Nortel Networks Telephone NN10033 111 User Manual

NN10033-111

Succession Multimedia Communications Portfolio

MCP Interworking

Basics

Standard MCP 1.1 FP1 (02.02) April 2003

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Overview

How is this chapter organized

This chapter provides a high-level overview of the interworking

between the Multimedia Communications Platform (MCP) and the

following systems:

• Primary Rate Interface (PRI)-enabled switches

• SIP-T-based switches

• third-party gateways

• traditional phones

• third-party voicemail servers

Interworking with PRI-enabled switches

The MCP uses the SIP PRI Gateway to perform interworking with

PRI-enabled switches (switches with PRI interfaces).

For more detailed information about the SIP PRI Gateway, please refer

to the MCP SIP PRI Gateway Basics document.

Functional description

The SIP PRI Gateway is a signaling and media gateway that

interconnects a SIP-based Voice over IP (VoIP) domain and a system

that uses ISDN Primary Rate Interface (PRI/Q.931).

A PRI-enabled system can be a switch that works on the

Public-Switched Telephone Network (PSTN) or Private Branch

Exchange (PBX). The Nortel Networks DMS-100 and the Nortel

Networks Meridian SL-100 are examples of PRI-enabled switches.

The SIP PRI Gateway’s primary function is to convert the packet-based

voice streams of the VoIP system to circuit-based voice streams of the

PRI-enabled system.

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Supported services

The SIP PRI Gateway supports the following call types:

• SIP to PRI

• PRI to SIP

• PRI to PRI

The SIP PRI Gateway supports the following call features:

• Basic call: The ability to make a call.

• Hold/Retrieve: The ability to hold and retrieve calls.

• Call transfers: The ability to forward the call to a third party after the

call is established. Call transfer is limited to SIP clients. Callers from

the PRI-enabled switches cannot perform this function.

• Call redirect: The ability to forward a call before it is answered.

• Codec negotiation: The ability to negotiate between different VoIP

codecs during call setup, mid-call, call transfer, and call retrieve.

The SIP PRI Gateway supports the following codecs:

— G.711 mu-law (PCMU)

— G.711 a-law (PCMA)

— G.723.1

— G.729a

• Call rejection: The ability to reject a call on nodal authentication

request.

• Calling party name and number: The MCP supports the

delivery/reception of calling party name and number information

to/from PRI-enabled switches. Incoming calling party information

privacy indication is honored by the MCP. For MCP call originations,

privacy indication is not used.

• Dual-Tone Multi Frequency (DTMF) outpulsing: The ability to

outpulse to PRI-enabled switches. However, DTMF detection is not

provided. Military DTMF digits A-D are currently not supported.

• ISDN trunk group selection: The selection is based on information

provided in a request Universal Resource Identifier (URI).

• Ringback: The SIP PRI Gateway provides ringback towards the

circuit-switched side of the network.

• PRI variant support: The SIP PRI Gateway supports different PRI

variants. For a list of the PRI variants that the SIP PRI Gateway

supports, please refer to the MCP SIP PRI Gateway Basics

document.

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Overview 5

• SIP/PRI mapping support, including:

— Protocol parameter mapping

— error codes

— PRI cause values with SIP responses

— presentation and screening indicators

• PRACK: the provisional response acknowledge message that

ensures that the ringing signal does not get lost.

• Type of Server (ToS): The ToS bit can be configured to indicate the

priority of the voice packet over data packet to ensure quality of

service.

• Long-call service: The ability to detect abandoned calls.

More detailed information about the functional capabilities of the SIP

PRI Gateway is available in the MCP SIP PRI Gateway Basics

document.

Interworking with CS 2000

Even though Nortel Networks Communication Server 2000 (CS 2000)

is a VoIP switch, it does not fully support the Session Initiation Protocol

(SIP). The following description provides information on how the MCP

interworks with the CS 2000.

The MCP does not support VoIP over Asynchronous Transfer Mode

(ATM) or pure ATM, as the MCP network nodes are IP-based and not

ATM based.

Functional description

The interworking between an MCP and CS 2000 uses Session Initiation

Protocol for Telephones (SIP-T) over User Datagram Protocol (UDP) to

transport ISDN User Part (ISUP), the call control part of the Signaling

System 7 (SS7) protocol.

SIP-T is an extension of the Session Initiation Protocol (SIP) that allows

SIP to be used to facilitate the interconnection of the Public Switched

Telephone Network (PSTN) with packet networks. SIP-T encapsulates

the ISDN User Part (ISUP) messages in the SIP messages and

translates ISUP information into the SIP header for routing purposes.

Although the MCP SIP Application Module supports receiving of

encapsulated ISUP messaging (using SIP-T), it does not send

encapsulated ISUP back out. For more information about the

capabilities of the SIP Application Module, please refer to the SIP

Application Module Basics document.

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Figure 1 shows a network view of the MCP/CS 2000 interconnection.

The MCP configuration enables direct connection between the MCP

and CS 2000 with no intervening SIP proxies. The CS 2000 and its

associated media gateways are placed in the private managed

network.

Figure 1 MCP/CS 2000 Network view

PSTN

network 1

network 2

SIP-T

SIP

SIP-T

SIP

MCP

network 1

MCP

network 2

Supported services

This section provides information on services that can interwork

between the two platforms.

VPN Dialing

A Virtual Private Network (VPN) dial plan enables an enterprise to have

one common dial plan across different geographic locations without

incurring long-distance expenses.

The MCP system uses domains to manage its call routing, while the CS

2000 system uses customer groups.

A profile is used to facilitate call routing between the two systems. The

header named “x-nortel-profile” is used to identify the CS 2000 profile.

The profile is used to map the domain to a PSTN customer group in the

PRODOMAIN table in the Database Module.

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Overview 7

The profile is added when a new domain is created in the MCP system.

Note: The profile name in the TELEPROF table on the CS 2000 side

must match the profile name in the PRODOMAIN table on the MCP

side. Otherwise, interworking between the MCP and CS 2000 will not

work.

Call Forward

Both the MCP and the CS 2000 support call forwarding. No extra

interworking considerations are necessary as call forwarding does not

interact across the MCP and CS 2000 domains, except to deliver a call

between the two domains.

Call Transfer

Call transfer works between the two platforms. The calls are transferred

because from each platforms view the other is just initiating a new call.

Ad Hoc Conference calls

There is no change in the conference application between the two

platforms. When an ad hoc conference call is made, the client

application that starts the conference controls where the conference

bridge is allocated. For example, if the conference call is made from a

client application on the MCP side, the conference bridge will be

allocated on the MCP.

Media Negotiation

The MCP provides media negotiation for the CS 2000 since the CS

2000 gateway controller is not capable of providing this function in the

current release.

The MCP requires a list of all commonly supported codecs across the

CS 2000 media gateways to be provisioned in the cs2k.xml file. The list

is required because the MCP does not determine which gateway the

CS 2000 will use.

Although the MCP also allows for the provisioning of packet times that

the CS 2000 will use, this is not recommended. By provisioning specific

packet times, the SDP packet time will no longer pass transparently

through the MCP. This means that the packet time negotiation that

would normally occur at the client level is now being handled at the

server level, and this could lead to voice connection issues.

The SDP sent by the CS 2000 media gateways is only guaranteed to

contain the v=, c=, m=, and a= SDP headers in the SDP message. The

gateways can accept reception of other SDP parameters, so no

screening needs to be done on the SDP that the MCP sends to the CS

2000. The only exception is the screening of the m= lines to only

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include the audio codecs supported by the gateways. For the SDP that

is received from the CS 2000, the MCP fills in the missing mandatory

parameters as specified in RFC 2327. Details on the restrictions on

SDP are specified in the SN03 IP Gateway InterOP Requirements

document.

Recursive Search

The MCP Back-to-Back User Agent (BBUA) handles SIP 302

responses to an INVITE sent out because of a received INVITE from

the CS 2000. The BBUA sends INVITEs for each contact in the SIP 302

response. The SIP 302 response will not be passed back to the CS

2000. The INVITES for each contact can be sent parallel or sequentially

depending on the mode in which the BBUA processed the initial

INVITE, which then resulted in the SIP 302 response.

Hold

SIP implements a Hold as a re-INVITE with the connection information

in the SDP set to 0.0.0.0. In this release, Hold cannot be handled by all

the CS 2000 media gateways. The MCP shields the CS 2000 from

seeing Hold re-INVITE requests. The MCP, when equipped with an

RTP Media Portal, can manage the RTP connections without affecting

the connection to the CS 2000.

Retrieve

Retrieve is also implemented in SIP through a re-invite. To retrieve a

party on hold the new invite contains valid SDP. The new SDP is used

to restore the media connection between the two clients.

Long Call Audit Timers

In order to prevent hung calls between the two platforms, a long call

audit timer is implemented between the two platforms. Both platforms

use the INFO ping capability described in the SIP INFO message

RFC2976. This involves sending an INFO message with no message

body. Upon receipt of this message, a client should send a 200 OK if

the call exists. If the response to the INFO is any of the following

response codes, then the platform sending the message assumes that

the call no longer exists and frees all resources associated with the call:

• 404 Not Found

• 408 Request Timeout

• 410 Gone

• 480 Temporarily Unavailable

• 481 Transaction Does Not Exist

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Overview 9

If the platform sending the INFO receives a response code other than

the ones listed above, it treats the response as a valid audit response,

and does not bring the call down.

The CS 2000 sends a 200 OK for an empty INFO message and a 481

response code if the call leg does not exist.

Supported media

The following are the media supported by the interworking of MCP and

the CS 2000:

• G. 711 PCMU

• G. 711 PCMA

• G. 729

• G. 723

• G3 Fax

• Modem

Note 1: This list represents a full list of the media capabilities the

MCP can use while interworking with the CS 2000. The actual list of

supported media types depends on the capabilities of the gateways

that the CS 2000 is using.

Note 2: G3 Fax and Modem media calls are used only for CS 2000

gateway calls as there are no MCP endpoints that support either of

these two media types.

QoS

MCP implements its Quality of Service through the DiffServ

(Differentiated Service) feature. The parameters of MCP QoS include

the following:

• QoS DiffServ code for signaling: specifies signaling quality for SIP

clients.

• QoS DiffServ code for audio: specifies audio quality for SIP clients.

• QoS DiffServ code for video: specifies video quality for SIP clients.

• QoS 802.1p for service priority: specifies service priority for SIP

clients.

Interworking with third-party gateways

The MCP can use SIP signaling between the MCP network and a

third-party SIP-enabled gateway. The third-party gateway provides the

necessary signalling interworking between the MCP network and the

other network to which the gateway is connected. For example, a

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line-based voicemail server requires that a Line Gateway be placed

between the MCP network and the voicemail server. This Line Gateway

would process SIP messages to/from the MCP network and create the

corresponding line signalling to/from the line network. “Interworking

with third-party voicemail servers” on page 13, provides more

information about how the MCP interworks with non-SIP aware

voicemail servers.

Interworking with traditional phones

The MCP provides Converged Desktop Services (CDS) to facilitate the

interworking with traditional phones of the TDM network. This allows

users to have a personal computer (PC) provide the multimedia portion

of their communication session while having the traditional telephony

system provide the voice portion of their communication session.

Functional description

A user’s Converged Desktop consists of a regular TDM telephone, and

a SIP Multimedia PC Client (PC Client) software provisioned as a

Converged PC Client. Figure 2 shows how the Converged PC Client

interconnects with the network. The Converged PC Client provides an

enhanced communication experience to the user, while the TDM

telephone works exactly as it does today.

Figure 2 Converged Desktop Services Network Diagram

Existing

switching system

MCP network

PRI

Various

line

protocols

SIP

Converged Desktop

Traditional

phone

Converged

PC Client

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Overview 11

Converged Desktop Services

CDS enhances the end user’s communication experience in a variety

of ways:

• Advanced Call Handling: The user can use the MCP Personal

Agent web pages to control the user’s availability. By providing this

ability to CDS users, features not easily accessible on existing TDM

switching systems now become viable. For example, a user can

activate MCP-based forking (using the Personal Agent web pages)

so that when the user’s desktop telephone is called, the users cell

phone rings as well. Once one leg of the forked call is answered, the

other leg stops ringing.

• Inbound call log: Allows the user to see who has called and when

the call occurred.

• Video calling line identification: Allows the user to see who is calling.

The picture is retrieved from the network-based address book

accessible on the Converged PC Client.

• Redirection of incoming calls at the Converged PC Client: Upon the

arrival of an incoming call, the user may click on the “Redirect”

button, and send the incoming call to another address. Once the

user answers the call (using the TDM phone), the redirect function

is no longer available.

• File transfer: If both the originator and terminator support the MCP

file transfer collaboration application, then files can be transferred

back and forth between the two users. The PC Client (both

Converged and non-Converged) is the only endpoint that supports

this functionality.

• Whiteboard sharing: If both the originator and terminator support the

MCP whiteboard collaboration application, then a whiteboard

session can be set up between the two users. The PC Client (both

Converged and non-Converged) is the only endpoint that supports

this functionality.

• Clipboard transfer: If both the originator and terminator support the

MCP clipboard transfer collaboration application, then the Windows

System Clipboard may be transferred between the two users. The

clipboard transfer application allows a user to “Copy (CTRL-C)”

items such as PowerPoint slides or sections of Excel spreadsheets

to the clipboard, and then sends them to the other party. The other

party then “Pastes (CTRL-V)” the items. The PC Client (both

Converged and non-Converged) is the only endpoint that supports

this functionality.

• Web Co-browsing: If both the originator and terminator are capable

of this functionality, then one user can automatically drive the

other’s web browser. The PC Client (both Converged and

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non-Converged) is the only endpoint that supports this functionality.

The Web Client supports the reception of web pages, but cannot

send web pages to a Converged PC Client.

• Instant Messaging (IM): The Converged PC Client can send and

receive messages from any client that supports the Nortel Networks

IM format. All MCP clients support sending and receiving of instant

messages with each other.

• Presence state indications: The Converged PC Client allows the

user to select a presence state in the MCP network. The Converged

PC Client also allows the user to see the presence states for the

Buddies defined in the user’s network-based address book.

Configuration Requirements

PRI is used as the interface between the existing TDM switching

system and the MCP. The MCP supports the following PRI protocol

variants:

• AT&T 4ESS(AT4)

• AT&T 5ESS (E10)

• AT&T TR 41459

• Bellcore National 2

• ETSI

• ECMA-143

• ETS 300 102-1

• Northern Telecom DMS-100 (DMS)

• NIS A211-1

• QSig

For more detailed information about the which PRI variants the SIP PRI

Gateway supports, please refer to the MCP SIP PRI Gateway Basics

document.

Successful interworking between the MCP and the TDM switch

requires that the TDM switch activate the Simultaneous Ring (SimRing)

feature and assign each user a SimRing number. A user in the TDM

switch that has acquired this SimRing number can be a CDS user. The

SimRing feature must send “SimRing” calls to a routable and unique

number for each CDS user.

The MCP system operator must provision a user as a CDS user. A CDS

user cannot use the SIP Multimedia PC Client for voice. The CDS

user’s TDM phone is used for voice. In addition, the CDS user may use

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Overview 13

other SIP endpoints (such as the SIP Multimedia Web Client or an

i2004 controlled by the IP Client Manager (IPCM) for voice over IP calls.

An MCP alias must be set up for each user so that the alias is the same

as the Calling Line ID sent from the TDM switch to the MCP over PRI.

When a non-CDS MCP user calls a CDS user, the call is sent out the

gateway to the CDS user’s TDM phone and the non-CDS user’s

public/private charge ID is used to identify them to the TDM switch as

the calling party. This charge ID is sent to the Converged PC Client, on

the SimRing leg of the call, and is used by the Converged PC Client to

contact the calling party’s MCP client. Therefore, the non-CDS user’s

charge ID must be included as an alias in the non-CDS user’s

provisioning. A non-CDS user’s charge ID cannot be shared amongst

users within a domain because the charge ID must be included as an

alias and a user's aliases must be unique within a domain.

Calls to a CDS user must terminate to the existing switching system of

the CDS user before the call is routed to the MCP. For example, the

originator’s existing switching system must route calls using the

existing systems, as opposed to sending the call to the MCP. This is

required since all calls from the SIP PRI gateway are implied to have

been triggered by the SimRing feature on the existing switching

system.

Interworking with third-party voicemail servers

There are three major types of third-party voicemail servers. The

following sections describe how the MCP can interwork with the

following types of third-party voicemail servers:

• SIP-based voicemail servers

• Trunk-based voicemail servers

• Line-based voicemail servers

SIP-based voicemail servers

SIP-based voicemail servers are SIP-enabled and can interwork

directly with the MCP network. SIP is used to set up connections

between the client and the voicemail server. The RTP Media Portal is

used to carry the media packets between the client and the voicemail

server.

Figure 3 shows how the MCP interconnects with a SIP-aware

third-party voicemail server.

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Figure 3 SIP-based Voicemail Server

MCP network

SIP

UNIStim

MGCP+

RTP

For more information on how the SIP Application Module uses Media

Gateway Control Protocol Plus (MGCP+) to control the RTP Media

Portal, please refer to the MCP Media Portal Basics document.

For more information on how the IP Client Manager uses Unified

Networks IP Stimulus (UNIStim) to control the i2004 Internet

Telephone, please refer to the MCP IP Client Manager Basics

document.

Trunk-based voicemail servers

Trunk-based voicemail servers cannot directly communicate with the

MCP network. A SIP PRI Gateway is required for the MCP to interwork

with a trunk-based voicemail server. The SIP PRI Gateway also

provides a media path from the IP network to the voicemail server on

the PSTN.

A terminal server, using a Simplified Message Desktop Interface

(SMDI), sends data regarding the storage and retrieval of voicemail

from the trunk-based voicemail server to the MCP network. The MCP

network does not send data back to the voicemail server over the SMDI

link. Figure 4 shows a network view of the MCP interconnection with a

trunk-based third-party voicemail server.

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Overview 15

Figure 4 Trunk-based Voicemail Server

IP network

SMDI

MCP network

SIP

SMDI

PRI

RTP

MGCP+

RTP

UNIStim

Line-based voicemail servers

Line-based voicemail servers cannot directly communicate with the

MCP network. A Line Gateway, also known as an analog station

gateway, is required for the MCP to interwork with a line-based

voicemail server.

Similar to interworking with trunk-based voicemail servers, an SMDI

terminal server is to exchange data between the MCP and the legacy

voicemail server. However, the SMDI links are used to both send and

receive data regarding the storage and retrieval of voicemail from the

line-based voicemail server to the MCP network. Figure 5 shows a

network view of the MCP interconnection with a line-based third-party

voicemail server.

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Figure 5 Line-based Voicemail Server

IP network

SMDI

MCP network

SIP

SMDI

RTP

MGCP+

RTP

UNIStim

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Upgrades

How this chapter is organized

This chapter provides information about upgrade procedures dealing

with MCP interworking with the following other systems:

• Primary Rate Interface (PRI)-enabled switches

• SIP-T-based switches

• third-party gateways

• traditional phones

• third-party voicemail servers

Interworking with PRI-enabled switches

MCP interworking with PRI-enabled switches does not involve

additional software deployment to the PRI-enabled switches. The MCP

SIP PRI Gateway uses standard PRI protocols and is compatible with

any PRI-enabled switch that understands these standard PRI

protocols:

• AT&T 4ESS(AT4)

• AT&T 5ESS (E10)

• AT&T TR 41459

• Bellcore National 2

• ETSI

• ECMA-143

• ETS 300 102-1

• Northern Telecom DMS-100 (DMS)

• NIS A211-1

• QSig

Because SIP revisions are backwards compatible, the MCP SIP

Application Module and the MCP SIP-PRI Gateway can be upgraded

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independently of one another. Refer to MCP SIP PRI Gateway Basics

and MCP SIP Application Module Basics for more information on

upgrades to those nodes of the MCP network.

Interworking with CS 2000

MCP interworking with the CS 2000 does not involve additional

software deployment. The functionality exists in the SIP Application

Module. Refer to the MCP SIP Application Module Basics for more

information on upgrades.

Note: Since the MCP and CS 2000 can be upgraded at different

times, the two products are backwards compatible by one release.

Interworking with third-party gateways

MCP interworking with SIP-enabled third-party gateways does not

involve additional software deployment to the third-party gateways. The

MCP SIP Application Module uses SIP to successfully communicate

with a third-party gateway that also speaks SIP. Each third-party

gateway software release must be validated and certified against the

current MCP release.

The MCP SIP Application Module and third-party gateways can be

upgraded independently of one another. Refer to MCP SIP Application

Module Basics for more information on upgrades to the MCP SIP

Application Module.

For information about the upgrade procedures of a specific third-party

vendor’s gateway, please refer to the documentation provided by that

third-party vendor.

Interworking with Traditional Phones

MCP interworking with traditional telephones does not involve

additional software deployment to the existing switching system.

However, the SimRing feature on the TDM switch must be activated in

order to route calls to CDS users.

The MCP network and SIP Multimedia PC Client (PC Client) may not

be upgraded at the same time. As new CDS functionality is introduced

in the network through MCP network node upgrades, the existing PC

Clients must continue to interwork with the network. Therefore, the

MCP network nodes must be backwards compatible with older PC

Clients and MCP nodes. In addition, PC Clients (both Converged and

non-Converged) must be backwards compatible with all previously

released PC Clients, as different versions of the clients may co-exist in

a given MCP network.

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Upgrades 19

Compatibility is maintained by version identifiers included in an MCP

user’s service package information. When new CDS functionality

becomes available in the MCP network, older Converged PC Clients

can continue to exist and operate in the upgraded MCP network;

however, they won’t be able to access the new Converged PC Client

services.

Note: In general, MCP network is upgraded before the PC Client is

upgraded.

Interworking with third-party voicemail servers

MCP interworking with third-party voicemail servers does not involve

additional software deployment to the third-party voicemail servers.

The MCP SIP Application Module uses SIP to successfully

communicate with third-party voicemail servers. For non-SIP-aware

voicemail servers, a gateway between the MCP network and the legacy

voicemail server is used, in which case the MCP SIP Application

Module still relies on SIP to successfully communicate with that

gateway (for example, the MCP SIP PRI Gateway connects the MCP

network to a PRI trunk-based voicemail server).

The MCP SIP Application Module and the MCP SIP PRI Gateway can

be upgraded independently of any third-party voicemail server (or

necessary intermediary gateway). Refer to MCP SIP PRI Gateway

Basics and MCP SIP Application Module Basics for more information

on upgrades to those nodes of the MCP network.

For information about the upgrade procedures of a specific third-party

vendor’s voicemail server, please refer to the documentation provided

by that third-party vendor.

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Fault management

For more information about fault management on a specific MCP

network node, please refer to the Fault Management chapter in the

corresponding documents:

• MCP SIP Application Module Basics

• MCP SIP PRI Gateway Basics

For information about CS 2000 fault management functionality, refer to

the CS 2000 Fault Management document.

For information about fault management functionality on any third-party

gateway or voicemail server, please refer to the documentation

provided by that third-party vendor.

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Configuration Management

How this chapter is organized

This chapter provides information about the tasks required to configure

the MCP to allow interworking with the following other systems:

• PRI-enabled switches

• SIP-T-based switches

• third-party gateways

• traditional phones

• third-party voicemail servers

Unless stated otherwise, all tasks are described from the MCP

perspective.

Configuring the MCP SIP PRI gateway

For more information about configuring the MCP SIP PRI Gateway,

please refer to the Configuration Management chapter of the MCP SIP

PRI Gateway Basics and MCP SIP Application Server Basics

documents.

Configuring CS 2000 interworking

This section describes the tasks required to configure the MCP for CS

2000 interworking. This section also describes the key settings required

for the CS 2000 interworking.

Configuration tasks

The configuration uses provisioning tasks that have been described in

detail in the MCP SIP Provisioning Client User Guide. This section

focuses on tasks related to configuring a domain for CS 2000

interworking.

The table below shows the provisioning tasks required depending on

whether a domain exists or not. This section provides procedures for

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tasks required in both scenarios. For other tasks, refer to the MCP SIP

Provisioning Client User Guide.

Table 1 Configuration Tasks for CS 2000 Interworking

Tasks

Domain Exists Domain Does Not Exist

Yes

Add Domain

List Domain (for modification)

Assign Services

Create Service Package

Yes

Add gateway, gateway route, trunk group Yes

Set Class of Service

Add Telephony Routes

Add Route Lists

Set Profile

Yes

All the tasks are performed by an administrator through the SIP

Provisioning Client.

The administrator requires the following access:

• Domain Management

• User Management

• IPCM Provisioning

• Gateway Routes

• Device Management

• Telephony Routes

• Service Package Creation

Procedure 1 Access the SIP Provisioning Client

From a web browser,

Enter the Uniform Resource Locator (URL) of the SIP

Provisioning Client.

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Enter your user name and password.

Click LOG IN NOW.

Gateway configuration

The MCP views the CS 2000 as a type of gateway. To complete calls

between the MCP and the CS 2000, gateways, gateway routes, and

virtual trunk groups need to be configured.

Note: A domain must be configured for CS 2000 interworking before

configuring the gateway.

Procedure 2 Configure a gateway for CS 2000 interworking

From the SIP Provisioning Client,

Click Gateways after you log in.

Click Add Gateway. The system displays Create new gateway

window.

Enter the information for a CS 2000 gateway host. The CS 2000

gateway host requires the following information:

• CS 2000 host name: The name of the server.

• Port: The port number is 5060.

• maddr: The IP address of the CS 2000’s Virtual Router

Distribution Node (VRDN).

• Transport protocol: Identify UDP as the transport protocol.

Example

An example of the gateway host information is shown as

follows:

SIPSERVER:5060;maddr=10.10.10.10;transport=udp

Enter cs2kin the Gateway Type field.

Click Save to conclude the procedure.

Procedure 3 Configure a gateway route for CS 2000 interworking

From the SIP Provisioning Client,

Click Gateways after you log in.

Click Add route. The system displays Create new gateway route

window.

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Enter a description of the new route in the Description box, for

example, CS 2000.

Click the Domain pull-down list and select a domain that is used

for CS 2000 interworking to associate with the route.

Click Save to complete the procedure.

For more information about using the SIP Provisioning Client and

configuring gateways, please refer to the MCP SIP Provisioning Client

User Guide and the MCP SIP PRI Gateway Basics documents.

Procedure 4 Configure a trunk group for CS 2000 interworking

From the SIP Provisioning Client,

Click Gateways after you log in.

Click Add Trunk Group. The system displays the Create new

trunk group window.

Select a CS 2000 gateway from the Gateway pull-down list.

Select a CS 2000 gateway route from the Route pull-down list.

Enter a virtual trunk group name.

Click Save to complete the procedure.

For more information about using the SIP Provisioning Client and

configuring virtual trunk groups, please refer to the MCP SIP

Provisioning Client User Guide and the MCP SIP PRI Gateway Basics

documents.

Telephony routes configuration

Telephony routes configuration is required to set up Class of Service

(COS), telephony routes, and route lists for CS 2000 interworking.

Procedure 5 Configure Class of Service for CS 2000 interworking

From the SIP Provisioning Client,

Click Domain after you log in.

Select the interworking domain you want to configure.

Click Telephony Routes and then Routing COS.

Enter a name for the Class of Service in the Name box. The

name can be alphanumeric.

Enter a description for the COS in the Description box.

Click Save. The system displays the newly created COS in the

lower part of the window.

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Repeat Step 4-6 to create additional COS values you need.

Use the Up and Down buttons to reorder the COS in the Current

Choices Available box. The higher the order, the more services

and priorities the COS has.

For more information about using the SIP Provisioning Client and

configuring COS, please refer to the MCP SIP Provisioning Client User

Guide and the MCP SIP PRI Gateway Basics documents.

Procedure 6 Configure telephony routes for CS 2000

interworking

From the SIP Provisioning Client,

Click Domain after you log in.

Select the interworking domain you want to configure.

Click Telephony Routes and then Add Telephony Route. The

system displays the Create New Telephony Route window.

Enter the parameters of the telephony route.

Add the route to a route list.

Click Save.

Repeat Step 3-6 to add more routes.

Click List Telephony Routes to display the list of routes created

for CS 2000 interworking.

Click Change Parameters. The system displays the parameters

related to the gateway route, including the following:

Set the parameters to the appropriate values.

Click Save to complete the procedure. You can also click Clear

to remove all the existing values in the fields.

For more information about using the SIP Provisioning Client and

configuring telephony routes, please refer to the MCP SIP Provisioning

Client User Guide and the MCP SIP PRI Gateway Basics documents.

Procedure 7 Configure route lists for CS 2000 interworking

From the SIP Provisioning Client,

Click Domain after you log in.

Select the CS 2000 domain you want to configure.

Click Telephony Routes and then Add Route List.

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Click Save to complete the procedure. You can click Clear to

remove all the existing values.

For more information about using the SIP Provisioning Client and

configuring route lists, please refer to the MCP SIP Provisioning Client

User Guide and the MCP SIP PRI Gateway Basics documents.

Configure domain or sub-domain profile

The MCP maps the domains (and sub-domains) of the MCP network to

the route lists/translations of the CS 2000 side, using profile

information. The following procedure configures the header profile for

the domain (or sub-domain) so that MCP knows what domain (or

sub-domain) to use in order to communicate with a CS 2000.

If a sub-domain is used, then the sub-domain data will override the data

in the parent domain.

Procedure 8 Configure domain profile for CS 2000 interworking

From the SIP Provisioning Client,

Click Domain after you log in.

Select the domain you want to configure to talk to CS 2000.

Click Set Profile.

Enter a profile name in the Profile field. The profile name must

be the same as the profile name in table TELEPROF in the CS

2000.

Click Save to complete the procedure.

Note: The same steps can be used to configure a

sub-domain. After selecting a domain in step 2, click on

Sub-Domain and select Set Profile for the sub-domain.

For more information about using the SIP Provisioning Client and

configuring domain profiles, please refer to the MCP SIP Provisioning

Client User Guide and the MCP SIP PRI Gateway Basics documents.

Node name configuration

The SIP Application Module and CS 2000 use unique names to identify

themselves in the network. Node names must be alpha-numeric strings

and can not contain special characters like "_" or "-". Configuring

service node names is performed through the MCP System

Management Console. For more information on using the Sytem

Management Console, please refer to MCP System Management

Console Basics.

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When the SIP Application Module is deployed from the management

server, the service node name is defined in one of two places,

depending on whether or not the SIP Application Module is configured

in an N+M cluster.

Non-N+M node name configuration

In a non-N+M cluster configuration, the service node name is added to

the "Server Properties" tab in the System Management Console. In a

non-N+M configuration, the service node name can be the node name

of the SIP Application Module platform. For more information on

configuring the SIP Application Module service node name in a

non-N+M configuration, please refer to the Configuration chapter in the

MCP SIP Application Module Basics document.

Once the service node name information is configured in the “Server

Properties” tab of the System Management Console, this information

must then be datafilled on the CS 2000 (table MGCINV).

N+M node name configuration

In an N+M configuration, the service node name has to be assigned to

the service instance so, unlike the non-N+M configuration, the service

node name can not be the node name of the SIP Application Module

platform. 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. For more

information on configuring the SIP Application Module service node

name in a N+M configuration, please refer to the Configuration chapter

in the MCP SIP Application Module Basics document.

Once the service node name information is configured in the “Transport

Management” tab of the System Management Console, this

information must then be datafilled on the CS 2000 (table MGCINV).

CS 2000 node authorization

The SIP Application Module allows only authorized network nodes to

send a SIP request to it with out requiring the request to be

authenticated. The SIP PRI Gateway is an example of an authorized

network node; the SIP Application Module does not challenge incoming

call requests from the SIP PRI Gateway. The CS 2000 node must be

added to list of authorized nodes so that it can send SIP-T messaging

to the SIP Application Module. This is achieved by adding the IP

address of the CS 2000 to the Authorized SIP Nodes field in the

“Authentication” tab of the System Manage Console. For more

information on configuring the SIP Application Module service node

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please refer to the Configuration chapter in the MCP SIP Application

Module Basics document.

Configuring third-party gateway interworking

The provisioning tasks required for the MCP to interwork with

third-party gateways are described in detail in the Gateways chapter of

the MCP SIP Provisioning Client User Guide document.

Configuring traditional phone interworking

Provisioning tasks to configure the MCP to interwork with traditional

phones must be made on both the MCP side and the existing switching

system.

MCP configuration

The provisioning tasks to configure an MCP user as a CDS user are

described in detail in the User Management chapter of the MCP SIP

Provisioning Client User Guide document.

Switching system configuration

This section describes the steps to configure and activate the SimRing

feature on the Nortel Networks DMS family of TDM switches. Please

refer to feature document AJ4934 Simultaneous Ringing for more

information about the functionality provided by SimRing.

For specific information on enabling the SimRing equivalent feature on

third-party switches, please refer to the documentation provided by that

third-party vendor.

Successful interworking between the MCP and the TDM switch

requires that the TDM switch activate the SimRing (or equivalent)

feature and assign each CDS user a SimRing number. The SimRing (or

equivalent) feature sends “SimRing” calls to the routable and unique

number for each CDS user.

Figure 6 shows the conceptual steps required for provisioning the

SimRing (or its equivalent) feature on a TDM switch.

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Figure 6 SimRing configuration procedure flow

Enable

SimRing

on switch

Assign

SimRing

to a line

View

SimRing

status

Enabling SimRing on the DMS

Before adding SimRing to a line, make sure that table OFCENG is

datafilled properly in order for SimRing to work. Sample datafill is

shown below:

>TABLE OFCENG

MACHINES NOT IN SYNC - DMOS NOT ALLOWED JOURNAL FILE UNAVAILABLE

- DMOS NOT ALLOWED

TABLE: OFCENG

>POS SIMRING_RES_CONTROL

SIMRING_RES_CONTROLY N

>POS SIMRING_CENTREX_CONTROL

SIMRING_CENTREX_CONTROLY N

>POS NO_OF_CLONE_TIDS

NO_OF_CLONE_TIDS 20

For detailed engineering information about configuring these OFCENG

tuples correctly, please refer to the North American DMS-100

Translations Guide Volume 17 of 20 (Document Number:

297-8001-350).

Assigning SimRing on a DMS line

To assign SimRing to an existing line, use the EST (establish)

command of SERVORD.

>SERVORD

SO:

>EST

SONUMBER:

NOW 0 8 16 PM

GROUPTYPE:

>SIMRING

SIMRING_PILOT_LEN:

>HOST 0 1 10 5

SIMRING_MEMBER_DN:

>6215115

SIMRING_MEMBER_DN:

>6215010

SIMRING_MEMBER_DN:

SIMR_STATE: INACT

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>ACT

SIMR_PIN:

>123

Viewing DMS SimRing information

In order to see if a user has SimRing active, the line must be queried

using SERVORD. Use the QLEN (Query LEN) or QDN (Query DN)

commands:

>QLEN 6212064

--------------------------------------------------------------

-----------------

LEN:

HOST 00 1 10 05

TYPE: SINGLE PARTY LINE

SNPA: 613

DIRECTORY NUMBER:

LINE CLASS CODE: IBN

IBN TYPE: STATION

6212064

CUSTGRP:

IBNTST SUBGRP: 0 NCOS: 0

SIGNALLING TYPE: DIGITONE

CARDCODE: 6X17AA

PM NODE NUMBER

PM TERMINAL NUMBER :

OPTIONS:

GND: N PADGRP: STDLN BNV: NL MNO: N

326

DGT SIMRING 0 ACT $

OFFICE OPTIONS:

AIN LNPOFFICE

SimRing has its own command for viewing its properties. Use the

command QSIMR (query SimRing) on a DN to display the set-up of

SimRing on the line:

>QSIMR 6212064

--------------------------------------------------------------

-----------------

Pilot DN: 6136212064

Pilot LEN: HOST 00 1 10 05

The SimRing feature is ACT

Member DN 1 - 6215115

Member DN 2 - 6215010

To query the feature using Table Control, go through table IBNFEAT.

Once in the table, position on the LEN, the NCOS, and the feature:

>TABLE IBNFEAT

JOURNAL FILE UNAVAILABLE - DMOS NOT ALLOWED

TABLE: IBNFEAT

>POS 00 1 10 05 0 SIMRING

HOST 00 1 10 05

SIMRING

0 ACT

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In order to view the member DNs of the SimRing group, go to table

PILOTGRP and position on the pilot's group key (located in the QDN

output):

>TABLE PILOTGRP

MACHINES NOT IN SYNC - DMOS NOT ALLOWED

JOURNAL FILE UNAVAILABLE - DMOS NOT ALLOWED

TABLE: PILOTGRP

>POS 0

0 (6215115) (6215010)$

Configuring third-party voicemail server interworking

The tasks required to configure the MCP for interworking with a

voicemail server can vary, depending on the type of voicemail server.

The MCP can be configured to interwork with the following types of

voicemail servers:

• SIP-based

• trunk-based

• line-based

The provisioning tasks required for the MCP to interwork with each type

of the above voicemail servers are described in detail in the Voice Mail

Servers chapter of the MCP SIP Provisioning Client User Guide

document.

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Accounting management

For more information about accounting functionality on a specific MCP

network node, please refer to the Accounting chapter in the

corresponding documents:

• MCP SIP Application Module Basics

• MCP SIP PRI Gateway Basics

The MCP provides both the public and private charge IDs to the SIP

PRI Gateway. For information on configuring the charge IDs for a user,

please refer to the MCP SIP Provisioning Client User Guide. For more

information on how the charge IDs are used, please refer to the MCP

Accounting Module Basics document.

For information about CS 2000 accounting functionality, please refer to

the CS 2000 Accounting document.

For information about accounting functionality on any third-party

gateway or voicemail server, please refer to the documentation

provided by that third-party vendor.

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Performance management

For more information about performance management functionality on

a specific MCP network node, please refer to the Performance

Management chapter in the corresponding documents:

• MCP SIP Application Module Basics

• MCP SIP PRI Gateway Basics

For information about CS 2000 performance management functionality,

please refer to the CS 2000 Performance Management document.

For information about performance management functionality on any

third-party gateway or voicemail server, please refer to the

documentation provided by that third-party vendor.

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Security and Administration

For more information about security and administration functionality on

a specific MCP network node, please refer to the Security and

Administration chapter in the corresponding documents:

• MCP SIP Application Module Basics

• MCP SIP PRI Gateway Basics

For information about CS 2000 security and administration

functionality, please refer to the CS 2000 Security and Administration

document.

For information about security and administration functionality on any

third-party gateway or voicemail server, please refer to the

documentation provided by that third-party vendor.

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Succession Multimedia Communications Portfolio

MCP Interworking

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 Interworking Basics document without the

express consent of Nortel Networks may void its warranty and void the user’s authority to operate the equip-

ment.

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, Nortel, Northern Telecom, and

NT, are trademarks of Nortel Networks.

Publication number: NN10033-111

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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