Alcatel-Lucent

OmniPCX Enterprise
Communication
Server

Distributed Architecture and

Networking

Document number
8AL020043216DRASA/02
Alcatel-Lucent OmniPCX Enterprise Communication Server release R8.0
Revised July 2007
Copyright © Alcatel-Lucent 2000–2007. All rights reserved.

Passing on and copying of this document, use and communication of

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Notice
While reasonable effort is made to ensure that the information in this
document is complete and accurate at the time of printing, we cannot assume
responsibility for any errors. Changes and/or corrections to the information
contained in this document may be incorporated into future issues.

This document introduces the Alcatel-Lucent OmniPCX Enterprise

Communication Server, its products and features. All documents associated to
this introduction cover most of the aspects for designing offers based on
current manufacturers and business partner agreements. They include
introductory explanations to position the offer in relation to client needs.
References to in-depth documentation are indicated to direct you to product
descriptions or product sites.
Who Should Use this Document?
As an introductory offer, Alcatel-Lucent vendors, clients, partners and
associates involved with the implementation of Alcatel-Lucent PCX systems,
can use this document.
Terms and Abbreviations Used in this Document
Terms and abbreviations specific to each subject may be noted in individual
glossaries.

A general overview document is available in this documentation set.

Changes Summary
The Standard Offer document set is based on a specified release and each
document can contain a “Offer Summary” chapter to indicate product
modifications associated with the release.

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Contents

Alcatel-Lucent OmniPCX Enterprise Communication Server.................1

Distributed Architecture and Networking.......Error! Bookmark not defined.

1 Introduction..................................................................................9

2 Distributed (IP) Architecture with Centralized Call Control............11

2.1 Framing............................................................................................... 14
2.2 Modem and Data Transparency............................................................15
2.2.1 Modem Transparency...................................................................15
2.2.2 Data Transparency.......................................................................15
2.3 Interconnecting IPMGs.......................................................................... 15
2.4 Configuration....................................................................................... 16
2.5 Maintenance........................................................................................ 16

3 Architecture with Distributed Call Control....................................17

3.1 Corporate ABC Networking: Infrastructure.............................................17

3.1.1 Private Network Configurations and Scalability..............................18
3.1.2 Private ABC Networks Using TDM Leased Lines.............................21
3.1.3 ABC Virtual Private Networks (VPNs) on ISDN/PSTN Networks........21
3.1.4 VoIP Networking in ABC Networks.................................................23
3.2 ABC Network Features.......................................................................... 27
3.2.1 Private Numbering (Dialing) Plan in an ABC Network.....................27
3.2.2 Network-Wide Telephone Features................................................28
3.2.3 User Group Features.....................................................................29
3.2.4 Mobility Features.......................................................................... 29
3.2.5 Centralized or Distributed Attendants............................................30
3.2.6 Centralized, Distributed, or Shared Voice Messaging Systems........31
3.2.7 Distributed Contact Center............................................................32
3.2.8 Adaptive Routing.......................................................................... 32
3.2.9 Alternate Route Selection (ARS)....................................................33

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 3.2.10Partial Rerouting........................................................................... 36
3.2.11Management of an ABC Network...................................................37
3.3 Heterogeneous Networking..................................................................42
3.3.1 ISVPN (Country Dependent)..........................................................42
3.3.2 DPNSS Networking........................................................................42
3.3.3 QSIG............................................................................................ 43
3.3.4 H.323 Interoperability...................................................................45
3.3.5 SIP Interoperability....................................................................... 47
4 H.323 Architecture......................................................................49

4.1 H.323 Architecture in Alcatel-Lucent OmniPCX Enterprise Communication

Server: H.323 Gateway............................................................................... 49
4.2 Direct RTP and TransitRTP Modes.........................................................50
4.3 H.323 Integrated Gatekeeper (IGK).......................................................52

5 Session Initiation Protocol (SIP) Networking................................55

5.1 Alcatel-Lucent and SIP.......................................................................... 55

5.1.1 Marketing SIP............................................................................... 56
5.1.2 SIP and Alcatel-Lucent OmniPCX Enterprise Communication Server
57
5.2 Supported SIP Standards...................................................................... 58
5.3 Integration of SIP End-points in the Alcatel-Lucent OmniPCX Enterprise
Communication Server Network..................................................................60
5.3.1 Low Cost Compatible SIP End Points..............................................60
5.3.2 Registering on the Alcatel-Lucent OmniPCX Enterprise CS.............62
5.3.3 Basic SIP Call via the Alcatel-Lucent OmniPCX Enterprise
Communication Server SIP proxy............................................................63
5.3.4 Media Considerations for SIP within the Alcatel-Lucent OmniPCX
Enterprise Communication Server...........................................................64
5.3.5 Calls Made to TDM Trunks.............................................................68
5.3.6 Supplementary Services...............................................................68
5.3.7 Voice Mail Account........................................................................ 69
5.3.8 Authentication/Verification............................................................70

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 5.3.9 What SIP Services can be offered by an “Integrated SIP Phone?”. . .70
5.3.10SIP Phones Management...............................................................71
5.4 SIP Trunking......................................................................................... 72
5.4.1 Public SIP Trunking....................................................................... 72
5.4.2 Private SIP Trunking......................................................................73
5.4.3 Architecture Example....................................................................74
5.5 Supported Telephony Features.............................................................74
5.5.1 Unknown SIP Devices....................................................................77
5.6 SIP Limitations..................................................................................... 78

Note: Some equipment, components, or features (particularly earlier

versions) may not be offered in every country (e.g., U.S. and
Canada). For more information, contact your Alcatel-Lucent
representative.

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1 Introduction
The Alcatel-Lucent OmniPCX Enterprise Communication Server is part of the
next generation of enterprise IP-

PBX communication systems. It is designed around native IP technologies and

allows the customer to deploy mixed or legacy solutions to support a
converged data-voice network.

This flexible architecture allows the Alcatel-Lucent OmniPCX Enterprise

Communication Server to be installed and operated in a totally distributed
design configuration on a single site (campus) or across multiple sites.

There are two types of infrastructures for interconnecting customer sites:

 Centralized call control (central Com Server controlling remote units or
shelves)
 Distributed call control (several Com Servers interconnected to form a
virtual PABX)
Both types of infrastructure can co-exist in the enterprise.

The Alcatel-Lucent OmniPCX Enterprise Communication Server allows a

network consisting of several sites to be built with a high level of service
integration and transparency. It also allows interconnection with legacy PABXs
or with next generation IP-PBXs using digital TDM or IP technology.

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2 Distributed (IP) Architecture with
Centralized Call Control
Only one communication server (Com Server) is required to handle calls
(although it may be duplicated for standby) for all sites. The Alcatel-Lucent
OmniPCX Enterprise Communication Server is supported on an IP appliance
server (IP AS), an IP rack server (IPRS), or an IP crystal server (IPCS). The
server is typically installed at the central site.

The Com Server architecture requires a voice enabled IP network

interconnecting each business site (physical connections often based on
routers over frame relay or leased lines).

Remote sites can be equipped with:

 IP telephony only, with small analog VoIP gateways for fax support (e.g.,
Audiocodes MP1x)
 IP telephones as well as complementary media-gateways for legacy
interfaces (NDDI / NDID analog networks, Z (analog), digital T0, T1, or T2)
 Media gateways used to connect digital phones, fax machines, analog
phones, and DECT base stations to the network
The remote equipment is controlled over the IP network (typically a WAN). All
inter-site connections use VoIP. In most cases, a typical remote site
configuration has between one and 50 extensions.

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The Alcatel-Lucent OmniPCX Enterprise Communication Server is able to
control up to 240 racks (IP Rack Server and IP Crystal Server), distributed over
an IP-based WAN, with a maximum of 90 IP Crystal Server racks.

Note: An IP media gateway (IPMG) consists of one, two, or three racks.

An IP Crystal has one rack.

An IP media gateway provides non-blocking TDM switching for local (internal)

calls and conversion between circuit switched networks and the IP packet
switched network. The characteristics of this configuration include:
Common hardware Crystal hardware
IP media gateway IP media gateway
Voice coding G.723.1, G.729A, G.711

Framing Configurable (see below)

Echo cancellation MADA1, MADA3 GIP4-4, GIP4-1

boards: 128ms boards: 128ms

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 Common hardware Crystal hardware
IP media gateway IP media gateway
MCV boards 32ms GIP-6 boards 32ms

Fax Group 3 9600 b/s maximum

Modem and data transparency See restrictions below

Local circuit switching Yes

Direct LAN switching among IP

Yes
clients

Three-party conference calls Yes

6- and 29-party conference

With GD or GA board With GPA2 board
calls

Internal tone generation for IP

Touch, Reflexes and analog Yes
telephones

Internal or external music-on-

Yes
hold connection

Internal voice guides Yes

Full feature transparency for

Yes
remote media gateways

H.323 gateway On GD and GA board On INT-IP A board

IP domains (including IP-

Yes
Phones and IPMGs of the site)

G.711 (toll quality) for intra

domain calls and
G.723.1/G.729A for inter- Yes
domain calls (bandwidth
optimization)

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 Common hardware Crystal hardware
IP media gateway IP media gateway
Call admission control on the
branch office WAN link
(defines the maximum number Yes
of authorized calls on the WAN
link)

DECT handoff (handover)

No
between IPMGs

Backup Continuity for legacy

Through redundant
phones and IP phones
(duplicate) INTIP A
in case of IP WAN
board
failure

2.1 Framing
Framing is the transmission period of voice packets on the network. The
framing used can be chosen to optimize the voice quality and bandwidth used.
Possible values are shown in the table below.
Algorithm Voice rate Framing (ms) Bandwidth at IP
(KBPS) level (KBPS)
G723 6.3 30 17

G729A 8 20 (default and 24

recommended value)

30 18.6

40 16

G711 64 20 (default and 80

recommended value)

30 74.6

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2.2 Modem and Data Transparency

2.2.1 Modem Transparency

Modem transparency service allows the support of legacy modem

transmission over a LAN through an IPMG.
Constraints:
 Packet loss rate on the IP network should be 0%
 Transmission delay should not be greater than 10ms (round trip delay).
Note: Such constraints on the IP network should be found on a “perfect
switched LAN network” and may not be currently available on a
wide area network (WAN).
 Modems must be compatible with the V92 standard
 Calling modem and called party (modems or trunk) must belong to the
same node (local service)
 Fax transmissions must be sent transparently
 Voice communications are in G711 mode within the IP domain where
modem transparency is activated

2.2.2 Data Transparency

Data transparency allows support of video transmission across the IP network

over the S0 interface.
 The constraints on the LAN are the same as for modem transparency
 Data transparency is available for video transmission, both in local and
network context

2.3 Interconnecting IPMGs

Typically, IPMGs are interconnected through the data network (LAN or WAN).
Optical fibers can also be used (avoids using the customer data network in
campus network). This is based on the use of integrated LAN boards (LAN 16)
and devices called media converters, which convert 10/100BaseT Ethernet for
transmission over fiber (these devices are not provided by Alcatel-Lucent).

2.4 Configuration
Configuring an IP Media Gateway is easy. It consists of:

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 Cabling
 IP parameter programming [dynamic (DHCP) or static IP addressing
(through craft port V24)]
 Overall functional control of local equipment

2.5 Maintenance
Maintenance (moves, adds, changes, or changes to user service parameters)
is simplified and is performed by the central organization.

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3 Architecture with Distributed Call
Control

3.1 Corporate ABC Networking: Infrastructure

The Alcatel-Lucent OmniPCX Enterprise Communication Server can support up
to 100,000 users (TDM, DECT, IP) using various geographic design topologies
(legacy or IP). In case of a legacy topology, the networked configuration can
support up to 32 fully/partially meshed Com Servers and 68 satellite Com
Servers for a total network configuration of 100 systems with full transparent
features.

The networked solution uses the powerful Alcatel-Lucent Business

Communications (ABC) protocol, which is based on and compatible with ETSI
QSIG.

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3.1.1 Private Network Configurations and Scalability

3.1.1.1 Homogeneous Network (up to 100 Nodes)

The Alcatel-Lucent OmniPCX Enterprise Communication Server can be
networked in various topologies: line, star, full meshed, or a combination of
these.

In general, interconnections are made through:

 TDM leased lines
 ABC-VPN over ISDN/PSTN
 H.323 VoIP

Homogeneous networks may consist of Alcatel-Lucent OmniPCX Enterprise

Communication Server R8.0 nodes only, or may include nodes of Alcatel-
Lucent OmniPCX Enterprise Communication Server R7.x, OmniPCX Enterprise
Communication Server R6.x, OmniPCX Enterprise Communication Server
R5.0Lx, OmniPCX Enterprise Communication Server R5.0Ux and OmniPCX
Enterprise 4400 R4.2.

3.1.1.1.1 Interworking of R8.0 with Previous Releases of Alcatel-Lucent

OmniPCX Enterprise Communication Server or A4400
Alcatel-Lucent customers can now migrate smoothly at their own pace from
the OmniPCX 4400 towards the Alcatel-Lucent OmniPCX Enterprise
Communication Server, and benefit from new features and services provided
by the new software release.

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A smooth transition is made possible by interconnecting nodes with different
versions of the Enterprise software in the same network using ABC-F2
protocol. The “message transparency” service included in the ABC-F2 protocol
allows nodes running different software releases to route messages.

This will allow customers to migrate on a node-by-node basis without

compromising existing services or features.

The following releases can be supported on the same ABC-F2 homogenous

network
 Alcatel-Lucent OmniPCX Enterprise Communication Server R5.0Ux
 Alcatel-Lucent OmniPCX Enterprise Communication Server R5.0Lx
 Alcatel-Lucent OmniPCX Enterprise Communication Server R5.1
 Alcatel-Lucent OmniPCX Enterprise Communication Server R6.0
 Alcatel-Lucent OmniPCX Enterprise Communication Server R6.1
 Alcatel-Lucent OmniPCX Enterprise Communication Server R6.2
 Alcatel-Lucent OmniPCX Enterprise Communication Server R7.0
 Alcatel-Lucent OmniPCX Enterprise Communication Server R7.1

3.1.1.1.2 Restrictions on Features

When a new ABC-F2 service is introduced in a software release, this service
can only be handled on a system with the same software release level (“end-
to-end service”). However, due to the “message transparency” service
included in the ABC-F2 protocol, nodes running a lower level of software
release can route new messages.

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Implementing “disallowed” configurations can lead to poor VoIP quality
because of voice compression/decompression cycles. Guidelines are available
to build a VoIP solution that ensures the best possible voice quality and
interactivity to voice communication over an IP network.

3.1.1.2 Supranetwork (More than 100 Nodes)

Several homogeneous Alcatel-Lucent OmniPCX Enterprise Communication
Server networks can be linked together in an extended configuration called a
“supranetwork.” Each subnetwork retains its own external connections,
attendants, centralized applications, and management, but ABC-F (Alcatel-
Lucent Business Communication Feature Transparency) and local PSTN routing
(ARS) services remain consistent throughout the enterprise network. The
advantage of this configuration is that a large company with an international
scope can have an enterprise-wide network while maintaining local
connectivity to the PSTN.

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3.1.2 Private ABC Networks Using TDM Leased Lines

Systems can be interconnected using:

 G.703/G.704 T1/E1 leased lines. ABC signaling is transported using TS16.
Voice is transported in remaining digital frames.
 Systems can be interconnected using several T1/E1 interfaces or a part of a
single T1/E1 (fractional T1/E1). It is also possible to provide ABC-VPN over
leased lines. If the leased line is saturated, additional calls are routed over
the public network (providing that the user has the correct class of
service).
Note: ABC networking over analog leased lines is also supported. In this
case, the Crystal media gateway is used to provide the tie line
interfaces required (ACEM, EMTL boards).

3.1.3 ABC Virtual Private Networks (VPNs) on ISDN/PSTN Networks

3.1.3.1 ABC VPN and Leased Lines

Cost restrictions related to private network designs often lead to bandwidth
limitations. However, to provide maximum availability, private links have to be
sized to support peaks in network traffic. This results in some bandwidth being
unused for long periods of time, and it increases monthly lease expenses. The
best way to optimize bandwidth is to reduce the number of leased lines based

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on average traffic requirements and overflow traffic during peak traffic periods
to an alternative network, as illustrated below:

3.1.3.2 Service Characteristics

The Alcatel-Lucent VPN configuration offers full ABC services regardless of the
network used for voice transport. Telephone signaling always uses a reserved
channel in the digital leased line (D channel).
The application uses virtual DID to overflow calls when the leased line is
saturated. A limited set of DID numbers are reserved for this service
(according to the traffic), so as to integrate users with or without DID numbers
into the network while saving DID subscription costs.
The network manager can manage the overflow precisely. The service can be
assigned user-by-user, and the cost then remains under control with a good
quality of service.
When the leased lines are saturated, a voice guide suggests an automatic
callback request to a user who is not entitled to an overflow operation.
In addition, a flag is associated with each overflowed call in the Alcatel-Lucent
OmniPCX Enterprise Communication Server metering (accounting) application.
This gives the network manager precise data to analyze, whether the service
remains consistent over a period of time.

3.1.3.3 ABC VPN without Leased Line for Voice

For some configurations with low traffic levels, a dedicated digital T1/E1 line is
not cost effective. Today, many enterprises have switched from owned leased
line networks to managed-switched voice networks (voice VPN).

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For such topologies, the Alcatel-Lucent OmniPCX Enterprise Communication
Server interconnects systems with full ABC services using the PSTN to support
ISDN services, or a private QSIG network to transport voice and a permanent
line to transport signaling, as shown below.

IP network:

3.1.3.4 Service Characteristics

The Alcatel-Lucent VPN configuration offers full ABC services. The service uses
virtual DID. A limited set of DID numbers is reserved (according to the traffic
requirements) so the application can integrate users with or without DID
numbers into the network while saving DID trunk expenses.

3.1.4 VoIP Networking in ABC Networks

The Alcatel-Lucent OmniPCX Enterprise Communication Server provides full

feature transparency via leased lines and ISDN/PSTN networks. It provides the
same transparent features when different Alcatel-Lucent OmniPCX Enterprise
Communication Server systems in a private network are interconnected
through an IP network.

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The H.323 gateways and ABC building blocks provide:
 Standard VoIP (H.323): G.711, G.723.1 and G.729A voice codecs and fax
 A high level of feature transparency (ABC protocol)
 Direct RTP between IP phones and distributed media gateways on different
nodes to minimize delay (no transit in the network)
 Ability for calls to overflow transparently to the ISDN/PSTN network when
the IP WAN is out of service or if IP trunks are saturated (See Call
Admission Control chapter)

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3.1.4.1 Call Admission Control on H.323 VoIP Networks Using Internal
Mechanisms
CAC (call admission control) is used to prevent a new call from being routed
over the IP network if there is not enough available bandwidth. The call is
rejected or is rerouted through the PSTN with complete feature transparency.

Due to the characteristics of IP, it is not possible to dynamically adapt to the

available bandwidth on the network. However, the Alcatel-Lucent OmniPCX
Enterprise Communication Server performs CAC statically in two ways in an
ABC network:
 Globally limiting the number of H.323/ABC calls on a node (e.g., A
maximum of ten incoming/outgoing calls. The 11th call will be rejected.)
This solution is well adapted to IP networks with a star topology (a single
WAN link per node).
 Limiting calls per direction when the site has several WAN links to other
sites. For example, 10 calls in direction one, five calls in direction two, etc.
This solution is well adapted to IP networks with a meshed topology.

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3.1.4.2 Call Admission Control in H.323 VoIP Networks Using an External
Gatekeeper
Gatekeepers perform address resolution and bandwidth management for
H.323 communications in an IP network.

Gatekeepers can be used for call admission control in Alcatel-Lucent OmniPCX

Enterprise Communication Server ABC networks.

An internal call between two Alcatel-Lucent OmniPCX Enterprise

Communication Server systems requires a positive answer from the
gatekeeper to proceed further. If a negative answer is received from the
gatekeeper, the call overflows to the PSTN.

3.1.4.3 Call Admission Control for SIP

CAC is used to manage the traffic for SIP phones in the same way as IP Touch
phones or IP Media Gateways as the SIP endpoints are registered in the OXE
SIP Proxy.

A SIP phone making or receiving an intra domain call will use the intra domain
codec for the domain, probably G711.

If the maximum number of domain calls is reached, the SIP phone will be
forced to use the pre-defined extra domain codec.

The CAC-SIP configuration (y/n) limits the number of SIP calls by domain to
maintain the voice quality.

Note: Video is not possible if CAC-SIP is enabled.

3.1.4.4 Management
VoIP management is integrated into Alcatel-Lucent OmniPCX Enterprise
Communication Server management allowing configuration, alarms,
accounting, and performance management.

It is important to regularly monitor the data network to prevent or

troubleshoot VoIP quality problems. Such monitoring may be performed using
external tools (sniffers, etc.). However, the Alcatel-Lucent OmniPCX Enterprise
Communication Server has embedded mechanisms that generate VoIP tickets
(CDR: Call Detail Record) to measure the voice quality of each call. These
tickets contain IP network performance indicators, and concern the transport
of time-sensitive RTP packets (packet loss – including consecutive loss of
several packets, delay, jitter, etc.).

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 Note: See Alcatel-Lucent OmniVista 4760 R4.2 management platform for
more information

3.2 ABC Network Features

Distributed call handling is available for enterprises with multiple sites.

Each site is equipped with a specific Com Server E-CS (duplicated or not),
using the same basic software as typical single-site configurations (classic or
IP configured).

The servers are interconnected to provide the customer with all the expected
benefits of networking (cost optimization, centralized resources, transparent
service for the end user, centralized management, etc.)

These services are supported by the enhanced ABC (Alcatel-Lucent Business

Communications) protocol, which is compatible with the QSIG-GF standard.

The ABC protocol has four main modules:

 ABC-F for feature transparency - provides users with the same telephone
services regardless of their location in the enterprise.
 ABC-M for network wide management - allows centralized, simplified
management.
 ABC-R for network wide routing - optimizes routing for cost/resource
optimization and simplifies routing management.
 ABC-A for network wide applications – allows applications to be centralized
or distributed (attendant call distribution and presentation, voice
messaging, contact centers, etc.)
Because of its support for QSIG-BC and GF protocols, Alcatel-Lucent OmniPCX
Enterprise Communication Server inter-PBX networking interoperates with
third party QSIG devices and/or PABXs.

3.2.1 Private Numbering (Dialing) Plan in an ABC Network

The private numbering (dialing) plan is homogeneous. Directory numbers can

have up to eight digits in an ABC homogeneous network (each number is
unique in the network).

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In a network of homogeneous ABC networks (supranetwork) or a QSIG
network, the private numbering (dialing) plan can be heterogeneous (between
homogeneous networks) and can support up to 20 digits.

3.2.1.1 Moving a User in an ABC Homogeneous Network

A user who moves from one node to another can keep the same phone
number (number portability). This is particularly important in campus
networks where users often move but want to keep their telephone numbers.

3.2.2 Network-Wide Telephone Features

The Alcatel-Lucent OmniPCX Enterprise Communication Server provides the

following network-wide features:
 Basic call
 Number and name identification
 Call/dial by name
 Call waiting indication
 Call offer
 Callback on busy links
 Callback on free or busy extensions
 Call park
 Call forwarding
 Camp on
 Individual call pick-up
 Data call
 Distinctive ringing
 Hold
 Intercom call
 Consultation call
 Broker call
 Transfer
 Conference
 Intrusion/Barge-in
 Unconditional forwarding
 Conditional forwarding
 Substitution (partial)
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 Do not disturb
 DISA
 Text mini-messaging
 Transparency of rotary and DTMF dialing
 Retransmission of last number dialed (redial)
 ISDN/PSTN supplementary services
 Three-party conference
 Six-party casual conference
 29-party meet me conference
 29-party moderated (master) conference
 Announcement/paging on speaker

3.2.3 User Group Features

User group features including busy lamp field (BLF) supervision and DSS
(direct station selection) are also available network wide:
 Associate in network
 Immediate forwarding to associate
 Conditional forwarding to associate
 Manager/assistant filtering/screening team
 Object supervision: free, partially busy, totally busy, ringing
 Hunt groups
 Twin set (simultaneous ringing of wired and wireless phones)

3.2.4 Mobility Features

Mobility features of the Alcatel-Lucent OmniPCX Enterprise Communication
Server include:
 WLAN
 DECT mobility in network: roaming, user rights, and accounting assigned to
home node number
For a DECT wireless handset, roaming is the ability to be located and called
anywhere within the area covered by the base stations. When several Alcatel-
Lucent OmniPCX Enterprise Communication Server systems are in a private
network, the mobile phones must be registered with the “roaming in network”
capability allowing them to be used within the radio coverage area of any
Alcatel-Lucent OmniPCX Enterprise Communication Server in the network. The
users at the guest node have the same access rights as when located on their

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home node because of the transparent transmission of class of service and call
detail records (for accounting purposes). Roaming also allows voice mail
message notification, screening, etc.
 Paging services in network
 Remote forwarding
 Substitution
 Ubiquity services

3.2.5 Centralized or Distributed Attendants

Attendants can be centralized or distributed over an ABC network and have

access to the following features:
 Basic call
 Number and name identification
 Attendant can initiate three-party conference
 Broker call
 Call by name
 Call distribution for decentralized and centralized attendants
 Call offer
 Chained call
 Class of service identification
 Class of traffic indication on line keys (internal call, external call, overflow)
 Do not disturb override
 Dynamic access to user call resources (DECT, voice mail, text mail, etc.)
 Inquiry/consultation call
 Hold
 Intrusion/Barge-in
 Large busy lamp field supervision
 Multi-tenant services
 Overflow of unanswered external calls
 Retransmission of last number dialed (redial)
 Routing and services for multi-company multi-department services
 Station reservation
 Text advertising message on busy user display
 Text mini-messaging

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 Traffic overflow for attendant group or attendants based on caller waiting
time
 Transfer with or without presentation
 Transparency in decadic (rotary) and Q.23 (touchtone) dialing
 Trunk allocation with or without outgoing call restriction
3.2.6 Centralized, Distributed, or Shared Voice Messaging Systems
The voice mail architecture can adopt different configurations:
 The service can be centralized for the entire network (i.e., a single voice mail
system can support multiple networked Alcatel-Lucent OmniPCX Enterprise
Communication Server systems). This service is available using the:
 Alcatel-Lucent messaging system (4635)
 Alcatel-Lucent messaging system (4645)
 Multiple voice mail systems (maximum 127) can be configured within the
network, each shared by a group of Alcatel-Lucent OmniPCX Enterprise
Communication Server platforms using the:
 Alcatel-Lucent messaging system (4635)
 Alcatel-Lucent 4645. Networking of several A4645 messaging systems
(through a VPIM, a single voice mail system can be dedicated to a
single Alcatel-Lucent OmniPCX Enterprise Communication Server
system)

1. Regardless of the configuration, voice mail services are the same as a

single voice mail system for the user and the network manager.
Note: See messaging systems chapter for details.

Note: Alcatel-Lucent One Touch Unified Communication is available. See

specific documentation for details about multi-site environments.

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3.2.7 Distributed Contact Center

The distributed Alcatel-Lucent OmniPCX Enterprise Communication Server CCD

(contact center distribution) capabilities are layered above Alcatel-Lucent
OmniPCX Enterprise Communication Server networking features and therefore
are supported in virtually any ABC network configuration. The Alcatel-Lucent
OmniPCX Enterprise Communication Server CCD offers two powerful built-in
mechanisms for multi-site processing:
 Cooperation between resources situated at different locations. These
cooperative facilities offer call routing options for situations in which the
service level cannot be guaranteed locally.
 Virtual call distribution using the virtual contact center concept, where call-
by-call distribution is made to the most appropriate resource regardless of
its location.
Note: See the Applications chapter for details about applications in a
multi-site environment (including CCx software suite, Web
softphones, etc.)

3.2.8 Adaptive Routing

3.2.8.1 On TDM Leased Lines

Adaptive routing is a powerful routing application that manages call routing in
an Alcatel-Lucent OmniPCX Enterprise Communication Server ABC network
based on TDM leased lines (T2 or T1 or T0). Adaptive routing is a routing
mechanism where the route is calculated before the call is set up. Call setup
includes the route to follow, thus avoiding routing in transit nodes.

Each Com Server knows the exact topology of the private network at all times
and has real time knowledge of route availability and traffic load. The Com
Server immediately selects the best available route based on the load on each
link and the number of hops required (transit PABXs) for completing the call.

Adaptive routing provides a high level of security (avoids loops) and link
optimization, and avoids performing routing management in the private
network.

With adaptive routing, the private numbering (dialing) plan can be distributed
throughout the network without requiring an additional management task,
(e.g., directory number 56000 on node 1, 56001 on node 2, 56002 on node 1,
56003 on node 4, etc.)

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3.2.8.2 Adaptive Routing in ABC-VPN Networks (Over Switched Networks)

Adaptive routing on switched networks is different than on fixed leased lines.

The private voice topology through the PSTN network can be “logically”
defined by creating “VPN jumps” to link the different nodes to form a fully
meshed or partially meshed topology.

Each VPN jump has an associated “cost.”

Before a private network call is set up, the routing service chooses the best
route, which is the route with the computed lowest cost.

3.2.9 Alternate Route Selection (ARS)

Alternate route selection, sometimes referred to as Least Cost Routing (LCR),

is used when a user of an Alcatel-Lucent OmniPCX Enterprise Communication
Server system dials an external number (not internal, not private ABC), for
example, a public call or private call (tie lines, etc.).

ARS allows the communication server to select trunk groups and modify dialed
digits based on a set of parameters such as dialed number, entity of the caller,
or time of day.

ARS is used in a stand alone or networked configuration for optimizing

external telecom call costs.
 Direct or indirect multi-carrier access (stand alone, network)

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 Forced on net (network)
 Break out (network)
ARS services are transparent to network users. The main characteristics of
ARS are:
 20,000 plus routes (not limited)
 4,000 route lists
 One to 30 digit numbers analyzed, with management of exceptions
 Number translation (add or delete)
 Overflow to ten different carriers
 Overflow to other carriers according to user’s class of service
 Informative voice prompt if user overflows to an expensive route
 Informative voice prompt if least cost route is unavailable
 ARS is time dependant based on:
 Day of week: 100 weekly tables
 Hour and minute of the day: five daily tables per weekly table
 24 periods per day

3.2.9.1 Forced On-Net Routing

The ARS/LCR application built into the Alcatel-Lucent OmniPCX Enterprise
Communication Server analyzes the public number dialed before setting up
the call (ARS prefix). If the called party belongs to the private network, the
ARS application will automatically choose the least costly route (in this case, a
private route) and provide ABC feature transparency.

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3.2.9.2 Break Out (Hop off Routing)
When a user calls an external party, the number dialed is analyzed by the
ARS/LCR application. Each Alcatel-Lucent OmniPCX Enterprise Communication
Server knows the exact topology of the private network at all times and has
real-time knowledge of the availability of the different routes and their traffic
load (information via the D-channel signaling). The Alcatel-Lucent OmniPCX
Enterprise Communication Server transparently selects a route over private
infrastructures to make an external call using the PSTN line with the lowest
cost.

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 91 603 860 4873

91 603 860 4873

3.2.9.3 Multiple Carrier Selection

When a user calls an external party, the number dialed is analyzed by the ARS
application. According to the call direction, the best-priced carrier is selected.

The address information is modified to conform to the carrier’s numbering

scheme (by adding, removing, or converting digits), and the called number
and converted number are stored in the call detail record application.

When the lowest priced carrier for this call is not available (congested), the
caller, if so entitled, is diverted to another carrier offering a higher cost. The
user is informed of a route change by a voice message. They can hang up and
try again later if the call is not urgent, or wait for the call to overflow. At the
end of the waiting queue, the call is diverted to the second carrier with the
number converted automatically as described above. The ARS application
allows ten different carriers to be selected successively.
3.2.10 Partial Rerouting

Partial re-routing is an innovative means of forwarding an external call to an

external number to economize the ISDN accesses.

This is applicable to business sets for all call forwarding to external numbers
including:
 Call Forwarding Unconditional (CFU)

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 Call Forwarding on Busy (CFB)
 Call Forwarding on No Reply (CFNR)
In the following diagram, A is calling B, B is forwarded to C

The forwarding FACILITY message is sent back to the public network and if C
replies, the call is setup via the public network and the call is released from
the OXE side as a blind transfer.

Note For accounting purposes the costs of this type of communication can
be divided so that A to B call is charged to A and the B to C call is
charged to B.

Although an AOC message is sent by the operator, the Alcatel-

Lucent OmniPCX Enterprise Communication Server does not
generate a CDR.

3.2.11 Management of an ABC Network

Refer to management chapter for more information.

The ABC management tools are based on PC clients using a Windows NT or

UNIX O/S. They provide a complete set of graphical user interface applications
designed to facilitate day-to-day management operations. The management
workstations are connected through an Ethernet link to a networked Com
Server. Dialog between the management workstation and the Alcatel-Lucent
OmniPCX Enterprise Communication Server is based on standard protocols:
FTP, CMIP/CMISE. Alcatel-Lucent uses CMIP/CMISE for voice management
rather than SNMP as CMIP/CMISE is more powerful and has the ability to
manage a complete set of instances (create, delete, etc.). SNMP is integrated
in order to integrate network supervision with other company management
tools.

Network management can be centralized or distributed.

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3.2.11.1 Configuration Broadcast Mechanism
The management workstation can manage one, several, or all the nodes in a
private Alcatel-Lucent OmniPCX Enterprise Communication Server network.
This mechanism does not use any bandwidth on the B channels of the ABC
links; only the signaling channel (D channel) is used. A network manager can
program or modify any system, station user, or voice mail system parameter
from one unique PC client window. The other network nodes and the
centralized directory are immediately and automatically informed, and their
databases are updated with consistent information.

3.2.11.2 Audit Service

Another management service is the audit service. This service has been
designed to facilitate consistent network monitoring, reconfiguration, and
evolution and to easily integrate voice mail. After a local configuration change,
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a new node automatically learns the network configuration and updates its
database (network translator, phone book, etc.) while the voice mail updates
the user’s database.

3.2.11.3 Voice Mail Management

Only one workstation management operation is needed from the
administrative station’s user management window to configure a user (on any
network node) and assign voice mail services (mailbox and user profile). The
voice mail system automatically creates a virtual user mailbox in the other
network’s voice mail system. Each user can then create distribution lists and
transfer messages, etc., to the other users in the network as though they were
on the same node.

3.2.11.4 Management: SNMP Support on Alcatel-Lucent OmniPCX Enterprise

Communication Server
The management of the Alcatel-Lucent OmniPCX Enterprise Communication
Server can be integrated with standard network management platforms such
as HP OpenView NNM (HP) and UNICENTER (Computer Associates) using
Simple Network Management Protocol (SNMP). These network management
platforms are referred to as hypervisors.

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Hypervisor Integration is a solution for managing converged voice and data
enterprise networks. It is particularly well suited for networks with elements
from multiple vendors.

Hypervisor integration provides two main features:

 Control of the Global Network: This is a standard, joint offer from the
hypervisor provider and Alcatel-Lucent. Customers will use the standard
hypervisor (UNICENTER, HPOV, Tivoli, etc.) with only those customizations
available through the standard hypervisor capabilities. Control of the global
network is provided via an overview of the network (i.e., topology map).
 Topology views may cover every network element and application. Alarm
status is indicated by a change in color of the associated icon. In depth
analysis of the alarm status may be conducted either by using the
hypervisor event manager or by using the proprietary tools designed by the
network item provider. Hypervisors do not provide configuration
capabilities because this is specifically linked to each network item.
 Specific Alarm Handling: This is a customized solution specifically defined
by the customer. The customer will handle all the global network alarms
with specific tools created for their help desk organization. This usually
requires in-house development by the customer.

3.2.11.4.1 Functional Options and How they Work on the Alcatel-Lucent

OmniPCX Communication Server
Functionality varies for each hypervisor. It depends on the generic capabilities
of the hypervisor and on the capabilities of the hypervisor agent Alcatel-
Lucent, partners, or the customer develops that.
Auto-discovery and Topology
The hypervisor uses ping and SNMP Get commands to discover IP components.
The hypervisor replies to SNMP Get commands with Alcatel-Lucent -specific
object IDs or specific extended attributes allowing for the discovery and
representation of Alcatel-Lucent IP components using Alcatel-Lucent -specific
icons on the topology map.

Hypervisors may poll IP components to get their status to display related icons
using related colors.

The status of IP components may be obtained by the hypervisor based on

answer (green)/no answer (red) or more precisely by using the Alcatel-Lucent -
proprietary status attribute (there is no standard status attribute defined in
MIB II) or status trap. MIB extensions and hypervisor agent development are
required to handle the extended status attribute or status trap functions.

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Alarm Handling
Alarms are sent by IP components using SNMP traps. Alcatel-Lucent uses a
specific SNMP trap (defined in the SNMP standard) that allows equipment
providers to create proprietary alarm traps to notify the hypervisor of alarms.
Security
The customer can define communities. SNMP requests can be done only
between hypervisor and OmniPCXs belonging to the same community. The
members of one community share the same community name. The
community name is sent with each SNMP message and acts as a clear text
password. SNMP requests with an unknown community name are rejected.
The new SNMPv3:
 restricts access of MIB information to authenticated hypervisors only. A
hypervisor requires a SNMPv3 account (login name and password) declared
in the CS configuration. The hypervisor agent, which carries out specific
functions (e.g. security), must also be declared. A maximum of 10 SNMPv3
user accounts can be configured on the hypervisor agent
 provides an encryption process for SNMP Protocol Data Units (PDUs). PDUs
can contain address and control information. To cipher the PDUs, a
hypervisor must declare a SNMPv3 passphrase in the CS configuration and
in the hypervisor agent configuration.

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3.3 Heterogeneous Networking
3.3.1 ISVPN (Country Dependent)

ISDN based VPN (ISVPN) provides an entry level networking solution based on
a subset of the ABC private network protocol using only the ISDN network.

No leased line is required for voice or telephone signaling. Voice is transported

over the standard public ISDN network, and the signaling transport is based on
the USS1 service provided by the ISDN network. The features provided by this
service are:
 Distribution of incoming public calls based on virtual DID
 Homogeneous private numbering (dialing) plan
 ISVPN call is seen as an internal call, not a public one
 Name identification (with transport of number)
 Optimized transfer (loop avoidance and optimized call forwarding)
 Callback request (on IP Touch and on Premium and Advanced Reflexes
phones)
 Intrusion/Barge-in
 Other Alcatel-Lucent OmniPCX Enterprise Communication Server features
over ISDN without the added value of ISVPN
 Call distribution in network: centralized attendants
 Inter-working with ABC networks
 Centralized management
ISVPN is also supported on Alcatel-Lucent OmniPCX Office, which is used to
build small, simple enterprise/office networks over ISDN.

3.3.2 DPNSS Networking

The Alcatel-Lucent OmniPCX Enterprise Communication Server offers a

compliant legacy Digital Private Network Signaling System (DPNSS) interface
and protocol, mainly used to connect the voice system to:
 DPNSS installed base not compatible with QSIG
 VPN offer of BT (FEATURENET) and Cable & Wireless in UK
 Alternative carriers (e.g., GSM operators in Germany and Sweden)
 Legacy applications
The DPNSS services supported are:

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 Voice call (six and seven)
 Callback when free (call completion on busy phone) (nine)
 Intrusion/Barge-in (ten)
 Call forwarding (11)
 Hold (12)
 Inquiry/consultation call (13)
 Transfer (13)
 Conference (13)
 Call offer (14)
 Service independent string (16) name
 Call waiting (17)
 Route optimization (19)
 Centralized night service (25)
 Do not disturb (32)

3.3.3 QSIG

The QSIG protocol provides the best method for connecting PBXs from multiple
vendors. QSIG is an open, international standard supported by the world's
leading PBX suppliers. The QSIG protocol is based on ISDN and ensures service
compatibility between public and private ISDN networks. QSIG can work in all
kinds of topologies, with no limitation on the number of nodes supported and
no restrictions on the numbering (dialing) plan.

The Alcatel-Lucent OmniPCX Enterprise Communication Server offers a

network interface implementing QSIG-BC (QSIG basic call). QSIG BC (ETS 300
172) is the part of the protocol that describes the basic call and extends the
public ISDN access protocol for use in private basic rate access and primary
rate access services. As one of the QSIG Forum founders, Alcatel-Lucent
supports this standard and is committed to implementing enhancements as
the OSI committee makes them standards.

3.3.3.1 QSIG Numbering (dialing) Plan

The following numbering plans are supported:
 Heterogeneous numbering plan
 Closed numbering plan
 Open numbering plan (up to 20 digits)

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3.3.3.2 QSIG-BC Services
The level of service of QSIG-BC (basic call) is the same as on traditional tie
lines with the following advantages:
 Caller and connected party number identification
 Voice call

3.3.3.3 Accounting
Each outgoing call on the QSIG trunk generates a Call Detail Record (CDR)
with the duration of the call.

3.3.3.4 Interaction between the Different Systems

An Alcatel-Lucent OmniPCX Enterprise Communication Server user can make a
call on a public trunk line located on another PBX through the QSIG-BC
interface. In this case, the Alcatel-Lucent OmniPCX Enterprise Communication
Server performs external call restriction. In addition, for all incoming QSIG-BC
calls, if the party’s class of service is sent by the connected PBX, the Alcatel-
Lucent OmniPCX Enterprise Communication Server is able to differentiate calls
originating from private subscribers and public trunks, and react as follows:
 Calls originating from private networks are treated as internal calls in
attendant queues.
 Public calls are stored in the unanswered call directory.
 For calls coming from the public network through QSIG-BC, overflow on no
answer or busy user can be processed either by the Alcatel-Lucent
OmniPCX Enterprise Communication Server or in the transit network,
depending on the management.

3.3.3.5 QSIG - GF/SS

GF (generic function) and SS (supplementary services) are the service
enhancements of QSIG-BC. Additional services have been defined as part of
the new standard. Those standards, as soon as they are available, are
implemented in the Alcatel-Lucent OmniPCX Enterprise Communication Server
proprietary network protocol.

The QSIG-GF provides name identification plus transparent features. QSIG-SS

provides telephone services such as:
 Calling line identification presentation
 Connected line identification presentation
 Calling/connected line identification restriction

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 Calling name identification presentation
 Connected name identification presentation
 Calling/connected name identification restriction
 Generic functional procedures
 Call forwarding - unconditional
 Call forwarding - busy
 Call forwarding on no answer
 Call transfer
 Path replacement
 Call completion to busy subscriber
 Call completion on no answer
 Call offer
 Advice of charge
 Call intrusion (Barge-in)
 Do not disturb
 Message waiting indication (when messaging system is the Alcatel-Lucent
4635 or Alcatel-Lucent 4645)

3.3.4 H.323 Interoperability

The Alcatel-Lucent OmniPCX Enterprise Communication Server provides an

H.323 v2 gateway function that supports “fast–connect” for interoperability
between other standard H.323 devices and gateways on the market. The
media transported are voice (G.711, G.729A, or G.723.1) and G3 fax (T.38
encoding).

Compliance with H.323 enables calls between telephones and an H.323 client,
for example, a PC running Microsoft NetMeeting or Netscape Communicator.
Any telephone user can call a PC by dialing its directory number, and the
Alcatel-Lucent OmniPCX Enterprise Communication Server translates the PC
telephone number (E 164) into its IP address.

If the PC user wants to call a telephone or use a trunk, the PC sends the INT-IP
IP address followed by the telephone directory number. The telephone service
level between the PC and PC/telephone is limited to basic calls due to the
current limitations of the H.323 standard.

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In addition, the gateway knows how to register on a standard external H.323
gatekeeper (Alcatel-Lucent or non-Alcatel-Lucent) using the RAS protocol,
obtaining address resolution and call admission control from the gatekeeper.

Only basic call is supported. H.450 services are not supported.

Note: Direct RTP with H.323 is not offered in Alcatel-Lucent OmniPCX

Enterprise Communication Server with third party H.323 gateways
or devices.

Interoperable gateways include the Alcatel-Lucent OmniPCX Office, Alcatel-

Lucent VSA on data switches (voice only), and Audio Codes MP102.

3.3.4.1 Remote Extension and Voice over IP

Remote extension can support both H.323 sets and analog sets.

The following services are available:

 Calling name presentation
 Calling number presentation
 Call detail record for external outgoing calls
This feature should not be used for purposes other than fax device connection.

For more information on remote extensions, refer to the OmniMobility section.

3.3.4.2 Interoperability with OmniPCX Office

The Alcatel-Lucent OmniPCX Enterprise Communication Server can be
networked with Alcatel-Lucent OmniPCX Office over an IP-based network. The
network call service provided is comparable to toll by-pass. Based on the ARS
services the Alcatel-Lucent OmniPCX Enterprise Communication Server and
Alcatel-Lucent OmniPCX Office systems, public calls to a company site can be
identified and routed over the IP WAN to the IP trunk at the distant site.

Because of the ARS services, routing services such as ISDN PRI overflow and
break out are also possible. This solution offers the following services:
 Basic call
 Calling party identification
 Overflow to ISDN / PSTN in case of IP network unavailability
Note: See VoIP design guide rules for building VoIP networks with the
Alcatel-Lucent OmniPCX Enterprise Communication Server and the

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 Alcatel-Lucent OmniPCX Office.

3.3.5 SIP Interoperability

See SIP chapter.

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4 H.323 Architecture
H.323 is used for:
 Peer-to-peer networking between several Alcatel-Lucent OmniPCX
Enterprise Communication Server platforms (see section 3.2 on ABC
networking for services provided using H.323).
 Communications between OmniPCX components and other H.323 devices
(see section 3.3 on heterogeneous networking for services provided using
H.323).

4.1 H.323 Architecture in Alcatel-Lucent OmniPCX Enterprise

Communication Server: H.323 Gateway
The Alcatel-Lucent OmniPCX Enterprise Communication Server supports one or
several H.323 gateways and one internal H.323 gatekeeper:
 The Alcatel-Lucent OmniPCX Enterprise Communication Server H.323
gateway allows communication between H.323 endpoints (H.323 devices,
gateways) with internal users or trunks of the Alcatel-Lucent OmniPCX
Enterprise Communication Server (analog/digital phones, IP-Phones, T2/T1
trunks).
 H.323 devices to call each other and internal users «behind» the H.323
gateway use the Alcatel-Lucent OmniPCX Enterprise Communication Server
H.323 gatekeeper. The registration, admission, status (RAS) protocol is
used between the H.323 device and the gatekeeper.

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The main characteristics of the H.323 gateways on the Alcatel-Lucent
OmniPCX Enterprise Communication Server are:
 H.225, H.245
 Fast-connect
 H.245 tunneling
 Registration, admission, status (RAS) protocol for communication with a
gatekeeper:
 Direct and routed mode
 Connection to an external gatekeeper or to the integrated
gatekeeper
 H.323 annex D (for T38 fax)
GD and GA boards (on Common hardware) and INT-IPA boards (on Crystal
hardware) support the H.323 gateway function. They must be equipped with
compressor daughterboard.

4.2 Direct RTP and TransitRTP Modes

The Alcatel-Lucent OmniPCX Enterprise Communication Server handles RTP
(Real Time Protocol) flows in two ways: Direct RTP and Transit RTP.
Direct RTP mode is used for pure H.323 devices (H.323 Gateways and
terminals) in an IP network to optimize the routing of voice flows: voice flows
are direct between end-to-end IP components (Gateways or Terminals),
without involving any Alcatel-Lucent OmniPCX Enterprise Communication
Server component. Direct RTP allows:
 Optimized flows (without transit) and bandwidth.
 Enhanced audio quality (no additional compression/decompression).
 Architecture optimization (less compression resource required).

Direct RTP mode is used when interconnecting:

 Two devices (IP phone or IP board) in an Alcatel-Lucent OmniPCX Enterprise
Communication Server ABC network
 An IP phone or IP board with OTUC or a SIP phone
The following diagram demonstrates flows with direct RTP.

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 The following conditions apply to Direct RTP mode:In an ABC network, all
the system releases must be R7.0 or later.
 There is no direct RTP flow between SIP and H.323. The DTMF is sent in SIP
and the out band in H.323.
 RTP cannot operate in multi-compression algorithm configurations
 The H.323 signaling link is physically disassociated from the RTP flow.
H.323 signaling is attached to the gateway and RTP is attached to the IP
phone. The GA/GD/INT IPA boards have two functions (see diagram below):
o Each GA and GD board supports up to 30 H.323 signaling sessions
for traffic to the IPMG itself, to an IP phone, or to another IPMG
o INTIPA supports 60 H.323 signaling channels
o Compressor daughter boards on GA/GD/INT IPA boards are used for
TDM/VoIP packetization/compression for traffic going to legacy
telephones connected to the IPMG
Note: RTP direct in H.323 is supported only between Alcatel-Lucent
OmniPCX Enterprise Communication Server systems interconnected
with ABC. RTP direct with other H.323 gateways/terminals is not
supported. RTP flow transits through gateways.

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 Note: The INT-IPB board does not provide H.323 gateway functions. An
INTIPA should be installed in the same cabinet.

In Transit RTP mode, the RTP flow between two IP devices goes through a
media gateway.

4.3 H.323 Integrated Gatekeeper (IGK)

The Alcatel-Lucent OmniPCX Enterprise Communication Server offers an
integrated H.323 gatekeeper, which provides optimized integration and
numbering (dialing) plan management of standard H.323 devices.

In addition to H.323 devices, customers may use wireless VoIP devices

(compliant with the standard 802.11), IP audio-conference devices, or T.38 fax
adapter terminals.

These different types of devices complement the existing IP Touch family of

telephones to meet unique end user requirements.

The integrated H.323 gatekeeper (IGK) allows the following services:

 Gatekeeper registration: devices are assigned a directory number

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 Address resolution: the H.323 device can be identified by the assigned
directory number and/or via its IP address (which is dynamic in the case of
DHCP)
 Gatekeeper direct mode supported. Routed mode not supported.
 The gatekeeper provides the requesting H.323 device with the IP address
of the directory number of the H.323 device. The requesting device
establishes the call directly based on the IP address (H.323 RAS direct
mode scenario).
 Because of the gatekeeper, H.323 devices can receive and make calls to
and from the traditional telephone world through the H.323 gateway that is
part of the Alcatel-Lucent OmniPCX Enterprise Communication Server
gateway
This function allows H.323 devices access to the rest of the traditional
telephone world (internal or external to the enterprise).

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5 Session Initiation Protocol (SIP)
Networking
This section describes Session Initiation Protocol (SIP) and how the Alcatel-
Lucent OmniPCX Enterprise Communication Server accommodates this
method of communications transport. The arrival of low level SIP telephone
sets and their increasing availability on the market offers the possibility of
analog phone replacement. A faster ROI with limited user functionalities.

5.1 Alcatel-Lucent and SIP

Alcatel-Lucent has chosen SIP as one of the corner stones of its standards
based developments for enterprises inter-active communication solutions. SIP
has been evaluated as the emerging standard for the future of real-time multi-
media IP communications for the following reasons:
 SIP evolved out of the Internet community and addresses the
implementation of real-time communication services in IP networks:
 Member of the WEB protocol family: XML, HTML, MIME, etc.
 Directed by the IETF standards body
 SIP will be at the center of carrier/service provider networks and is
currently poised at the center of several major deployment projects
 Completely endorsed by key industry application vendors:
 Standardization has progressed significantly and ongoing work is fuelled by
strong acceptance from the majority of the communication industry
 It is the only protocol that provides “across-the-board” solutions according
to three major application points:
 End-points (devices, stations, phone-sets, soft-clients)
 Trunking (private or public)
 Interactive media-based applications
As any emerging protocol of significant importance for the future of Voice over
IP, the implementation of any ubiquitous solution will occur over a period of
time, and employ a systematic phased approach. The scope and breadth of
the SIP protocol will follow the industry standards process (IETF) and will
progressively enlarge the level of services for each application as the market
advances.

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5.1.1 Marketing SIP

Historically, the enterprise communication market has been dominated by two

desktop device technologies that support and vehicle telephony services,
these being digital and analogue phones. Until the recent emergence of IP
convergence in 2000-1, analogue phones remained the only alternative
technology to digital for the following reasons:
1. Extremely low prices due to economies of scale generated by the
residential market
2. Provides a “minimum” of communication services
3. Vast choice of suppliers and availability
4. Independent of the communication infrastructure vendor: “free to
choose”
5. Easy to deploy – connects to an analogue port
The arrival of IP convergence technologies has raised the issue of how to
address the same basic requirements for low cost communication devices, but
in an IP network. Although there are a number of alternatives using IP; soft
phones, wifi, PDAs, there is still an opportunity to address physical, low cost
desktop devices using IP technologies. In this context, SIP has been identified
as a technological enabler.

Consequently, Alcatel-Lucent has been at the forefront of pioneering and

enabling SIP as an end-point technology, and has recognized early in the
game, the potential to become the ubiquitous low-cost device preference
expected of the market. However, the reality of deployment will depend on a
number of market pre-requisites that must be addressed before SIP becomes a
true and viable solution:
 The devices must deploy correctly a minimum criteria within the SIP
protocol: RFC3261, RFC3262, RFC3264, RFC3265, RFC3515.
 The devices need to support a minimum of IP infra services and
compatibility (codecs, POE, DHCP, QoS tickets etc.)
 The communication system has to control the SIP endpoints across network
topologies (IP Domains, Call Admission Control, redundancy, survivability,
etc.)
 The level of telephony services provided to SIP phones is lower than the
one provided to analogue sets today and is dependant on the capabilities
of the set (e.g. conferencing resources in set). Standard SIP Telephony

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 feature set is still limited today. Careful assessment of end users
requirements must therefore been done before installing the SIP sets.
 Manageability/serviceability tools must ensure smooth/economic
deployment (domain names, proxy addresses, VLAN assignments, etc.)
 Finally and most importantly, the economies of scale are triggered that will
allow true low-cost prices to be achieved: ISPs/ASPs (operators/carriers)
must finally launch mass-market SIP.
5.1.2 SIP and Alcatel-Lucent OmniPCX Enterprise Communication
Server

SIP is an IP signaling protocol designed to establish, to maintain and to end

multimedia sessions between different parties. It operates on a client-server
mode. It is based on the exchange of text messages with syntax similar to that
of HyperText Transport Protocol (HTTP) messages. Elements of the SIP world
are identified by SIP Uniform Resource Locators (URLs) similar to e-mail
addresses.

It is important to note that SIP does not provide an integrated communication

system. SIP is only in charge of initiating a dialog between interlocutors and
negotiating communication parameters, particularly concerning the media
involved (audio, video). The Session Description Protocol (SDP) describes
Media characteristics. SIP uses the other standard communication protocols on
IP: for example, for voice channels on IP, Real-time Transport Protocol (RTP)
and Real-time Transport Control Protocol (RTCP). In turn, RTP uses G7xx audio
codecs for voice coding and compression.

Unlike H.323, the SIP protocol can rely on the IP network transport protocol in
datagram mode User Datagram Protocol (UDP) in addition to the IP network
transport protocol in Transmission Control Protocol (TCP) connected mode.
UDP has the advantage of being an unconnected protocol that facilitates swift
exchanges. It does not guarantee datagram reception and transmission
sequence preservation. Thus, SIP carries out these functions, using
retransmission, acknowledgement and sequencing mechanisms.

As such, SIP is a protocol that can be used for managing Voice over IP (VoIP)
sessions over an IP network but with certain limitations for call handling.

Initially, the Alcatel-Lucent OmniPCX Enterprise Communication Server SIP

implementation addresses the integration of SIP end-points with other devices
controlled by the Alcatel-Lucent OmniPCX Enterprise Communication Server.
 TDM sets and trunks
 IP sets and trunks (IP Touch, H.323 trunks etc.)

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This implementation prepares the communication application architecture, to
provide SIP enabled applications to a variety of sets and devices (whether SIP
or not). This currently concerns:
 OmniTouch Unified Communications
 GVP IP (Genesys Voice Portal)
 Selected Alcatel-Lucent Application Partner Program (AAPP) products such
as Thomson 2030 and FCI IP Ranger
The Alcatel-Lucent OmniPCX Enterprise Communication Server comes with an
embedded SIP Gateway.

The SIP Gateway consists of the gateway function, and a proxy and
registration server:
 The gateway deals with the inter-working functions between SIP and
OmniPCX phones or trunks.
 The proxy deals with SIP routing and SIP end point (phones) location. The
proxy looks up the internal database in order to locate (i.e., find the IP
address) SIP end points.
 The registrar receives registration from SIP end points, and stores mapping
of SIP phone numbers and associated IP addresses in an internal database.
Authentication for the registration uses MD5.
End points can use UDP or TCP transport.
The Alcatel-Lucent OmniPCX Enterprise Communication Server SIP
proxy/gateway is embedded in the communication server, and thus benefits
from the “high availability” feature provided by duplicate (redundant) servers
(Backup avoids re-registering of SIP end points after a failure).
One SIP gateway is supported per system.

5.2 Supported SIP Standards

The Alcatel-Lucent OmniPCX Enterprise Communication Server is based on the
latest standards. The following RFCs are supported (they can be found on the
IETF site at http://www.ietf.org):
 RFC 1321 The MD5 message digest algorithm
 RFC 2327 SDP Session Description Protocol
 RFC 2617 HTTP Authentication: Basic and Digest access authentication
 RFC 2822 Internet message format
 RFC 2833 DTMF in RTP Payload

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 RFC 3261 SIP: Session Initiation Protocol
 RFC 3262 Reliability of Provisional Responses
 RFC 3263 Locating SIP servers
 RFC 3264 An Offer/Answer Model with the Session Description Protocol
(SDP)
 RFC 3265 SIP-Specific Event Notification
 RFC 3323 privacy method for SIP
 RFC 3324 Short term requirements for network asserted identity
 RFC 3325 Private extensions to SIP for asserted identity within trusted
networks
 RFC 3398 ISDN used part to SIP mapping (only to QSIG)
 RFC 3515 Transfer (SIP REFER method)
 RFC 3842 A Message summary and message waiting Indication event
package
 RFC 3891/2 The SIP referred –by mechanism
 RFC 3966 The telephone URI for telephone numbers
 T38 ITU-T Procedures for real –time group 3 fax/communications over IP
Other SIP extensions are drafts (work in progress) versions:
 Transfer: draft-ietf-sip-cc-transfer-05.txt

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5.3 Integration of SIP End-points in the Alcatel-Lucent
OmniPCX Enterprise Communication Server Network
Alcatel-Lucent OmniPCX Enterprise Communication Server offers a SIP
proxy/SIP gateway module that allows SIP end-points (devices/terminals) to
benefit from standard SIP proxy functions (registrar, routing, etc.). A number
of call handling services available on the Alcatel-Lucent OmniPCX Enterprise
Communication Server can also be accessed: homogenous directory number,
public network class of service, voice mailbox including message waiting
notification, calls to PSTN including restriction and charging.
A SIP end point (device) that interconnects with the Alcatel-Lucent OmniPCX
Enterprise Communication Server is called a “SIP integrated phone.”
A SIP integrated phone has an “account” on the Alcatel-Lucent OmniPCX
Enterprise Communication Server system and has the following capabilities:
It registers on the Alcatel-Lucent OmniPCX Enterprise Communication Server
SIP Registrar
By registering, the Alcatel-Lucent OmniPCX Enterprise Communication Server
database links the IP address of a SIP device with a directory number, and
provides:
 A SIP address in the form: sip:<directory-number>@<omnipcx-adress>"
where <omnipcx-address> is the name of the domain or an IP@ of the Alcatel-
Lucent OmniPCX Enterprise Communication Server
 a User name and first name in the Alcatel-Lucent OmniPCX Enterprise
Communication Server phone book
 a Class of service (COS) for barring/restriction and accounting
 an OmniPCX voice mailbox
 an OmniPCX with communications that are protected by digest
authentication
5.3.1 Low Cost Compatible SIP End Points

After careful examination and evaluation of potential suppliers, Alcatel-Lucent

has chosen Thomson and FCI as strategic partners in the field of professional
SIP phones, whose product line is currently residential oriented, but will evolve
and mature in time to become a reliable and deployable enterprise IP device.

Before deciding a SIP versus low-end end-to-end (proprietary) technology, it is

important to understand not only the end-user feature requirement, but the

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evolution of the network and IP foundation technologies that might be
deployed in an overall enterprise communication strategy.
For example:
 POE compliance/options: 802.3af
 Codec support: G711, G.723 & G.729
 DTMF in-band support
 Encryption protocols: AES
 Authentication protocols: 802.1x
 Wideband telephony: 200 – 7KHz voice
 XML services
 Dual subnet redundancy servers
 Etc.
The following product guideline indicates the advantages and visibility of
Alcatel-Lucent’s support of SIP proposed by Thomson, and the alternative end-
to-end (proprietary) technology. This is designed to allow Alcatel-Lucent’s
customers to weigh the consequences and potential strategic impact on their
core-enterprise communication strategy.
Entry level IP phones Thomson ST FCI IP Ranger Alcatel-Lucent
2030 IP Touch 4018
SIP SIP Non-SIP
supported supported

Basic features

Mono-line / multi-line Multi-line(5) Dual-line Multi-line

Hands-free/loudspeaker LS LS Full hands-free
CLIP name and number Yes Yes Yes
Hold & broker call Yes Yes Yes
Transfer Yes* Yes Yes
Call waiting No Yes Yes
Call forwarding Yes Yes Yes
Entry level IP phones Thomson ST FCI IP Ranger Alcatel-Lucent
2030 IP Touch 4018
3-party conf Yes (*) Yes (only Yes
G729A
supported)
Voice-mail services Yes Yes Yes + record,
(access, MWI, forward) menu, VG, …
Advanced features Local Local All OXE

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Entry level IP phones Thomson ST FCI IP Ranger Alcatel-Lucent
2030 IP Touch 4018
SIP SIP Non-SIP
supported supported
Infra support
PoE: 802.3af + option Yes – Class 0 No Yes – Class 2
DHCP / Tftp Yes Yes Yes
Qos 802.1p/DiffServ Yes Yes Yes
Automatic Voice VLAN No (802.1Q) Yes Yes
Assignment
CAC / Codec selection No Yes Yes
Codec: G711, G723.1, Yes Yes Yes
G729A
Wideband ready No No No
Authentication Http digest Http digest IPSec
(application level) auth. auth.
802.1X No No Yes
Communication No No Yes
encryption
Com Servers Yes Yes Yes
redundancy (level 2)
Com Server spatial Yes (with DNS) No Yes
redundancy (L3)

Call Admission Control Yes Yes Yes

(CAC)
Media Gateway No No Yes
signaling backup
Passive Comm Server No No Yes
(PCS)
System integration with Barring, CDR, Yes Yes
OmniPCX ARS
Management integration Partial Yes Yes
with OmniPCX
DTMF: RFC2833, G711 Yes Yes Yes
in band
5.3.2 Registering on the Alcatel-Lucent OmniPCX Enterprise CS

The Alcatel-Lucent OmniPCX Enterprise Communication Server registrar stores

the correlation between directory numbers and the location of the user's
device.
 Example: a SIP user called “Bob” has the following SIP address:
[email protected]
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 sip.company.com is called the domain, and must be the same as the SIP
proxy domain
 5000 is the user address in the domain. It must be a number that will allow
other devices managed by Alcatel-Lucent OmniPCX Enterprise
Communication Server to establish a call.
Additional identities can be added to the address using the system
management tool, for example, a name like [email protected]. This
identity can also be used for calling the device/user.

First, Bob is registered on his SIP phone (by the administrator or by user-
programming according to the set capabilities). The address is
sip:[email protected].

The SIP phone can be reached at IP address 192.168.3.2.

Furthermore, the user has the directory number 5000 in the Communication
Server due to previous configuration.

5.3.3 Basic SIP Call via the Alcatel-Lucent OmniPCX Enterprise

Communication Server SIP proxy

5.3.3.1 SIP to SIP Call

If the above example is used:

If another “SIP integrated phone” dials the DN:5000, the call is routed by the
phone to its "outbound" proxy, which is the Alcatel-Lucent OmniPCX Enterprise
Communication Server SIP proxy.

The SIP proxy locates the called SIP phone and forwards the message
accordingly

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5.3.3.2 Calls from Other Alcatel-Lucent OmniPCX Enterprise Communication
Server Devices to SIP End-points
Users of IP Touch phones must dial a directory number to call the SIP device.
The Alcatel-Lucent OmniPCX Enterprise Communication Server translates the
telephone number to a SIP address to reach the called SIP terminal.

To call a SIP integrated phone with a directory number 5000, an Alcatel-Lucent

OmniPCX Enterprise Communication Server user will dial 5000. The OmniPCX
SIP gateway will then send sip:[email protected] to the SIP proxy.

Alcatel-Lucent IP Touch phones that have an alphanumeric keyboard/keypad

can also use the call by name service (phonebook name) to call SIP integrated
phones, since names of SIP integrated phones can be in the OmniPCX
phonebook.

5.3.3.3 Calls from SIP End-points to Other Devices on Alcatel-Lucent OmniPCX

Enterprise Communication Server
SIP integrated phones can call other devices by dialing sip: <directory
number>@<omnipcx-address>, where the DN can be a set number or a
public number (see section 6.3.4). Some SIP phones provide speed dialing so
those users only dial the directory numbers.

Names of SIP integrated phones can be in the OmniPCX phonebook. These

names can be exported to the A4760 LDAP directory. Therefore, SIP terminals
with a LDAP client can use the Alcatel-Lucent OmniPCX Enterprise
Communication Server “call by name” (“dial by name”) service if they can
automatically append a default domain name (for example,
“@sip.mycompany.com”) to the directory number retrieved from the A4760
LDAP directory.

5.3.4 Media Considerations for SIP within the Alcatel-Lucent OmniPCX

Enterprise Communication Server

5.3.4.1 Type of Media

Three situations can be identified:

a) Communications between “SIP integrated phones” controlled by the

same proxy

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Any type of media can be used according to the SIP device type. For example:
softphones equipped with the right peripheral equipment (camera, handsets,
etc.) will allow users to establish videoconferencing.

b) Communications between “SIP integrated phones” to/from other Alcatel-

Lucent OmniPCX Enterprise Communication Server devices

Voice media is supported including G711 A/U, G723.1 or G729A.

This is also true for communications between “SIP integrated phones” and
other SIP clients (see SIP trunking chapter).

c) Communications to fax machines. Fax T38 Annex D is supported over IP.

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5.3.4.2 Fax Machines
Alcatel-Lucent OmniPCX Enterprise Communication Server is compliant in SIP
with ITU-T 38 D Annex recommendations

The Modem/Data Transparency feature is an Alcatel-Lucent development, so

this feature is not supported for inter-working with third party SIP gateways as
for Audiococodes MP Series SIP gateway.

Fax G3
T38 Fax relay

CENTRAL AREA
MP112
Fax G3

Fax G3 IP
ISDN PRA GD
network

Types of FAX that are supported are Group 3, Super Group 3 (V34 FAX).

Modulations Supported and bit rates supported:

 V27 : 2400 bits/s – 4800 bits/s
 V29 : 7200 bits/s – 9600 bits/s (TBC for Common hardware)
For Fax communications over IP only one modulation/demodulation is
recommended

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5.3.4.3 RTP Flow
Media over IP is transported in Realtime Transport Protocol (RTP) packets. RTP
flows are direct between any SIP device and other IP devices of the Alcatel-
Lucent OmniPCX Enterprise Communication Server, such as IP Phones and
media gateways or an IP voice messaging system (A4645).
Direct RTP means that RTP flows are not transiting through the system
(typically, an IP board) controlled by the Alcatel-Lucent OmniPCX Enterprise
Communication Server. The quality of the voice is therefore better, and the
required number of IP boards is less important.

Scenario 1: Communication between SIP integrated phone and non-SIP

(Alcatel-Lucent User) phone

Scenario 2: Communication between SIP integrated phone and unknown SIP

phone

Scenario 3: Communication between non-SIP (Alcatel-Lucent User) phone and

unknown SIP phone

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Definition: unknown SIP phone is not “integrated” in Alcatel-Lucent OmniPCX
Enterprise Communication Server.

5.3.5 Calls Made to TDM Trunks

A “SIP integrated phone” has an Alcatel-Lucent OmniPCX Enterprise

Communication Server user profile with a specific class of service containing
the barring/restriction class of service.

Call accounting is performed for calls originating from SIP integrated phones
that are directed towards TDM trunks.
Example:
User Bob (directory number 5000) dials a PSTN number. User Bob's SIP
integrated phone is declared in the Alcatel-Lucent OmniPCX Enterprise
Communication Server database. If digest authentication is enabled, the
requesting message is challenged. A new request with security credentials is
sent towards the SIP proxy. The embedded proxy routes the request to the SIP
gateway. The Alcatel-Lucent OmniPCX Enterprise Communication Server
processes the call according to user Bob's public network class of service.
Proper billing information is collected.

5.3.6 Supplementary Services

The following telephone services are provided when SIP sets communicate
with the Alcatel-Lucent OmniPCX Enterprise Communication Server:

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 Registration with or without authentication
 Basic incoming and outgoing calls
 Conference setup and participation
 Broker call
 Three-party conference
 Voice mail access and notification
 On hold and resume
 Unattended and attended transfer
 Call forwarding (unconditional, on busy, on no answer)
 Call diversion on ringing
 Do not distrub
 Call back using call log
 CLIP / CLIR (secret identity)
 DTMF transparancy

Important note:

These are interconnected services supported by the Alcatel-Lucent

OmniPCX Enterprise Communication Server gateway, and not
services provided by the call handling for SIP phones. Therefore,
these services can only be provided if the involved SIP phone is able
to support them. Presentation of the service on the phone is also
independent of the Alcatel-Lucent OmniPCX Enterprise
Communication Server (display, etc.).

For example, certain SIP phones use their own resources to set up a
conference (e.g. the OmniPCX conference bridge is not used when
the user of the SIP phone initiates a conference since the conference
bridge is provided by the SIP phone).

5.3.7 Voice Mail Account

SIP integrated users can have an account on voice messaging systems

supported on the Alcatel-Lucent OmniPCX Enterprise Communication Server:
 A4635
 A4645
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 OTUC My Messaging
SIP integrated phones can be forwarded to the OmniPCX voice messaging
system. They receive a message waiting indication. Navigation in a voice mail
account from a SIP integrated phone is performed via touchtone signals.

5.3.8 Authentication/Verification

The SIP proxy can identify the SIP integrated phone based on information
received through the exchanged messages (invite).

Inside a corporate network, SIP authentication (verification) is not required as

both the phone set and the proxy are managed and controlled by a system
administrator, which ensures an adequate level of security.

Nevertheless, authentication can be set up between the phone and the proxy.

The SIP proxy performs HTTP Digest Authentication on call initiation or on mid-
call messages: this means that any SIP session between the SIP phone and the
proxy is authenticated:

With HTTP digest authentication, user login and password are encrypted (MD5
process). This is based on shared keys authentication.

5.3.9 What SIP Services can be offered by an “Integrated SIP Phone?”

From a technical point of view

a) Native SIP phones or a SIP application on a PC like the eyeBeam

softphone from xTen

b) Small FXS gateways like Cisco ATA 186, Audiocodes MP1x series
connecting analog devices, with one directory number per analog port.

For example, the same OMNIPCX SIP Proxy will see a gateway with two analog
accesses as two different SIP devices (with two registrars). These gateways
should be able to register.

The Alcatel-Lucent Application Partner Program has tested MP1x Audiocodes

successfully. Therefore, connectivity of a SIP analog FXS gateway can be
supported on the Alcatel-Lucent OmniPCX Enterprise Communication Server.

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Other gateways (T2/T1 to SIP gateways, for instance Cisco 2600/3600), which
are seen as “hunt groups,” are not able to benefit from SIP integrated phone
services (see SIP trunking).

5.3.10 SIP Phones Management

Unlike phonesthat are stimuli terminals completely controlled by the system,

SIP phones are “intelligent” end points that require item (local) management.

Generally, the phones have integrated Web server management tools. These
are required to manage SIP phones on an individual basis.

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5.4 SIP Trunking
SIP trunk capacity is an example of the evolution speed in SIP technology.
Capacities have increased from 360 in R6.2 to 900 in R7.0. This means that
the Alcatel-Lucent OmniPCX Enterprise Communication Server can support
900 simultaneous communications.

The aim of SIP trunking is to allow connections to a SIP operator with the same
service level as ISDN. This includes the possibility of different types of access,
compatibility with existing services, and integration with the management
tools for configuration, observation and call metering.

Several SIP trunks groups can be created:

 One main trunk group is created for the external SIP extensions and OTUC
 Separate trunk groups are created for external gateways.

5.4.1 Public SIP Trunking

The communication server performs the SIP trunk group signaling

When the OXE is connected to the PSTN via a SIP carrier:

 SIP signaling exchanges are between the Communication Server and the
SIP carrier gateway.
 Voice flows are exchanged between Alcatel-Lucent IP phones or IP Media
Gateway and the carrier Trunking gateway.
 The SBC is the access point to the carrier infrastructure

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 Carrier SIP
Carrier Gatew ay
SIPTrunk
Group infrastructure
PSTNtrunking
CS gateway
SIPSignalling D

Custom er OXE SBC

infrastructure Signalling
DirectRTP PSTN

A Session Border Controller (SBC) is a gateway that typically resides at the

boundary of an IP network. This device is used to control the signaling and
media streams. SBCs are put into the signaling and /or media path between
calling and called party.

The call barring category, connection category and entity are all processed

Call Detail Records (CDR) are generated as for legacy trunk groups, and the
Call duration mode is used for accounting purposes.
CAC
Call Admission control (CAC) is configured in the OXE to control the number of
calls through the PSTN trunking gateway and so controls admissible calls
between SIP endpoints using the IP domain feature. CAC is used for:
 SIP phones
 Analog sets or FAX device behind SIP gateways (declared as SIP users)
Note: CAC is not applied to SIP video flows, Bandwidth must be calculated
to avoid deterioration in voice quality.

5.4.2 Private SIP Trunking

The Alcatel-Lucent Omni PCX Enterprise Communication Server is capable of

interworking with private external SIP gateways as for private legacy trunk
groups. This means that numbers containing a prefix can be routed in the
same way as QSIG trunk groups.

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Private trunking services including management are the same as for public
trunking except that the transfer and call forwarding are performed with SIP
protocol.
Current limitations for SIP trunking are:
 A maximum of 5 SIP trunk groups
 Up to 960 trunks per trunk group

5.4.3 Architecture Example

In this example some branch offices are equipped with a single analog SIP
gateway for both analog sets and FAX devices. The result is Direct RTP for
SIP voice an direct T38 for Fax communications
Central area IP Domain 1 / CAC = 2

Fax G3
IP Domain 4 / CAC = 10 MP 112
M
SIP signaling
Fax G3 FAX flows
WAN
OXE RTP flows
signaling Fax G3

D
C/D

C/D MP 114
M/D
: Modulation/ Demodulation
IP Domain 2 / CAC = 3

5.5 Supported Telephony Features

The tables in this section contain the telephony features supported by the
Alcatel-Lucent OmniPCX Enterprise Communication Server with current
limitations.

In the following tables:

User side feature means if service is available to user on OmniPCX side when
communicating with/through a SIP trunk
SIP public trunk side means that a service activated on OmniPCX will involve
the SIP network to perform the service. If yes, means that the service can also
be activated on carrier side.

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 SIP SIP
SIP-related User public privat
Features side trunk e Comments
side trunk
(NGN side
involve
d)
Basic call OutGoing Yes Yes Yes Block dialing for OG
with number and
name display Canonical form for public
numbering

<sip:
+33390677517@my_domain>

RFC 3261

Basic Call InComing Yes Yes Yes Canonical form for public
with DDI with numbering
number and name
display <sip:
+33390677517@my_domain>

Range of DDI numbers/users

must be provisioned at the NGN
side

RFC 3261

Calling line / name Yes Yes Yes Provided there is compliance

identification with RFC 3323, 3324 & 3325 is
presentation ensured and OXE is considered
(CLIP/CNIP) as a trusted element

Calling line Yes Yes Yes Provided there is compliance

identification with RFC 3323, 3324 & 3325 is
restriction (CLIR) ensured and OXE is considered
as a trusted element

Connected line / Yes Yes Yes

name identification
presentation (COLP)

Connected line / Yes Yes Yes

name identification
restriction (COLR)

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 SIP SIP
SIP-related User public privat
Features side trunk e Comments
side trunk
(NGN side
involve
d)
Call forwarding Yes No Yes User side: Performed by joining
unconditional (CFU) both half calls within Alcatel-
and on busy/No Lucent OmniPCX Enterprise
answer (CFB/CFNR) Communication Server. RTP
optimized.

Private trunk: provided there is

compliance with RFC 3261 (302
Moved Temporarily message)

Call hold (CH) Yes Yes Yes Using SIP-related RFC : RFC 3264

Enquiry / Yes Yes Yes Using basic SIP features like

Broker/Conference invite, call hold and conference
(Conference in OXE, not using
SIP conference packages)

Attended Transfer/ Yes No Yes User side: Performed by joining

both half calls within Alcatel-
Early Attended Lucent OmniPCX Enterprise
transfer Communication Server. RTP
optimized.

Private trunk: Provided there is

compliance with RFC 3515 (SIP
Refer method)

DTMF overdialling Yes Yes Yes Provided there is compliance

with RFC 2833

Fax support Yes Yes Yes T38 ITU-T annex D

Same integration Yes No Features like DDI, differentiated

with user and system ringing, parking, overflow to
call handling attendant, charging, overflow to
features than for ISDN
ISDN trunks

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 SIP related SIP SIP private Comments
Features public trunk side
trunk
side
Authentication for Yes Yes Provided there is compliance
OutGoing calls/ with RFC 2617 & RFC 1321
Incoming calls

Dynamic registration Yes Range of DDI numbers/users

of OmniPCX SIP must be provisioned statically at
gateway to Carrier the NGN side
SIP proxy (with or
without
authentication)

Alternate SIP proxy Yes OXE side

through ARS

SIP keep alives to SIP Yes

proxy

Message Waiting No Yes Provided there is compliance

Indication with RFC 3842

Several SIP trunks Yes -Yes OXE side

Call Admission Yes Yes « IP domain CAC » takes into

Control SIP account SIP phones inside IP
domains

These SIP devices typically have no directory number in the system and are
not registered on the Alcatel-Lucent OmniPCX Enterprise Communication
Server SIP proxy.

5.5.1 Unknown SIP Devices

Unknown SIP devices (that are not SIP integrated phones) cannot be reached
directly from non-SIP devices, as their SIP identity is not known within the
Alcatel-Lucent OmniPCX Enterprise Communication Server. These devices can
only be reached through external SIP gateways (see previous chapter).

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Through configuration of the system, it is also possible to stop SIP incoming
calls from unknown SIP devices to Alcatel-Lucent OmniPCX Enterprise
Communication Server sets.

5.6 SIP Limitations

The maximum supported SIP sets per Alcatel-Lucent OmniPCX Enterprise
Communication Server is 1000 (analog behind MP112 SIP GW included)

A SIP set cannot be a Hotel set or a Call Center agent.

A SIP set cannot be supervised by CSTA.

A SIP set cannot be part of a group of sets or a pick-up group, and cannot be
part of a Boss/Secretary configuration.

Military features such as MLPP and Call Restriction do not support SIP.

The Multi-tenancy feature does not support SIP.

SIP sets cannot access phone feature prefixes.

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