Nortel Networks Computer Accessories NN10029 111 User Manual

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 description

— 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 page 22

— “Voicemail server interoperability and MWI” on page 22

— “Registration—static and dynamic” on page 24

— “Network address book” on page 25

— “Overload control” on page 25

— “Reliability and fault tolerance” on page 26

• “OAM&P strategy” on page 28

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Overview

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 subscription and notification (For more

information on the Presence feature, see the MCP SIP Presence

Basics document.)

The SIP Application Module handles SIP sessions and 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 different

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 whether to process the request as a pure Proxy or

BBUA.

The Proxy Server processes SIP requests and responses, rewrites

headers, modifies request-URIs (Universal Resource Indicator),

performs location look-up, and forwards requests 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 maintains 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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Overview 5

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

Application Module remains on the signaling path

— modifying the Session Description Protocol (SDP) using values

supplied by the RTP Media Portal to control media endpoints

— 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 components. This is important for services such as billing,

which need to be aware of all events that take place on a session.

The BBUA in the network also provides a barrier for clients that 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 service

User

Agent

Client

User

Agent

Client

userA

userB

Internal Protocol

Routing in a SIP network is based on the same hop-by-hop principle

as routing e-mail within the Internet. The next hop for a SIP request

is determined by a proxy using the domain or the host part 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

SIP Application Module. The SIP Application Module supports

routing using table lookup in the SIP database or using the 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 decision is made

based on subscriber service logic. If the decision is to redirect the

request, a 302 Response 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), into a single server with the enhanced services provided

by the Back-to-Back User Agent.

Note: J. Rosenberg et al, SIP: Session Initiation Protocol,

Internet Draft draft-ietf-sip-rfc2543-bis09.txt, IETF, Feb 27, 2002.

Network configuration

The SIP Application 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

SIP Application Module and RTP Media Portal with public ports and

ports that are internal to the private network. This network configuration

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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Overview 7

Figure 2 Example of network configuration

SIP Application

Module

BigMart.com

Management

Module

SIP Audio

Servers

BiggerMart.com

Accounting

Modules

PSTN Gateways

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Interfaces

The SIP Application Module interfaces with numerous other

components. See Figure 3, “Network interfaces.”

Figure 3 Network interfaces

Private network

PSTN

or PBX

Public network

Legend

Access

client

Access

client

Signaling

Media stream

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Overview 9

Protocols

The SIP Application 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

DTP

SQL

SIP

OMI

PCP

Private network

PSTN

or PBX

MGCP+

SIP

Public network

SIP

Access

client

SIP

Access

client

Legend

Signaling

Media stream

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T a ble 1, “SIP Application 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

• SIP clients

• SIP Audio Server

• IP Client Manager (IPCM)

• SIP PRI Gateway

• Web Client Manager

OMI

Open Management Interface

Interface to the SIP Management Module

MGCP+

Media Gateway Control Protocol

Interface to the RTP Media Portal

PCP

DTP

Perfect Channel Protocol for logs and alarms going to

the Management Server

Data Transport Protocol

Note: The external interfaces use an IP network to interconnect the

components listed in this table.

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Hardware

Refer to T a ble 2, “Minimum hardware requirements,” for the list 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

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

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

— Bearer Path Control

— Privacy Control service

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• Service package enforcement

• Authentication services

• Converged PC service

• Network/Address Hiding

• 911 Notification support

• Instant Messaging

• Presence

• Voicemail server interoperability and MWI (message waiting

indication) notification

• Registration

• Network address book

• Overload control

• Reliability and fault tolerance

Routing and Translation services

Foreign termination

If an incoming request specifies a domain that is not served 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 order 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 for a particular domain and uses that protocol field to

provide the correct IP address of the server for the specified 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 differ from, for example, the H.323 server). This allows 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 SIP Application Module attempts to look up the foreign

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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Overview 13

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-record is the traditional 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 fail, the SIP Application Module rejects 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 Application Module supports unattended Call Transfer through

the Refer mechanism. Unattended Transfer (or Blind Transfer) refers to

cases where the transferor redirects the transferee to the transfer target

without first conferring with the transfer target. The transferor 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.

Local termination

The SIP Application Module first determines whether the incoming SIP

request terminates to a client in a domain managed by the

SIP Application Module. The SIP Application Module 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 the URL is a Telephony routing

number, the Telephony routing number is sent through the Telephony

routing software within the Location Server.

The Telephony routing software must perform digit translation to find a

gateway to terminate a call to. These tables 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 T e lephony routing stages,” shows the

relationship between the telephony routing stages provided by the SIP

Application Module. If the COS value of the subscriber and subdomain

route do not match, then the SIP Application Module checks the parent

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

DNS A

record*

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

Route lists The Telephony 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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Overview 15

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 originator’s domain. This option can be used

to redirect an incoming request from another domain that is routing

to a restricted route list.

Route lists consist of

• private telephony routes, which are used for private telephony-style

digit dial plans

• gateway routes, which provide access to the gateways

• SIP telephony routes, which point to other SIP Application Modules,

and SIP domains and subdomains for interdomain routing using

telephony-style dial plans

SIP Aliases

Alias URLs can be used to refer to a SIP client in the network. For

example, a user “sip: userA@domainX.com” can also be referred 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 precedence over gateway routing

translations, and routing information pertaining to userA is retrieved. If

an alias of “sip:41037@domainX.com” is not configured, then gateway

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 user routing preference, the routes available, and routing

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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With simultaneous ringing, 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 sequential ringing, the call tries to terminate to only 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.

Call Processing Language

The SIP Application Module supports the use of the Call Processing

Language (CPL), based on the IETF CPL draft, draft-ietf-iptel-cpl.txt.

SIP clients can change 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 process that is actually done through the Call Manager

in the Personal Agent. Third-party clients can also upload scripts using

the Registration mechanism. The Registration function of the

SIP Application Module stores the request. When the SIP Application

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 SIP Application 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 SIP Application Module screens requests bound for devices that

are not fully SIP compliant, for example, the Communication Server for

Enterprise (CSE) 2000. These components cannot process all types of

signaling and certain media change requests. Therefore, the

SIP Application 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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Overview 17

devices. Information for each component is stored in .xml format to

provide flexibility when describing the capabilities of the component.

Bearer Path Control

The SIP Application Module uses the RTP Media Portal to control

media streams originating from and terminating to non-compliant SIP

devices if they do not support media negotiations. The exception to this

occurs when the originating and terminating parties are both the same

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 not to pass that information outside

the boundaries of the trusted network. The information is passed in a

Remote-Party-ID header with the privacy indicator set to “full.” The SIP

Application Module removes this header any time it forwards the

message out over a public network interface.

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 authentication when the

server receives an incoming SIP request. The SIP Application Module

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 request with a valid password and user name attached.

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 the note 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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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 proxies in the NodalAuth field of the Authentication tab.

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. Subscribers for a particular

system belong to local domains. Local domains are provisioned

through the Provisioning Client.

• Foreign Domain: A foreign domain is a domain that is either

provisioned as foreign for this SIP Application Module or not

provisioned at all for this specific system. It basically represents a

domain that is not served.

Converged PC service

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 mirrored calls to the SIP Application

Module through the SIP PRI Gateway. This allows the SIP Application

Module to present a call window on the end user's PC when the user'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 existing telephony switch to add

multimedia capabilities.

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The Converged service adds the following capabilities 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 Network Address Translation

(NAT) devices change port and address information in the IP packet

header, most are not currently SIP or SDP aware. IP addresses in these

messages are therefore sent out unchanged through the NAT. If the SIP

Application Module were to forward these messages on unchanged,

sensitive IP information would be given to untrusted clients. In order to

remedy this, the SIP Application Module sanitizes the messages before

forwarding them.

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 header). A media portal is necessary in order to replace the IP

information in the SDP headers. The SIP Application Module queries

the Media Portal (using MGCP+) for a new IP and port combination to

replace the IP and port put there by the client. This effectively anchors

the media stream at the Media Portal.

Clients therefore see the SIP Application Module as their signaling

endpoint and the Media Portal as their RTP media endpoint. 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 RTP Media Portal, refer to the MCP RTP Media

Portal Basics document.

Note: The SIP Application Module cannot map SDP information

without an RTP Media Portal. It only performs address mapping for

SIP header fields. Therefore, SDP passes through untouched. If the

server must map SDP address information, then you need an RTP

Media Portal.

The SIP Application Module is configured to use an RTP Media Portal

to originate and terminate media streams (RTP/RTCP). The

SIP Application Module uses extended Media Gateway Control

Protocol (MGCP+) to allocate and release resources on the RTP Media

Portal for each session as needed.

Enterprise Clients

The SIP Application software uses the RTP Media Portal to hide

sensitive IP address information about SIP clients behind a firewall in

an Enterprise network. The exception to this occurs when the originator

and terminator of the request are both part of the same network. This

status is determined by checking the domains in the From header and

Request-URI of the SIP Invites. If both SIP clients belong to the same

Enterprise network, the SIP Application Module does not use the RTP

Media Portal. Administrators can override this behavior by provisioning

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

Enterprise

Domain

Public

Domain

SIP

Client

SIP

Client

SIP

MGCP+

RTP/RTCP

Media Portal

911 Notification support

The SIP Application Module supports Instant Message notifications to

a specified On-Site Notification (OSN) location whenever a user makes

a call to an emergency number such as 911. The software provides this

service using the same mechanism that allows users 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 subscriber. Since telephony routes are

only unique within a subdomain, you cannot have more than one OSN

location for each subdomain.

For each new emergency subscriber that the administrator creates,

there must be both

• an emergency number to route to the Public Safety Answering Point

(PSAP)

• a SIP subscriber assigned to the OSN location that is to receive the

notification.

Each OSN location must have a specific subscriber assigned, such

as sip:guarddeskA@nortelnetworks.com.

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For more information and the procedure for setting up Instant Message

notifications to emergency numbers, see the SIP Provisioning Client

User Guide.

Instant Messaging

Instant Messages are routed in parallel only to a subscriber's

dynamically registered routes (see “Registration—static and dynamic”

on page 24). This is in contrast to session initiation requests, which 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.

Presence

When a user initially registers, by default their presence status is set to

“on-line” in the SIP registration message. Users subscribe to watch the

status of other users, and to coordinate the status of their own devices.

This information is maintained in an in-memory table on the SIP

Application Module (Presence software). The information that is stored

in this table includes:

• the user to be watched

• the party requesting the subscription

• the correlation information identifying that particular subscription

request

• contact information regarding where to send the notifications that

are generated as a result of the subscription being active

When a user changes 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 presence state was. If the update causes a material

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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Overview 23

transmits the following information over a data link to a voicemail

server:

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

• the calling number

• 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 sends Notify messages for MWI

information. The software uses Real-Time Transport Protocol (RTP) to

carry the voice media.

There are two configurations through which the SIP Application Module

supports voicemail:

• 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.

• 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 supports 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 considers:

• MESSAGE DEPOSIT: An incoming call for a subscriber gets routed

to voicemail because the called subscriber is unavailable, busy, or

has all calls forwarded to voicemail.

• 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 document 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 message 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 Module. The SIP Application Module must respond to this

message to let the voicemail server know that the SMDI link is still up.

Registration—static and dynamic

Registration can take two forms:

• Static

Users or administrators can perform static registrations. With static

registration, the user can obtain a presence when not logged into

the network. The user can obtain a presence and an account 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 care of this re-registration automatically for

the i2004. Dynamic registration is automated and behind the

scenes.

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Overview 25

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] interface or direct database access depending

on the client).

In order to receive updates to the Address Book after the initial

download, the client subscribes 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

incrementally update their view of the information (again either through

a SOAP interface or direct database access depending on the client).

A List of Buddies is incorporated as part of the Address Book. Each

subscriber must create their own personal Address Book and designate

their own Buddies. For each of these specified entries, the client

automatically subscribes to their presence event package. This allows

them to monitor and update the network presence of each Buddy (for

example, online or offline).

Overload control

Overload Control monitors the Incoming Protocol Message Queue

Length. If this queue length crosses a configurable threshold value, the

system performs Session Blocking, allowing no new incoming requests

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 appropriate actions are invoked and are not cleared until all

aspects of the system have crossed below the assigned threshold

value.

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Reliability and fault tolerance

The SIP Application Module provides reliability and fault tolerance

through multiple SIP Application Modules 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 current states, including the NSD activated on them.

An initializing server configures itself with one of any inactive NSDs. If

all NSDs are active, the initializing server becomes the standby. This

prevents conflicts where more than one 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 activate and raises an

alarm. The alarm clears when the resources become available.

When there are two or more active servers, the group is called a cluster.

You can configure both the N+M strategy 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 calls that have already been established continue and

the parties maintain voice path. Any future requests during that session,

however, fail (for example, Hold, Retrieve, and Web Pushes) since the

session information is no longer available. Any sessions that were not

in the active state before the failover are lost. The originating clients of

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Overview 27

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" server, then request a LOCK from the

Management Console. The LOCK forces 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 SIP

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 be

guaranteed. For example:

Sessions involving SIP clients will survive until the

clients encounter a "no response" or "unknown call"

response to a request on their active session. At that

point the clients will release the session and its

associated media resources.

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.

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 Module Basics and the MCP System Management

Console Basics documents.

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Upgrades

For information on upgrading from one full release to another, refer to

the Installation and Commissioning document you receive with 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 Management Console Basics

document.

at the System Management Console

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

Select the Update command. The following window appears.

Figure 3 The update window, retrieving the load list

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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 load being updated (see Figure 4, “Load list

for updating”).

Figure 4 Load list for updating

Select the version you want to update. Click on the Apply

button.

The configuration window appears, showing the tabs. 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 managed object (MO) appears as being successfully

added onto the managed element (ME).

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Figure 5 Progress of update

Once the update has completed, the following window appears.

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 additional information on the Management Module, refer to

the MCP Management Module Basics and the MCP System

Management Console Basics documents.

How this chapter is organized

This chapter is organized as follows:

• Alarm clearing procedures

• Recovery procedures

Alarm clearing procedures

Procedure 1 Clearing the SLE701 (SLEE Health Monitor) alarm

at the alarm browser

The SIP Application Module raises this warning alarm under one

of two conditions:

• The number of application contexts (AC) available for use

are inadequate for the level of traffic (in which case the

administrator needs to back off the traffic 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 causing 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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framework. For example, when you register your phone, a

number of AC instances are invoked to process the registration.

Procedure 2 Clearing the SMDI101 alarm

at the alarm browser

This alarm is raised when the Simplified Message Desk

Interface (SMDI) telnet session between the SIP Application

Module and a terminal server is lost. The SIP Application Module

uses the SMDI protocol to communicate information between

itself and a voicemail server. If the connection goes down,

• Message Waiting Indication 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. Also, administrators should

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 procedures for configuration required at the

service provider premises. This chapter is organized as follows:

• “Overview” on page 38

• “Adding a component” on page 40

• “Configuring the SIP Application Module tabs” on page 44

— Application Server

— Long Call Service

— Presence

— Presence Location Service

— Authentication

— Media Portal

— Locate User Svc

— Database Base

— Data Synchronization

— Prov Sync Service

— Overload Controls

— Forward Location Service

— Local Accounting Manager

— In Memory Database

— Location Service

— Registration

— Server Properties

— Server Subscription

— SIP TCF Base

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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

Overview

CAUTION

Before making any changes to the base

configuration, consult your next level of support.

Nortel Networks performs the initial installation and commissioning.

Once the installation and commissioning are completed, 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 is 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 Module Basics and the MCP System Management

Console Basics documents. This chapter describes the configurable

parameters affecting operation of the SIP Application Module and the

procedures for configuration required at the service provider 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 removes

all related software and configuration data. When the deployment is

complete, the SIP Application Module should be unlocked, enabled,

and available to provide service.

Before a SIP Application Module can be deployed, the server must

have been configured at the System Management Console. This server

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Configuration 39

represents the physical hardware on which the SIP Application Module

resides. Once the servers are configured, the SIP Application 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 familiarize themselves

with the available options.

Administrators can also find help text with descriptions and acceptable

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

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Adding a component

This procedure assumes that the server on which the SIP Application

Module will be deployed 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

To 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

Select Add->Component as shown.

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Figure 3 Adding a component

Select the SIP Application Module software load you want from

the load list that appears (see Figure 4, “Load list”) and click on

the Apply button. There may or may not be multiple software

loads to choose from.

Figure 4 Load list

The Configuration window (shown in Figure 5, “Configuration

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 Configuration window (top half)

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 services that the SIP Application Module requires.

The following sections describe each tab in detail and provide

guidance on how to configure the SIP Application Module. Many

of the fields are already filled with default values. Administrators

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.

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

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 all 0.0.0.0 IP addresses have been replaced

with the correct IP address. Verify other non-default parameters

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

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

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

Field

Value

Description

Show GUI

Type=checkbox

Default=unchecked

This is a read-only field.

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

Opaque

Headers

Type=string

This field contains a “+” delimited

list of headers that should not be

Range=0-2048 characters

Default=to+from+call-id+via+ passed through the server.

cseq+content-type+content-

length+contact+record-route+

route+proxy- require+rseq

Private IP

Address

Type=valid IP address

Range=0-4096 numbers or

blank

This field contains the private IP

address of the server.

Note: Do not leave this field blank

or the software will not deploy.

Default=0.0.0.0

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

allows the service provider to set the length of time between

endpoint audits. The Long Call Service detects abandoned calls

and releases the resources used by such calls.

Figure 9 Completing the Long Call Service tab fields

Table 4 Long Call Service tab field descriptions

Field

Value

Description

Duration

Type: Integer

This field shows the length of time 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.

Range: 0–Max_Integer

Default: 60 minutes

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

to configure context and expiration information for the Presence

service.

Figure 10 Completing the Presence tab fields

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

This field indicates the initial number

of contexts to create. This should not

exceed the maximum number of

contexts.

Default=1

maximumNumberOf

Contexts

Type=integer

Range=1-15000

Default=15000

This field indicates the maximum

number of contexts to create.

maximum Expires

Type=integer

This read-only field contains the

Range=60-86400 maximum allowable expiration value

Default=3700

for a presence subscription request,

in seconds.

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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

Type=integer

This read-only field contains the

maximum number of idle contexts

at any time. It should not exceed

the maximum number of

contexts.

Number of Idle Range=1-100 numbers

Contexts

Default=10

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

This read-only field contains the

maximum number of contexts to

create.

Use DNS

Type=checkbox

Default=unchecked

Turns DNS server functionality on

and off.

Use Location

Server

Type=checkbox

Default=unchecked

Turns Location Server

functionality on and off.

DNS Provider

URL

Type=string

Range=1-1024 numbers

Default=dns://0.0.0.0

This field indicates the address of

the DNS Server format

>dns://0.0.0.0

DNS Default

Transport

Type=string

Range=udp, tcp

Default=udp

Transport type used to

communicate with the DNS

server.

Location

Server URL

Type=string

This is the address of the

Default=sip://0.0.0.0:5060:udp Location Server.

Location

Type=string

This field is not used.

Server Timeout Default=3600

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Click on the Authentication tab. The Authentication tab enables

or disables authentication for requests and sets additional

authentication information.

Figure 12 Completing the Authentication tab fields

Table 7 Authentication tab field descriptions (Sheet 1 of 2)

Field

Value

Description

Methods to Type=string

This field indicates which SIP methods to

authenticate.

Authorize

Realm

Default=register

Type=string

This field indicates the string that is

Range=0-256 characters displayed to the user to indicate what

realm they need to supply a password for.

Private Key Type=string

An extra key used to uniquely generate

Range=0-256 characters authentication challenges.

Default=MCP

Nonce

Interval

Type=integer

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.

Range=10000- 600000

milliseconds

Default=600

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Table 7 Authentication tab field descriptions (Sheet 2 of 2)

Field

Value

Description

This field contains a + -delimited list IP

Authorized Type=IP address

SIP Nodes Range=0-2000 numbers addresses. Use the SIP PRI Gateway and

SIP Audio Server addresses.

Nodal Auth Type=checkbox

Default=unchecked

When this field is checked, the SIP

Application Module redirects requests.

When unchecked, this field authenticates

requests and only the SIP PRI Gateway

and SIP Audio Server listed in the previous

field can send INVITE messages to the SIP

Application Module without authentication.

Nortel Networks recommends that you do

not change this field.

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

provider to set port and firewall information pertaining to the

Media Portal.

Figure 13 Completing the Media Portal tab fields

Table 8 Media Portal tab field descriptions

Field

Value

Description

Initial

Capacity

Type=integer

Range=113-16384

Default=113

This field is not used.

Fire Wall

Type=checkbox

Default=unchecked

This field is not used.

MGCP Port Type=integer

This field indicates the UDP Communications

Range=1025-65535 port number where the Media Portal sends and

Default=3903

receives MGCP+ messages.

Click on the Database Base tab.

General properties for the SIP Application Module’s connection

to the database are defined in the Database Base tab. See the

MCP Database Module Basics document for more information

and field descriptions. Modifications to the Database Base

require that the Database Base be locked. A lock of the

Database base releases all SIP Application Module 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 Database Base is unlocked, all SIP Application

Module resources must be reallocated causing a re-read of the

resources from the database. This tab also contains connection

information for the database.

Figure 14 Completing the Database Base tab fields

Note: See the MCP Database Module Basics document for

field descriptions.

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Click on the Locate User Svc tab. This tab allows administrators

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

Figure 15 Completing the Locate User Svc tab fields

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

Field

Value

Description

DNS SRV

Default

Transport

Type=string

Range=UDP, TCP

Default=UDP

This field indicates the transport type

used to communicate with the DNS

SRV server.

DNS SRV URL

Type=string

Range=1-64 numbers

Default=dns://0.0.0.0

This field indicates the address of

the DNS SRV server.

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 communicate

with the Location Server.

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

Field

Value

Description

This field indicates the address of

Location Server

URL

Range=1-64 numbers

Default=sip://0.0.0.0:5065 the Location Server.

Use Location

Server

Type=checkbox

Default=unchecked

This field is not used.

Click on the Data Synchronization tab. This tab allows the

service provider to set the context and expiration information

relating to the synchronization of in-memory and persistent data.

Figure 16 Completing the Data Synchronization tab fields

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Table 10 Data Synchronization tab field descriptions

Field

Value

Description

maximumNumberofIdleContexts Type=integer

This is the maximum

number of idle contexts at

any time. It should not

exceed the maximum

number of contexts.

Range=1-10000

numbers

Default=100

initialNumberofContexts

Type=integer

Range=1-10000

numbers

This is the initial number of

contexts to create. It

should not exceed the

maximum number of

contexts.

Default=10

maximumNumberOfContexts

maximumExpires

Type=integer

Range=1-10000

numbers

This is the maximum

number of contexts to

create.

Default=1000

Type=integer

Range=60- 86400

seconds

This is the maximum

allowable expiration value

for a DataSync

subscription request, in

seconds.

Default=3700

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

parameter forwards provisioning modifications on user and

device records to the SIP Application Module whenever

modifications occur or when additions or deletions are made.

The Prov Sync Service tab allows the SIP Application Module to

keep its configuration data updated with any changes that are

made through the 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

Table 11 Prov Sync Service tab field descriptions

Field

Value

Description

Sync Time

Period

Type=integer

This field indicates how often (in

Range=5-60000 seconds seconds) the SIP Application Module

Default=10

queries the database for provisioning

changes.

Task Group Type=integer

Size Default=5

This is a read-only field.

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

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

Field

Value

Description

Mem Polling Interval Type=integer

This field indicates the number of

seconds to wait in between checks

on memory usage.

(sec)

Range=a positive

integer

Default=3

Minor Alarm

Threshold (%)

Type=integer

This field indicates the threshold at

Range=0-100 numbers which Minor overload is

Default=80

encountered for both CPU and

memory.

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

Field

Value

Description

This field indicates the threshold at

Major Alarm

Threshold (%)

Type=integer

Range=0-100 numbers which a Major overload is

Default=85

encountered for both CPU and

memory.

Critical Alarm

Threshold (%)

Type=integer

This field indicates the threshold at

Range=0-100 numbers which a Critical overload is

Default=90

encountered for both CPU and

memory.

Call queue high

threshold

Type=integer

This field contains the number of

Range=1-500 numbers elements in the queue that, if

Default=25

exceeded, causes the system to

disallow new calls.

Call queue low

threshold

Type=integer

This field contains the number of

Range=0-500 numbers elements in the queue that causes

Default=5

the system to allow new calls.

Other queue high

threshold

Type=integer

This field contains the number of

Range=1-500 numbers elements in the queue that, if

Default=15

exceeded, causes the system to

disallow other session types, such

as registrations, instant messages,

or subscriptions.

Other queue low

threshold

Type=integer

This field contains the number of

Range=0-500 numbers elements in the queue that causes

Default=2

the system to allow other session

types.

Database queue

high threshold

Type=integer

This field contains the number of

Range=1-500 numbers elements in the queue that, if

Default=5

exceeded, causes a cluster

overload.

Database queue low Type=integer

threshold

This field contains the number of

Range=0-500 numbers elements in the queue that causes

Default=0

the cluster overload to clear.

Protocols Monitored Type=string

Range=0-1024

Plus sign (+)-delimited list of

protocols whose IO queues are

monitored for excessive delays.

characters

Default=sip

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Click on the Local Accounting Manager tab. For more

information on the Local Accounting Manager tab fields, see the

MCP Accounting Module Basics document. This tab contains

information pertaining to the Accounting Manager and billing

records, including IP addresses, ports, file rotation size and time,

and recording unit queue size.

Figure 19 Completing the Local Accounting Manager tab fields

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

provider to set information relating to local domains, event

subscriptions, and the nonce used for authentication.

Figure 20 Completing the In Memory Database tab fields

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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-MaxInt numbers

Default=64

Set this to the number of

domains and subdomains

you expect to have.

IMDomainInfo table

sync interval

Type=integer

Range=300- 3600 numbers

Default=600

Set this to however often

you want the system to

scan for changes to

domain provisioning.

IMUserInfo table limit

Type=integer

Set this to the number of

users you expect to host.

Range=32768-MaxInt

numbers

Default=131073

IMUserInfo table sync

interval

Type=integer

Range=1800- 86400

numbers

Set this to however often

you want the system to

scan for changes to the

user’s provisioned

Default=1800

presence information.

IMDomainSubBan

table limit

Type=integer

Range=16-MaxInt numbers

Default=64

Set this to the number of

domains and subdomains

you expect to have.

IMDomainSubBan

table sync interval

Type=integer

Set this to however often

Range=300-86400 numbers (in seconds) the SIP

Default=900

ApplicationModulechecks

the database for changes

in the domain ban

information.

IMWatchList optimal

table size

Type=integer

Set this to the total number

of user ban list entries you

expect.

Range=32768-MaxInt

numbers

Default=32768

IMWatchList table sync Type=integer

interval Range=300-1800 numbers

Default=900

Set this to however often

(in seconds) the SIP

ApplicationModulechecks

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

IMWatchers optimal

table size

Type=integer

Set this to the number of

subscriptions expected at

any interval in time.

Range=65536-MaxInt

numbers

Default=131073

IMWatchers table

cleanup interval

Type=integer

Range=300 to 86400

numbers

Set this to however often

(in seconds) you want the

system to clear out any

expired event

Default=1800

subscriptions.

IMAuthNonce table size Type=integer

Set how many nonces you

want the system to keep

for authentication.

Range=1-10 numbers

Default=5

IMAuthNonce

generation interval

Type=integer

Default=60

Sets how often (in

seconds) the system

creates a new nonce. This

is a read-only field.

Subscriber cache

interval

Type=integer

Range=60-3600 numbers

Default=60

Set this to however long

you want the system to

cache, in memory,

subscriber information

from the database, in

seconds.

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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

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

maximumNumberOf

Contexts

Type=integer

Range=1-512 numbers

Default=100

This is the maximum

number of contexts to

create.

UseDNS

Type=checkbox

Default=unchecked

Turns DNS server

functionality on and off.

UseLocationServer

DNSProviderURL

Type=checkbox

Turns Location server

functionality on and off.

Default=unchecked

Type=string

Range=0-1024 numbers

Default=dns://0.0.0.0

This is the address of the

DNS server.

DefaultTransport

Type=string

Range=udp or tcp

Default=udp

Transport type used to

communicate with the

DNS server.

LocationServerURL

Type=string

Range=0-1024 numbers

Default=sip://0.0.0.0:5060:udp

This is the address of the

Location Server.

LocationServerTimeout Type=integer

Default=3600

This is a read-only field.

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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 Location Service tab field descriptions (Sheet 1 of 2)

Field

Value

Description

Maximum Number of

Idle Contexts

Type=integer

This is a read-only field. This is

Range=1-100 numbers the maximum number of idle

Default=10

contexts at any time. It should

not exceed the maximum

number of contexts.

Initial Number of

Contexts

Type=integer

This is a read-only field. This is

Range=1-100 numbers the initial number of contexts to

Default=10

create. It should not exceed the

maximum number of contexts.

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

Field

Value

Description

Maximum Number Of

Contexts

Type=integer

This is a read-only field. This is

Range=1-100 numbers the maximum number of

Default=150

contexts to create.

Use DNS

Type=checkbox

Default=unchecked

Turns DNS server functionality

on and off. Check this field if you

are using the DNSSvr service to

resolve foreign domains.

Use Location Server

DNS Provider URL

Type=checkbox

Turns Location Server

functionality on and off.

Default=unchecked

Type=string

This field indicates the address

of the DNS Server format

>dns://0.0.0.0

Range=1-1024

numbers

Default=dns://0.0.0.0

DNS Default Transport Type=string

Range=udp, tcp

Transport type used to

communicate with the DNS

server.

Default=udp

Location Server URL

Type=string

This is the address of the

Default=sip://0.0.0.0:50 Location Server.

60:udp

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.

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

Table 16 Registration tab field descriptions

Field

Value

Description

maximumNumberOf Type=integer

This is the maximum number

of idle contexts at any time. It

should not exceed the

IdleContexts

Range=1-10000 numbers

Default=50

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 Type=integer

This is the maximum number

of contexts to create.

Contexts

Range=1-10000 numbers

Default=5000

Valid Maximum

Expires

Type=integer

This is the maximum allowable

Range=60-86400 numbers expiration value for a

Default=86400

registration request, in

seconds.

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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

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

Field

Value

Description

Label: Type=string

This is a label that identifies the public

address for the server.

Range=1-80 characters

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

This is a label that identifies the private

address for the Server. Only add an

Default=Private_Service_Address entry for a standalone configuration.

Value: Type=string in the form of an IP

address (x.x.x.x)

This field contains the private machine

logical IP address of the SIP

Application Module assigned to the

label just above this field.

Label: Type=string

Range=1-80 characters

Default=blank

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.

Value: Type=string

Range=1-80 characters

Default=blank

This field contains 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.

Label: Type=string

Range=1-25 characters

Default=blank

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 field descriptions,” on

page 79, which contains the actual

value of private_static_address. In

other words, this is where 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 defined in the Transport

Management tab.

Value: Type=string

Range=1-25 characters

Default=blank

This field should contain the string

private_static_address.

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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

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

Field

Value

Description

maximumNumberOf Type=integer

This is the maximum

number of idle contexts at

any time. It should not

exceed the maximum

number of contexts.

IdleContexts

Range=1-10000 numbers

Default=1

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 Subscription tab field descriptions (Sheet 2 of 2)

Field

Value

Description

maximumNumberOf Type=integer

This is the maximum

number of contexts to

create. The range is 1 to

10000.

Contexts

Range=1-10000 numbers

Default=10

provServerURL

Type=string

The comma-delimited list of

Provisioning Modules this

Range=0-4096 characters

Default=0.0.0.0:5095:8000:http server is to communicate

with. Format is [IP

ADDR]:[SIP PORT]:[HTTP

PORT]:[PROTOCOL].

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 contains many

parameters pertaining to the SIP Application Module’s transport

configuration. It includes information regarding the transport 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.

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Figure 26 Completing the SIP TCF Base tab fields

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

Field

Value

Description

Transport Config Type=string

Specifies the transport, IP addresses,

and ports. Includes both the public

Default=UDP=0.0.0.0:5

060:optional:name=sipL and private interfaces. If your system

scConduit;TCP=0.0.0.0: consists of only public IPs, do NOT

5060:optional:name=sip duplicate the string. This field

LscConduit

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

This controls the maximum time in

milliseconds to wait for an INVITE to

receive a Final Response after

Range=120000-

3600000 numbers

Default=128000

receiving a provisional Response.

Application Type Type=string

This is the type of SIP Server on the

Range=callstate_server, node.

stateful_server,

stateless_server,

user_agent

Default=callstate_server

Time Transaction Type=checkbox

This field specifies whether the SIP

transactions should be timed. This

field is read only.

Default=checked

Add Defaults

Type=checkbox

Default=checked

Recommendation is to

check the box.

Specifies whether to fill in missing

mandatory headers with default

values in the SDP message bodies.

CheckMandatory Type=checkbox

Controls whether the mandatory SDP

headers are checked for presence in

the SDP messages.

Headers

Default=unchecked

Recommendation is

NOT to check the box.

Maximum

Number of

Redirections

Type=integer

Range=3-10 numbers

Default=5

Maximum number of redirections

allowed before a request is dropped.

Initial Maximum

Hop Value

Type=integer

Range=5-50 numbers

Default=20

Maximum number of hops allowed

before a request is dropped.

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

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 there are third-party clients, you

may want to check the box so that the SIP

Application Server will enforce the service

package.

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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 whether or not the SIP Application

Server will or will not forward messages to a foreign server.

Figure 28 Completing the SipFwdAdapter tab fields

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

Field

Value

Description

maximumNumberOf Type=integer

This is the maximum number

of idle contexts at any time. It

should not exceed the

IdleContexts

Range=1-10000 numbers

Default=50

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 Type=integer

Contexts Range=1-10000 numbers

Default=5000

This is the maximum number

of contexts to create.

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

Field

Value

Description

forwardSubscribe

Type=checkbox

Default=unchecked

If box is checked, the system

allows subscribe messages to

be forwarded.

forwardRegister

allowedEvents

Type=checkbox

If box is checked, the system

allows register messages to

be forwarded.

Default=unchecked

Type=string

This field indicates the valid

event packages for the SIP

Application Module to

process.

Range=0-512 characters

Default= presence,

address-book,

message-summary,

service-package

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

of subfields. The next series of screens and tables give

information on what data to enter and where to enter it. For more

information on the function of this tab, see the section following,

“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.

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Figure 29 Completing the Transport Management tab fields

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

Field

Value

Description

Service

Name

Type=string

Range=1-20 characters

Default=BBUA

This field indicates the name of the

service this Reliability Manager is

supporting.

Server ID

Type=integer

Range=1-4 numbers

Default=1

This field indicates the ID number

for this server.

StandAlone Type=checkbox

Is the server standalone or part of a

reliable group? Check the box if the

server is standalone.

Server

Default=checked

Label

Type=string

Range=1-15 characters

Default=Public_Static_Address

This field contains a unique label

that references the value.

Value

Label

Type=string in form of an IP

address

This field contains the value to

assign this label (IP address).

Type=string

Range=1-15 characters

Default=Private_Static_Address

This is a unique label that

references the value.

Value

Type=string in the form of an IP This field contains the value to

address

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

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

Range=1-15 alphanumeric

characters

Default=Service_Node_Name

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 name 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 also needs to

be entered on the CS 2000.

Value

Type=string in the form of an IP This field contains the value to

address

assign this label (IP address).

Range=<host name of node>

HeartBeat

Port

Type=integer

Range=40001

Default=40001

This is the port for all servers to use

to send or receive reliability

messages. This is a read-only field.

Sending

Interval

Type=integer

Range=50-2000 numbers

Default=250

This is the interval in milliseconds

between reliability messages.

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

Field

Value

Description

HeartBeat

Timeout

Type=integer

Range=1-10 numbers

Default=3

This is the number of seconds

before a server is declared failed.

Discovery

Period

Type=integer

Range=2-60 numbers

Default=3

This is the number of seconds a

server stays in Discovery Mode.

Active

Pending

Period

Type=integer

Range=2-10 numbers

Default=4

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

{

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

Field

Value

Description

Type=checkbox

Default=unchecked

Use these checkboxes

when you want to delete a

checkbox’s relevant

section, indicated by a

bracket in the figure above.

HB Address: HB

Address Enabled

Type=checkbox

Default=unchecked

Check this checkbox to

indicate that the address is

enabled.

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

Field

Value

Description

HB Address:

Remote Server

Address

Type=string in the form of a valid This is the reliable IP

IP address

address for a group server.

HBAddress:Local Type=string

Server Address Range=1-50 characters

This is the reliable IP

address for a group server.

Default=Private_Static_Address,

Public_ Static_Address

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Figure 31 Transport Management tab subfields, cont’d

{

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

Field

Value

Description

B/C

Type=checkbox

Default=unchecked

Use these checkboxes when

you want to delete a

checkbox’s relevant section,

indicated by the brackets in the

figure above.

NSD Number Type=integer

This is the unique number for

this Network Service

Descriptor (NSD).

Range=1-3 numbers

Default=1

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

Field

Value

Description

NSD Enabled Type=checkbox

Default=unchecked

Check the box if this NSD is

enabled.

Service

Parameter:

Label

Type=string

This is a unique label that

references the value.

Range=1-15 characters

Default=Private_Service_Address,

Public_Service_Address

Service

Parameter:

Value

Type=string in the form of a valid IP This field contains the value to

address

assign this label (IP address).

Range=1-15 numbers

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Figure 32 Transport Management tab subfields, cont’d

{

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Table 25 Transport Management tab subfield descriptions, continued

Field

Value

Description

Type=checkbox

Default=unchecked

Use this checkbox when you

want to delete a checkbox’s

relevantsection, indicatedby

the bracket in the figure

above.

Interface Configuration:

Transport Enabled

Type=checkbox

Default=checked

Check the box if this

Transport is enabled.

Interface Configuration:

Protocol

Type=string

Default=SIP

This is the protocol this

interface supports.

Interface Configuration:

Transport

Type=string

Range=TCP or UDP

Default=UDP

This is the transport for this

interface.

Interface Configuration:

Address

Type=string

This is the service IP

address label.

Range=1-50 characters

Default=Private_Service

_Address, Public_

Service_Address

Interface Configuration:

Port

Type=string

This is the port for this

interface. The range is 1024

to 65535.

Range=1024-65535

numbers

Default=5060

Interface Configuration:

Params

Type=string

These are the optional

Default=name=sipLscCo parameters for this interface.

nduit:interface=qfe0

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

Transport Management: Active-Hot Standby Server Heartbeat Mechanism

The Transport Manager (referred to as the reliability managed object)

is responsible for the heartbeat communication and state maintenance

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 address of every other 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 when this condition may not hold is when a new server is

being introduced into an existing group that is currently providing

service. In this case, the new server is provisioned with the information

for all the other servers while the other servers are not yet updated with

provisioning for the new server.

The reliability manager is configured by provisioning of engineering-

and network-related parameters. The engineering parameters

determine the timeout intervals and failure detection thresholds. The

NSD parameters define 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 within 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 address label in place of an explict address. This

capability simplifies the configuration process where the same data are

provisioned multiple times. For the Transport Manager configuration,

the Module Parameter label/value pairs are values associated 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_Static_Address and Private_Static_Address labels. 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 between the servers. Therefore, take care to configure 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-objects. The reliability manager internally launches network

services by communicating with other managed-objects 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.

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” application server that can exist 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 service instance) needs seven addresses

on the public network and seven addresses on the private network

(total for both servers).

Configuration of the NSD is what defines those SIP Application 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 active 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 service address tag values (their provisioned IP

addresses) should be different from the provisioned static addresses.

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Additional SIP TCF Base tab configuration information

This section contains additional configuration information for the IP and

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:

<transport>=<host address>:<host port>:<optional

parameters>

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 T a ble 26, “UDP/TCP/SSL

Config values.”

Table 26 UDP/TCP/SSL Config values

Parameter

Value

Description

Host Address

IP v4 address

This field contains the address of the

host on which you want the

connection to open.

Host Port

Optional

Integer

This field contains the port on which

to open the connection. The standard

port for SIP is 5050.

hostaddr:<IPv4>

The hostaddr value specifies the

primary public address of the SIP

Application Module.

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 general SIP server for TCP and UDP and starts

execution of two SIP server ports on network 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 networks.

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.

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 Off parameter

When Retransmission Off is false, the SIP server follows the

retransmission policies identified by the RFC 2543 (see note for

specific reference) specification for SIP.

Note: J. Rosenberg et al, SIP: Session Initiation Protocol, Internet

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 Invite transaction can remain open before it is forced

closed. A SIP Invite transaction, having received a provisional

response and waiting on a final response, is allowed to persist only as

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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.

Time Transaction parameter

When the Time Transaction value is true, all transactions are timed.

Invite transactions are forced closed and the Invite Timer duration has

expired. For all other transactions, the time-out duration is fixed at

64000 mS.

Add Defaults parameter

When the Add Defaults value is set to true, SDP message bodies in SIP

messages with missing mandatory SDP headers are regenerated with

default mandatory headers. This occurs when messages are proxied

through the SIP Application Module.

Note 1: Set this parameter to true when downstream servers fully

support the SDP specification.

Note 2: Set this parameter to false when downstream servers do not

fully support the SDP specification.

Check Mandatory Headers parameter

When the Check Mandatory Headers value is true, SDP messages are

screened for required header content.

Note: Missing headers cause message rejection.

When the Check Mandatory Headers 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 all 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 Application Module does not do any accounting management.

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 Application Module. 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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Security and Administration

How this chapter is organized

This chapter is organized as follows:

• “Security” on page 97

• “OAM&P strategy” on page 97

Security

The SIP Application Module with Back-to-Back User Agent functionality

controls the Media Portal (Media NAT) over an MGCP-type protocol.

The SIP Application Module ensures security of clients and the network

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 maintain connection to clients through NAT and/or firewall by

the keep-alive mechanism.

• Provides client authentication.

• 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 interface is not available from the public interface.

OAM&P strategy

The Management Module performs the security and administrative

functions for the SIP Application Module. 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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Appendix A: Basic call flows

Using SIP as the signaling protocol to establish a communication path

between endpoints, the SIP Application Module provides the following

call services:

• Voice plus video

• Call transfer

• Authentication

The following sections provide sample diagrams and descriptions of the

call flows that enable 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-client voice-plus-video call. Each client in the 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 follows this diagram.

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Figure 1 Client-to-client voice plus video diagram

Bearer path

Signaling path

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Appendix A: Basic call flows 101

Figure 2 Client-to-client voice plus video call flow

User B:

User A:

UserB@site1.domain1

Private:44-41112

UserA@site1.domain1

Private:44-41111

(site code+station code)

Pubic: 972-208-1234

(site code+station code)

Pubic: 972-684-1111

SIP

Application

Module

Client A

Client B

1. Invite sip:userB@domain1

To: userB@domain1

From: userA@domain1

SDP A

1. Invite sip:userB@domain1

To: userB@domain1

From: userA@domain1

SDP A

2. 180 SDP B

3. 200 OK SDP B

4. Ack

Call established between A and B (voice and video)

5. Bye

6. 200 OK

The following steps provide more detail about the call flow:

1. Client A sends an Invite to Client B.

2. Client B responds with 180 SDP (Session Description Protocol)

3. Client B responds with 200 OK.

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4. Client A sends an ACK message to the SIP Application module,

which sends the ACK on to Client B.

Note: The terminating client starts sending packets. The

connection is established.

5. Client B sends Bye to end the call.

6. Client A responds with a 200 OK.

Call transfer

Figure 3, “Call transfer to client diagram,”and Figure 4, “Call transfer

(blind) to client call flow,” show the basic call flow for a call transfer. A

detailed, step-by-step illustration follows these diagrams.

Figure 3 Call transfer to client diagram

Client A

Client C

Client B

Bearer path

Signaling path

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Appendix A: Basic call flows 103

Figure 4 Call transfer (blind) to client call flow

SIP

Application

Module

Call established between A and B

Client C

Client A

Client B

1. Invite (hold)

2. Invite (hold)

3. 200 OK

4. 200 OK

5. Ack

6. Ack

7. Refer (to C)

8. Refer (to C)

9. 202 (accepted)

10. 202 (accepted)

11. Invite (to C)

12. Invite (to C)

13. 200 OK

14. 180 Ringing

15. 200 OK

16. Ack

17. Set up

18. Ack

19. Notify (200 OK)

20. Notify (200 OK)

21. 200 OK

22. 200 OK

Call established between A and C

23. Bye

24. Bye

25. 200 OK

26. 200 OK

Call established between B and C

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The following steps provide more detail about the call flow:

1. Client A initiates the transfer of B to C. A selects blind transfer.

2. The SIP Application Module sends a Hold to B.

3. Client B sends a 200 OK message back to the SIP Application

Module. The 200 OK messages contain an Allow Header, which

lists the SIP methods that the client being transferred supports. If

Refer is in that list, then Refer is used for the transfer; otherwise,

Bye-Also is used.

4. SIP Application Module sends a 200 OK message back to Client A.

5. Ack message from Client A to the SIP Application Module.

6. Ack message from the SIP Application Module to Client B.

7. Client A sends Refer-to header with C’s information in it and a

Referred-by header with A’s information in it.

8. The SIP Application Module sends the Refer message to B.

9. 200 Accepted

10.200 Accepted

11. The SIP Application Module sends an Invite to B to establish the

new call between B and C.

12.Invite to C.

13.200 OK

14.Ringing (SIP) – SIP/2.0 180 Ringing (SIP clients do not send SDP

in the 180)

15.200 OK

16.Ack

17.New media connection is set up between B and C.

18.Ack

19.Client B notifies the SIP Application Module.

20.The SIP Application Module notifies Client A.

21.Client A sends a 200 OK.

22.SIP Application Module sends a 200 OK to Client B.

23.Client A hangs up.

Figure 5, “Failed call transfer call flow,”and the steps following the

figure show the call flow for a failed transfer.

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Appendix A: Basic call flows 105

Figure 5 Failed call transfer call flow

SIP

Application

Module

Client A

Client B

Client C

Call established between A and B

1. Invite (hold)

2. Invite (hold)

3. 200 OK

4. 200 OK

5. Ack

6. Ack

7. Refer (to C)

8. Refer (to C)

9. 202 (accepted)

11. Invite (to C)

10. 202 (accepted)

12. Invite (to C)

13. 486 Busy Here

14. 486 Busy Here

15. Ack

16. Ack

17. Notify (486 Busy Here)

18. Notify (486 Busy Here)

19. 200 OK

20. 200 OK

21. Invite (Unhold) or Bye

22. Invite (Unhold)

23. 200 OK

24. 200 OK

25. Ack

26. Ack

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The following steps provide more detail about the call flow:

1. Client A initiates the transfer of B to C. A selects blind transfer.

2. The SIP Application Module sends a Hold to B.

3. Client B sends a 200 OK message back to the SIP Application

Module. The 200 OK messages contain an Allow Header, which

lists the SIP methods that the client being transferred supports. If

Refer is in that list, then Refer is used for the transfer; otherwise,

Bye-Also is used.

4. SIP Application Module sends a 200 OK message back to Client A.

5. Ack message from Client A to the SIP Application Module.

6. Ack message from the SIP Application Module to Client B.

7. Client A sends Refer-to header with C’s information in it and a

Referred-by header with A’s information in it.

8. The SIP Application Module sends the Refer message to B.

9. 200 Accepted

10.200 Accepted

11. The SIP Application Module sends an Invite to B to establish the

new call between B and C.

12.Invite to C.

13.SIP Application Module receives a 486 Busy Here response from

Client C. This response could be any type of 4xx, 5xx, or 6xx error

message.

14.The SIP Application Module sends a Notify message to Client B.

The body of the Notify message contains the 486 Busy Here, in this

example

15.Client B responds with an Ack.

16.The SIP Application Module sends an Ack to Client C.

17.Client B Notifies Client A, through the SIP Application Module, that

Client C is Busy.

18.Notify goes to Client A.

19.Client A responds with a 200 OK.

20.200 OK to Client B.

21.There are now two possibilities:

• Invite (Unhold): Client A can re-establish the call to Client B, in

which case steps 22-26 apply.

• Bye: Client A can hang up, in which case this is the last step.

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Appendix A: Basic call flows 107

22.SIP Application Module sends an Invite (Unhold) to Client B.

23.Client B responds with a 200 OK.

24.The SIP Application Module sends the Invite to Client A.

25.Client A responds with an Ack.

26.The SIP Application Module sends an Ack to Client B and the call

is re-established.

Authentication

Figure 6, “Authentication diagram,”and Figure 7, “Authentication call

flow,” show the basic call flow for Authentication. You can configure the

SIP Application Module to support Authentication on Invite or

Authentication on Registration. Following these diagrams is a detailed,

step-by-step example of Authentication on Registration.

Figure 6 Authentication diagram

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Figure 7 Authentication call flow

SIP

Application

Module

Client A

Database

1. Register

2. 100 Trying

3. Retrieve user profile

4. 401 Unauthorized

5. Register

6. 100 Trying

7. Add user

8. User registered

9. 200 Registration successful

The following steps provide more detail about the call flow:

1. Client A sends a Register (SIP) message to the SIP Application

Module.

Initial requests never contain the user's credentials (basically, the

initial request just contains a password). Client A makes the

request, the SIP Application Server rejects it and gives them a

piece of information called a nonce in the 401 Unauthorized

message. The client takes that nonce and uses it to encrypt their

password information and sends this back in the second request.

2. The SIP Application Module returns a 100 Trying message to Client

A, then

3. The SIP Application Module attempts to retrieve the FROM party's

subscriber information to see if they've been marked as INACTIVE

in the system. This also causes the information to be cached at the

SIP Application Server, so the same dip is not made to the

database on the subsequent registration attempt. This profile

information allows the system to determine what their password is

in order to authenticate them.

4. In this case, the Database Module has returned a 401

Unauthorized message to the SIP Application Module, which sends

the information on to Client A.

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Appendix A: Basic call flows 109

5. Client A sends another Register (SIP) message to the SIP

Application Module.

6. Again, the SIP Application Module returns a 100 Trying message to

Client A.

7. The SIP Application Module tells the Database Module to add this

user to the registration tables (SQL).

8. The Database Module tells the SIP Application Module that the

user is registered (SQL). Nothing is returned unless there is an

error. If there is no error code, the registration worked.

9. The SIP Application Module then contacts Client A with a

successful registration message (200 Registration Successful).

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Succession Multimedia Communications Portfolio

MCP SIP Application Module

Basics

NORTEL NETWORKS CONFIDENTIAL: The information contained in this document is the

property of Nortel Networks. Except as specifically authorized in writing by Nortel Networks, the holder of

this document shall keep the information contained herein confidential and shall protect same in whole or in

part from disclosure and dissemination to third parties and use same for evaluation, operation, and mainte-

nance purposes only. Changes or modifications to the MCP SIP Application Module without the express

consent of Nortel Networks may void its warranty and void the user’s authority to operate the equipment.

Information is subject to change without notice. Nortel Networks reserves the right to make changes in

design or components as progress in engineering and manufacturing may warrant.

*Nortel Networks, the Nortel Networks logo, the Globemark, UNIStim, MCP, Oracle, Nortel, Northern Tele-

com, and NT, are trademarks of Nortel Networks.

Publication number: NN10029-111

Product release: MCP 1.1 FP1 Standard

Document release: Standard MCP 1.1 FP1 (02.02)

Date: April 2003

Printed in the United States of America.

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