati on Pr oto col , Inte rn et Draft draft-ietf-sip-rfc2543-bis09.txt, IETF, Feb 27, 2002.

When Retransmission Off is true, the SIP server does not retransmit SIP messages. This value is not changeable.

Invite Timer parameter

The Invite Timer value specifies the number of milliseconds a non-finalized SIP Invit e transac tion can remain open bef ore it is forced closed. A SIP Invite transaction, having received a provisional response and wa iting on a final r esp on se, i s a llo we d to persist only a s

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Time Transaction parameter

Add Defaults parameter

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long as this timer setting. Expiration of this timer causes resources allocated to the transaction to be released. Activation of this timer is controlled by the Time Transaction parameter.

When the Time Transaction value is true, the default value of 128,000 mS is used.

When the Time Transaction value is true, all transactions are timed. Invite transactions a re fo rc ed closed a nd th e Invite Timer dur a tion has expired. For all other transactions, the time-out duration is fixed at 64000 mS.

When the Add Defaults value is set to true, SDP message bodies in SIP messages with missi ng man dator y SDP heade rs a re re gener ated with default mandatory headers. This occ urs when messages are proxied through the SIPApplication Module.

Note 1: Set this parameter to true when downstream servers fully support the SDP specification.

Note 2: Set this paramete r to false when downstr eam servers do not fully support the SDP specification.

Check Mandatory Headers parameter

When the Check Mandatory Head ers value is true, SDP messages are screened for required header content.

Note: Missing headers cause message rejection.

When the Check Mandatory Head ers value is false, SDP messages are not screened for required header content.

OAM&P strategy

The SIP Application Module is fully integrated with the Management Module. Perform a ll configuration at the Management Console window . For additional information on the Management Module , refer to the

MCP Management Module Basics and the MCP System Management Console Basics documents.

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Accounting management

The SIP Applicat ion Module d oes not do any a ccounting manage ment. For more information on accounting, please see the MCP Accounting Module Basics document.

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Performance management

The Management Module manages the performance functions for the SIP Applicatio n Module. For addition al information on t he Management Module, refer to the MCP Management Module Basics and the MCP System Management Console Basics documents.

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Security and Administration

How this chapter is organized

This chapter is organized as follow s:

• “Security” on page 97

• “OAM&P strategy” on page 97

Security

The SIP Applicati on Module with Back-to-B ack User Agent functional ity controls the Media Portal (Media NAT) over an MGCP-type protocol. The SIP Applicatio n Module ensures security o f clients and the ne twork in the following ways:

• Uses MGCP+ to communicate with the Media Portal (over the private LAN) to control which ports are opened or closed.

• All signaling traffic traverses the SIP Application Module. It is the only node to which clients terminate SIP signaling.

• Hides address assigned by the Enterprise NAT from other users.

• Helps main tain con nection to cli ents thr ough NAT and/or firewall by the keep-alive mechanism.

• Provides client aut hen ti cat io n.

• Port 5060 is the only port required to be opened on the public interface.

• The SIP Application Module is managed from the private LAN. A management inte rface is not available from the public in terface.

OAM&P strategy

The Management Mo dule performs the security and adminis trative functions for the SIP Applic ation Mo dule. For addi tional info rmati on on the Management Module, refer to the MCP Management Mo dule Basics and the MCP System Management Console Basics documents.

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Appendix A: Basic call flows

Using SIP as the signaling protocol to establish a communi cat ion path between endpoint s, the SIP App licati on Modul e provid es the followi ng call services:

• Voice plus video

• Call transfer

• Authentication

The following secti ons provide sample diagrams an d descriptions of the call flows that en able these specific services. For an overview of a basic call flow, see the MCP Basics document.

Voice plus video

Figure 1, “Client-to-client voice plus video diagram,”and Figure 2, “Client-to-client voice plus video call flow,” show the basic call flow for a client-to-clie nt voice- plus-vi deo call . Each clie nt in th e diagram has a User Agent. The SIP Application Module provides Back-to-Back User Agent service, treating each SIP call as an independent Ingress and Egress leg. A detailed, step-by-step illustration follo ws this diagram.

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100 Appendix A: Basic call flows

Figure 1 Client-to-client voice plus video diagram

Bearer path

Signaling path

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Nortel Networks NN10029-111 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Overview

How this chapter is organized

Overview

Functional description

Network configuration

Interfaces

Protocols

Hardware

Services and features

Routing and Translation services

Foreign termination

Call Transfer service

Local termination

Telephony routing

Route lists The T elephony routing service is an enhancement to the Location Server on the SIP Application Module. This enhanced Location Server function has the ability to translate PSTN numbers into

SIP Aliases

Multiple Route Termination

Call Processing Language

Interworking services

Discriminator service

Bearer Path Control

Privacy Control service

Service package enforcement

Authentication services

Converged PC service

Network/Address Hiding service

Enterprise Clients

911 Notification support

Instant Messaging

Presence

Voicemail server interoperability and MWI

Registration—static and dynamic

Network address book

Overload control

Reliability and fault tolerance

Manual failover

Restart If you want to simply force an immediate manual fail-over, then you can request a Restart from the Management Console.

OAM&P strategy

Upgrades

Updating the SIP Application Module software

OAM&P strategy

Fault management

How this chapter is organized

Alarm clearing procedures

Configuration

How this chapter is organized

Overview

Adding a component

Configuring the SIP Application Module tabs

T ransport Management: Active-Hot Standby Server Heartbeat Mechanism

Active-standby server group configuration

Additional SIP TCF Base tab configuration information

TCF Config details

Retransmission Of f parameter

Invite Timer parameter

Time Transaction parameter

Add Defaults parameter

Check Mandatory Headers parameter

OAM&P strategy

Accounting management

Performance management

Security and Administration

How this chapter is organized

Security

OAM&P strategy

Appendix A: Basic call flows

Voice plus video