03 System Services

Alcatel-Lucent Enterprise Communication Solutions
2013 Offers
Standard Offer
Chapter 03 System Services
October 2013 Offer - Ed.01

Ref.: ENT_MLE_015926
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This document introduces the Alcatel-Lucent OpenTouch and OmniPCX Enterprise Communication
Server, their 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-
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As an introductory offer, this document can be used by Alcatel-Lucent vendors, clients, partners and
associates involved with the implementation of Alcatel-Lucent systems.
ENT_MLE_015926 ed 01Chapter 03: System Services 2/75 Alcatel-Lucent Enterprise Communication Solutions
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Table of contents
1. Architectural reliability ............................................................................................................ 6

1.1 Communication server redundancy (duplication) ............................................................................. 6
1.1.1 IP rack server (Common hardware) and Appliance server ................................................................................. 6
1.1.2 Com server changeover ...................................................................................................................................... 7
1.1.3 Database consistency ......................................................................................................................................... 7
1.2 Communication server survivability for OpenTouch users ................................................................ 8
1.3 Secure call processing with the passive communication server ........................................................ 9
1.4 Secure call processing for OpenTouch users with the passive communication server .................... 11
1.5 Backup signaling for IP media gateways ......................................................................................... 12
1.6 IP Touch sets survivability ............................................................................................................... 13
1.7 Business continuity of IP Touch in case of IP failure ........................................................................ 14
1.8 Business continuity for out of service Digital/Analog devices ......................................................... 14
1.9 Fallback when SIP Voice Messaging is not reachable ...................................................................... 14
1.10 Dual LAN attachment ...................................................................................................................... 14
1.11 Uninterruptible power supply ......................................................................................................... 15
2. Time and date management .................................................................................................. 16
2.1 System time clock ........................................................................................................................... 16
2.2 NTP service on OmniPCX Enterprise components ........................................................................... 16
2.2.1 NTP services ...................................................................................................................................................... 16
2.2.2 Synchronization method .................................................................................................................................. 17
2.2.3 NTP and current Alcatel-Lucent OmniPCX Enterprise Communication Server time setting services............... 17
3. Additional IP services ............................................................................................................. 18

3.1 Fax machines................................................................................................................................... 18
3.2 Modem and data transparency ....................................................................................................... 18
3.2.1 Modem transparency ....................................................................................................................................... 18
3.2.2 Data transparency ............................................................................................................................................ 19
3.3 Framing ........................................................................................................................................... 19
3.4 Trivial File Transport Protocol (TFTP) .............................................................................................. 19
3.5 Dynamic Host Control Protocol (DHCP) server ................................................................................ 19
3.6 Automatic VLAN Assignment (AVA) ................................................................................................ 21
3.6.1 Different VLAN assignment methods ............................................................................................................... 21
3.6.2 Automatic VLAN Assignment (AVA) through 802.1 AB/ LLDP MED.................................................................. 22
3.6.3 Automatic VLAN Assignment (AVA) through DHCP .......................................................................................... 22
3.7 VoIP supervision ............................................................................................................................. 24
4. General call routing ............................................................................................................... 25
4.1 Class of Service (CoS)....................................................................................................................... 25
4.2 Routing table................................................................................................................................... 26
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4.2.1 Time and Class of Service (COS) tables ............................................................................................................. 27
4.2.2 Routing in the network ..................................................................................................................................... 27
4.2.3 Night service ..................................................................................................................................................... 27
4.3 Call restriction for alarms and emergencies .................................................................................... 27
4.4 Priority calls Multi-Level Precedence and Pre-emption (MLPP) .................................................... 28
4.5 Call accounting (metering) .............................................................................................................. 29
5. Business Telephony routing for branch offices ........................................................................ 30
6. Integrated accounting ........................................................................................................... 32
6.1 Configuring the application ............................................................................................................. 33
6.2 User counters view ......................................................................................................................... 33
6.3 Real time records printing ............................................................................................................... 34
6.3.1 Ethernet ............................................................................................................................................................ 34
6.3.2 V24 support ...................................................................................................................................................... 34
6.4 Printing traffic analysis records (optional) ...................................................................................... 36
6.5 Printing financial reports (optional) ................................................................................................ 36
7. OmniPCX Record.................................................................................................................... 38
8. Integrated voice guide services .............................................................................................. 40
8.1 Integrated voice guides ................................................................................................................... 40
8.2 Voice Guides ................................................................................................................................... 41
8.2.1 Chained voice guides ........................................................................................................................................ 42
8.2.2 Voice messages................................................................................................................................................. 42
8.2.3 Message examples............................................................................................................................................ 42
8.2.4 Main characteristics ......................................................................................................................................... 43
8.2.5 Customized voice messages ............................................................................................................................. 44
8.3 Internal Music-on-hold.................................................................................................................... 44
8.4 External Music-on-hold ................................................................................................................... 45
8.4.1 Distinctive features ........................................................................................................................................... 45
8.4.2 Connection ....................................................................................................................................................... 45
9. IP Telephony domains and related features ........................................................................... 46

9.1 Call Admission Control (CAC) and codec selection .......................................................................... 46
9.2 Defense mechanisms ...................................................................................................................... 48
9.3 Multi-time zone .............................................................................................................................. 48
9.4 Multi-country .................................................................................................................................. 48
9.5 Calling line identification (CLI) per IP Domain ................................................................................. 49
10. SIP and web services .......................................................................................................... 50
10.1 Marketing SIP .................................................................................................................................. 50
10.2 SIP basics ......................................................................................................................................... 51
10.3 SIP and OpenTouch ......................................................................................................................... 51
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11. Session Initiation Protocol (SIP) networking ........................................................................ 53
11.1 Session Initiation Protocol (SIP) networking ................................................................................... 53
11.2 Alcatel-Lucent and SIP ..................................................................................................................... 53
11.2.1 SIP and Alcatel-Lucent OmniPCX Enterprise Communication Server........................................................... 53
11.3 Supported SIP standards ................................................................................................................. 54
11.4 Integration of SIP end-points in the Alcatel-Lucent OmniPCX Enterprise Communication server
network ...................................................................................................................................................... 55
11.4.1 Integration of SIP end-points in the PCX network........................................................................................ 55
11.4.2 SIP end-point configuration and registration ............................................................................................... 58
11.4.3 SIP end-point deployment on a large scale .................................................................................................. 59
11.4.4 Telephony services available for SIP end-points .......................................................................................... 60
11.4.5 Basic SIP call via the PCX SIP proxy............................................................................................................... 62
11.4.6 Media considerations for SIP within the PCX ............................................................................................... 62
11.4.7 Calls made to TDM trunks ............................................................................................................................ 64
11.4.8 Voice mail account ....................................................................................................................................... 64
11.4.9 Authentication verification .......................................................................................................................... 65
11.4.10 Keep-alive dialog .......................................................................................................................................... 65
11.4.11 What SIP services are offered by an integrated SIP set? ............................................................................. 65
11.5 SIP trunking ..................................................................................................................................... 65
11.5.1 Public SIP trunking........................................................................................................................................ 66
11.5.2 Domain name resolution.............................................................................................................................. 67
11.5.3 Architecture example ................................................................................................................................... 67
11.5.4 Supported telephony features ..................................................................................................................... 68
11.6 Unknown SIP sets............................................................................................................................ 71
11.7 SIP communication operations at Communication server changeover ........................................... 71
12. H.323 architecture ............................................................................................................. 72
12.1 H.323 Architecture in Alcatel-Lucent OmniPCX Enterprise Communication Server: H.323 gateway72
12.2 Direct RTP and transit RTP modes ................................................................................................... 73
12.3 H.323 integrated gatekeeper (IGK).................................................................................................. 74
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1. Architectural reliability
The network requirements of each client are defined by the balance between service continuity,
network needs, and cost. The resulting choice is inevitably linked to the Return On Investment (ROI)
that can only be defined by the client.
As the system reliability and availability increase in importance, the requirements for additional
hardware increase. This section describes some of the options and features.
1.1 Communication server redundancy (duplication)

The Alcatel-Lucent OmniPCX Enterprise Communication Server provides a unique and secure backup
mechanism when mission critical applications require high resiliency. Communication Server
redundancy (duplication) allows switch over from one communication server to its mirrored
communication server through an IP link.
In this type of configuration, two Communication Servers coexist in the same system. One server is
active, and is the primary Communication Server. The other server is constantly in watchdog mode
on standby. If the primary server fails, the standby automatically takes over.
Note: To avoid possible communication server performance distortion, the CPU hardware must be
identical for both the main and standby communication servers.
1.1.1 IP rack server (Common hardware) and Appliance server

During normal operation, a polling dialog is established and maintained between the main and
standby communication servers. Interruption of this dialog indicates to the standby machine that the
main communication server not available. The standby communication server then takes over as the
main communication server.
Primary and secondary Communications Servers can be located in two geographical sites and can be
in different IP sub networks thanks to dual IP addresses.
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All the following applications are able to address the dual IP addresses:
Network Management Center OmniVista 8770
Voice mail applications
Alcatel-Lucent OmniTouch Contact Center - Standard Edition solution except CCOutbound
solution
Alcatel-Lucent OmniTouch IVR (CC-IVR)
XML Web Services (My Phone WS, My Messaging WS, My Management WS, My Assistant WS,
IP Touch XML Services)
Note: To avoid possible desynchronization between databases, it is recommended to secure the IP
link between primary and secondary Com Servers.
1.1.2 Com server changeover

When the changeover takes place, active calls are maintained and calls in the process of being set up
are interrupted.
The data involved in the updates includes:
Status of the different elements (including boards and terminals)
Configuration information
Accounting tickets (call detail records)
CCD data
Note: A manual change over command is also available in maintenance mode.
In case of ABC network configuration (multiple Communication Servers interconnected through ABC-
F2 TDM or IP links), active networked communication are maintened when the changeover takes
place.
1.1.3 Database consistency

The standby Com Server is updated continuously and is ready to act as the primary server at any
time. ALL data, including databases, applications, and communication-handling software, is run in
parallel on both servers. This operation ensures a reliable, secure switch over from one server to the
other.
When the standby Com Server is unreachable, the main Com Server stores for a limited period of
time the history of MAO commands used to update is database. Two situations can occur:
If the standby Com Server becomes operational before the expiration of the storage duration,
the main Com Server sends MAO commands to the standby Com Server which
automatically updates its database. As a result, the two databases become consistent.
If the standby Com Server is still unreachable after the expiration of the storage duration, the
main Com Server stops storing MAO commands and delete them. When the standby Com
Server becomes reachable, the two databases must be consistent via a database cloning
operation (or master copy operation).
Automatic database synchronization is available for Com Servers located on CS-2 boards (Common
Hardware), Appliance Servers and Blade Centers. This operating mode is identical in a double main
configuration.
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Information handled by MAO commands are:
Agent login/logout (Alcatel-Lucent OmniTouch Contact Center - Standard Edition)
Configuration of set parameters (secret code, language, user name, keys, etc.). This applies
to digital and cordless sets
Set status (in or out of service)
Interphony service
Hotel/Hospital application
Configuration of attendant parameters (and attendant groups)
Configuration of entity parameters
Configuration of Call Distribution Tables (CDT)
1.2 Communication server survivability for OpenTouch users

The Communication Server Duplication service can be deployed in an OpenTouch configuration to
improve the reliability of system operations.
An Alcatel-Lucent Communication Server hosted on an Appliance Server (or a CS-2 (Common
Hardware board)) is used as stand-by Com Server. It takes over when the OpenTouch is not
reachable or out of service.
Communication Server Duplication offers the same level of service in both standard and OpenTouch
configuration, such as:
Real-time duplication of telephony data, provided by the Com Server hosted on the
OpenTouch, including traffic observation, accounting records and Contact Center data
Note: Data relating to CCA, OmniVista 8770, Alcatel-Lucent OmniTouch Unified Communications, and
OmniTouch Fax Server are not duplicated.
Detection of OpenTouch Server loss and switchover
Continuity of telephony services when a switchover occurs (e.g. communications established
are maintained), except for:
o ABC communications on hybrid logical links (e.g. ABC link through IP)
o Communications with SIP devices or established via a SIP trunk group
A switchover can occur when the OpenTouch server activity is interrupted, due for example to a
power failure or network problem. The dialog established between the OpenTouch server and Stand-
by Com Server is lost, and the stand-by Com Server becomes the Main Com Server.
The Stand-by Com Server and OpenTouch server can be located:
In the same IP sub-network. Continuity of telephony services is ensured, when the
OpenTouch server is out of service (e.g. power failure)
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In different IP sub-networks (spatial redundancy). Continuity of telephony services is ensured,
when the OpenTouch server is either not reachable (e.g. IP link failure) or out of service
1.3 Secure call processing with the passive communication server

The Passive Communication Server (PCS) provides call handling services to a media gateway or group
of media gateways if the Alcatel-Lucent OmniPCX Enterprise Communication Server is unavailable.
If the IP links to the site which hosts the Communication servers are broken or the Communication
servers are out of service, call processing continues at a local level.
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PCS use in SIP environment:
The PCS can rescue SIP phones and SIP trunk groups (SIP trunking), provided the SIP devices
(external gateways/SIP proxies, SIP phones) can handle primary and secondary DNS server
addresses (to access the main and backup Com Servers), and a proxy server address (to access the
PCS). The Domain Name (DN) resolution is performed with DNS A (DNS SRV is not supported).
SIP registrations on the main Com Server are not duplicated on the PCS. When the PCS becomes
active (Com Servers are down), the SIP sets must register on the PCS to be in service.
The Alcatel-Lucent OmniTouch 8450 Fax Software solution does not support backup SIP proxy.
In normal conditions:
The Communication servers control the calls within the network
The IP Phones and/or Media Gateways within a region are defined for the PCS
Automatic or manual synchronization of the region is carried out on a general or individual
PCS basis
If Communication Server loss occurs:
The telephony services are restarted locally
Centralized services such as voice mail are no longer available
All standalone features defined inside active call processing are maintained by the PCS
including OmniTouch Contact Centers.
CDRs (Call Detail Record) are recorded in the PCS
When the IP link to the OmniPCX Enterprise is back IN SERVICE, the PCS switches to standby mode
either after a timer or at a configured time (typically the PCS is configured to switch at night to avoid
telephone disruption).
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When the PCS switches to standby mode:
IP phones and media gateway reboot and are under the OmniPCX Enterprise control
Accounting tickets (CDRs) are transmitted to the OmniVista 8770
The PCS is defined with the same provisioning level as the Communication Server. It can be hosted
on a Common Hardware CPU, Appliance Server, or Blade Server.
Note: Because of the differences in the Database structures, IP Crystal Servers cannot be used in a
PCS configuration. A PCS cannot be duplicated.
1.4 Secure call processing for OpenTouch users with the passive
communication server
The Passive Communication Server (or PCS) is a feature of the Alcatel-Lucent OmniPCX Enterprise
Communication Server. It offers a SIP survivability level of service for OpenTouch users.
In a OpenTouch configuration, the PCS provides the continuity of telephony services and the Contact
Center in case of:
Loss of the OpenTouch host:
o If the Communication Server Duplication service is not used, the PCS secures call
handling when it is deployed on the main site for all Media Gateways
o Additional PCSs can be deployed per Branch Offices (per Media Gateway)
Breakdown of IP network links:

o WAN/LAN out of service
o IP links lost between central site and Media Gateways
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If the OpenTouch server crashes (configuration without using the Communication Server duplication),
a phone reset takes place. After this reset, only telephony and the Contact Center CCD/CCS remain
operational.
The CCA, CC-IVR, OmniVista 8770 Server, Alcatel-Lucent OmniTouch Unified Communications, XML
Web Services, and OmniTouch Fax Server are no longer available.
The telephony service remains operational for a period of thirty days. It is recommended to repair the
OpenTouch server as soon as possible.
Note: Other PCSs can be deployed in Branch Offices (typical OmniPCX Enterprise deployment).
1.5 Backup signaling for IP media gateways

Note 1: This feature is only available for Standard Users.
If the IP link between the communication server and a common hardware IP media gateway is lost, a
backup signaling link is used to re-establish the signaling path over the PSTN. This service is designed
to ensure continued telephone service at remote sites.
During the backup connection, users can make and receive calls over the local PSTN network
connection (see the next section for more details), and VoIP calls between the remote and all the
other sites can be redirected via the public network.
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The communication server monitors the links to each media gateway using a polling dialog. An
interruption of this dialog informs the communication server that a failure has occurred. The
communication server then attempts to reach the remote media gateway over the PSTN (via GD
internal modems). During this time, the remote media gateway restarts.
Return to normal
In backup mode, the media gateway periodically polls the IP network.
When the IP network connection is re-established, the media gateway switches the communication
server signaling back to the normal link.
All calls are maintained while normal IP network signaling is re-established.
In addition, when the IP network is unavailable, voice inter-site communications can be established
via the public network. Dialed internal numbers are translated automatically into public numbers.
This mechanism can also be used when the Call Admission Control of the remote site is reached.
Note 2:
Because a minimum of telephone services are offered for inter-site communications, it is not
recommended to use this private to public overflow permanently.
Within the Alcatel-Lucent OmniPCX Enterprise Communication Server architecture, it is possible to
mix both features, PCS and "Backup Signaling Link of Media Gateways", if they do not back up the
same sites.
1.6 IP Touch sets survivability

Dynamic IP address allocation of IP Touch sets (Alcatel-Lucent 8 Series sets), located in remote sites,
can be done with a central DHCP server. During its first initialization, the IP Phone stores all the IP
addresses (its address, TFTP server address, default gateway address, and subnet mask address)
given by the DHCP Server. In this way, when an IP Phone is not able to reach a DHCP server (for
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example, IP WAN network outage), the IP Phone initialization is possible and it can establish its
telephony signaling link with the Communication Server via the media gateway and the backup
signaling link on PSTN.
The signaling link between a remote IP Touch and the Communication Server is direct. This means
that, even if the remote Media Gateway is out of order, the IP Touch is able to communicate with
other devices.
Note: Static IP addressing from the IP phone is always available. This feature is compatible with the
IP Touch Security feature providing encryption and firmware integrity.
When there is no PCS and no backup signaling link, SIP survivability may apply to Alcatel-Lucent 8
series phone Extended Edition. When the Com Server is not reachable via the IP network, the set
operates as a SIP set via the SIP proxy.
Calls are established by the SIP proxy server through the PSTN.
Rescued phones must be situated in domains where there is a router implementing SIP proxy
features. They must have been initialized in the Alcatel-Lucent proprietary mode at least once and
sent SIP binaries and parameters to the Communication Server.
1.7 Business continuity of IP Touch in case of IP failure

IP Touch is able to maintain the active communication even if the connection with the
Communication Server fails. The communication is maintained until either user or remote party
hangs up.
When Com Server is lost, the phone is frozen, except for the audio management: user is able to
handle HP+/HP-, mute/un-mute, voice mode switch (Handset, Hands-free, Headset). In addition, an
error message is displayed on the phone screen.
After call completion (hanging-up of called or callee), the IP Touch restarts and registers
automatically to either a PCS (in NOE mode) or an AudioCodes SIP survival gateway (in SIP or SIP
TLS mode).
1.8 Business continuity for out of service Digital/Analog devices

To avoid loosing communication, an automatic immediat forward towards either an immediate
forward number or associated overflow number is proposed when the digital or analog caller devices
is out of service.
1.9 Fallback when SIP Voice Messaging is not reachable

To avoid loosing communication, the calling device rings when it is forwarded (any kind of call
forwarding) to an out of order or unreachable SIP Voicemail (e.g. OT Messaging).
1.10 Dual LAN attachment

The servers, hosting Alcatel-Lucent OmniPCX Enterprise Communication Server or OpenTouch
packages, provide two Ethernet ports to prevent network related problems. If one of these two ports
fails, traffic is automatically routed to the second port so that there is no service interruption.
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Only one Ethernet port is active at a given time. Dual Ethernet does not support:
Load balancing
Split of data stream per application
Dual Ethernet is not compatible with encryption. When encryption is used, one Ethernet port is
connected to the encryption device. The other Ethernet port is not used.
1.11 Uninterruptible power supply

An external UPS provides a battery backed up
supply to keep equipment operating when there
is a power cut and effective protection against
damaging surges.
UPSs are designed for USB compatibility and
include power management software.
A battery pack, an external UPS may be used to
provide power redundancy. In case of a long
power failure, the external UPS sends an alarm
to stop the OpenTouch server, and shutdown the
different Virtual Machines, then the host,
correctly.
This provides sufficient time to stop the system
properly without risking any loss of data.
Note: Only the UPS recommended by Alcatel-
Lucent for the OmniPCX Enterprise are
supported.
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2. Time and date management
Most applications need to retrieve or record reliable time information: the messaging application must
provide date-related data for all types of messages.
For company critical applications such as call centers, messaging systems, or accounting systems,
access to time-sensitive information from a reliable or universal time source is mandatory.
Most company applications, whether related to voice or not, must share timer information provided
by a reliable time source.
Today, the various application components of an information system are usually synchronized on an
independent time source:
The system clock of the server hosting the application provides the time used by the
application.
The communication server is synchronized according to the operating time source (when
available) or PBX internal clock.
Alcatel-Lucent OmniPCX Enterprise Communication Server system administration determines the date
and time used by any feature requesting such data. An internal clock is configured and can be
updated via ISDN or NTP.
In as much as a customer may need different time zones for the same system, different time zones
can be configured according to IP domains (see: module Topology - Distributed Architecture with
Centralized Call Control - Multi-Time Zone).
2.1 System time clock

The OpenTouch server hosts a Network Time Protocol (NTP) server. This server synchronizes
system time with a specified external clock.
Each Virtual machine is automatically configured by the installation wizard to synchronize with the
host's NTP server.
2.2 NTP service on OmniPCX Enterprise components

Time sources can slowly shift and eventually present substantial differences. This leads to
inconsistency when gathering information extracted from various related applications.
The NTP protocol was designed to address this issue and provide reliable timing information for
synchronizing a large set of clients over the Internet.
The Alcatel-Lucent OmniPCX Enterprise Communication Server, as an open platform for voice
applications, supports NTP protocol.
2.2.1 NTP services

An Alcatel-Lucent OmniPCX Enterprise Communication Server node can be an NTP client, which
allows it to obtain time information from one or many NTP server(s).
NTP implementation on Alcatel-Lucent OmniPCX Enterprise Communication Server is based on NTP
distribution v4 and is compatible with NTP v2 and NTP v3 implementation.
The supported services are:
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Activation/deactivation of NTP daemon
Display of the NTP server chain, in order to know the primary time source
Creation/deletion/addition of an NTP server
Broadcast server mode
Authentication mode
The authentication mode supported on Alcatel-Lucent OmniPCX Enterprise Communication Server is
based on the private key method (also known as symmetric key method). Both server and client must
own the same keys. These keys must be included in a file protected against unauthorized access.
2.2.2 Synchronization method

As soon as the NTP daemon is activated, information is exchanged between the Alcatel-Lucent
OmniPCX Enterprise Communication Server node and the NTP server, which permits an estimation of
time differences between the server and the Alcatel-Lucent OmniPCX Enterprise Communication
Server node to be made. If time difference is greater than 128 ms, the Alcatel-Lucent OmniPCX
Enterprise Communication Server internal clock is adjusted to the server time.
When running, the NTP Daemon rejects packets with a time difference greater than 128ms. If the
time difference remains greater than 128 ms for over 900 seconds, the server time is adjusted.
Approximately 2,000 seconds are required to solve a one-second time difference between the server
and the Alcatel-Lucent OmniPCX Enterprise Communication Server node.
2.2.3 NTP and current Alcatel-Lucent OmniPCX Enterprise Communication Server

time setting services
Date and time setting on the Alcatel-Lucent OmniPCX Enterprise Communication Server can be
executed via:
System commands
The attendant console
The information provided on T2 ISDN access (when this service is provided by the public
operator)
When time/date modifications are made, the system clock is updated (step adjustment). The date
and time is updated on the extension display.
When the NTP service is activated, the other time setting services become obsolete.
Restriction
NTP Broadcast service is not supported (restriction on NTP 4.1.1 current distribution).
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3. Additional IP services
3.1 Fax machines

Alcatel-Lucent OmniPCX Enterprise Communication Server is compliant in SIP and H.323 with ITU-
T 38 D Annex recommendations.
Fax relay converts Group 3 fax protocol (T.30 and T.4 standards) into Fax over IP protocol.
External fax calls to (or from) H.323 or SIP terminals use the standard T.38 protocol over the
IP trunk
Fax calls between IP Media Gateways (whether in the same node or not) are performed using
the standard T.38 protocol
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)
3.2 Modem and data transparency
3.2.1 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
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Voice communications are in G711 mode within the IP domain where modem transparency is
activated
3.2.2 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
3.3 Framing
Framing is the transmission period of voice packets in the IP 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 (KBPS) Framing (ms) Bandwidth at IP level
(KBPS)
G711 64 10 96
20 80
30 74.6
G729A 8 10 40
20 24
30 18.6
40 16
G723 6.3 30 17
3.4 Trivial File Transport Protocol (TFTP)

Alcatel-Lucent has implemented specific system mechanisms to simplify management and
maintenance of its solutions. The Trivial File Transport Protocol (TFTP) standard automatically
downloads software files to the IP Touch phones and IP media gateways. This facilitates software
upgrades and changes.
The TFTP Server function downloads a configuration file (lanpbx.cfg) and the IP device software (IP
phones and boards) into flash memory in a binary file format. At each device start (e.g., when
phones are added to the installation), the local processor automatically updates with the latest
software version, when required.
3.5 Dynamic Host Control Protocol (DHCP) server

The integrated Dynamic Host Control Protocol (DHCP) Server automatically assigns the necessary IP
parameters when the IP telephone is installed without requiring system administrator involvement.
The IP telephone receives the network sub-mask and router default IP addresses from the DHCP
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Server. The DHCP Server also supplies the TFTP address, which allows the IP Touch telephone to
receive the downloaded software file.
The DHCP server also provides the media gateways and other IP devices connected to the network
with similar IP parameters. The communication servers PC DHCP server is intended for small sites
where physical integration is a benefit. It is not intended for large enterprises where one of more
external DHCP servers may already be implemented.
The customer can use an existing standard DHCP server or IP Touch telephones can be programmed
with the IP parameters when not in use.
DHCP requests:
The DHCP server can be hosted on:
An existing PC server
The communication server (embedded DHCP server)
The TFTP server used to download the configuration file (lanpbx.cfg) can be hosted on an external
server. However, the TFTP server embedded in the communication server is commonly used. This
entails that, in a duplicated configuration in dual IP subnetworks configuration, there are two
different possible IP addresses for the TFTP server.
The DHCP server embedded in the communication server is designed to deal with this situation: only
the main communication server answers to DHCP requests and it sends its own main IP address as
TFTP server IP address.
If an external DHCP server is used, there are two solutions to deal with duplicated configurations in
dual IP subnetworks:
An external TFTP is used to download the configuration file.
Two TFTP addresses are configured on the DHCP server. This second solution is supported by
Alcatel-Lucent IP Touch 8 series phone sets. The two TFTP IP addresses must be
configured in DHCP option 43 (already used for AVA).
If a DHCP server supports several IP subnets, each subnet must be equipped with a DHCP relay
server to transfer requests to the DHCP server.
October 2013 Offer
3.6 Automatic VLAN Assignment (AVA)
Reorganization is common practice in larger companies. Employees in large enterprises often move
from one geographical location to another due to organizational changes.
In addition to furniture and PC desktop relocation for each move, each employees telephone must
also be moved. This requires TDM PBX reconfiguration, and incurs costly expenses. Multiplied by
thousands of moves per year, moving telephones has become a significant responsibility for telecom
managers in large enterprises.
IP phones can be moved from one location to another without requiring special expertise because IP
phones are plug and play.
However, local area networks have become largely segmented using VLANs (virtual LANs). Changing
VLAN (or an IP subnetwork) requires reconfiguring the IP address. Using a DHCP server can easily
solve this.
For security and traffic reasons, it is normally recommended that IP phones and PCs be placed in
different VLANs. Therefore, major LAN switch vendors require that the IP phones use 802.1Q VLAN
tagging to indicate to the switch in which voice VLAN the phone is located. The voice VLAN often
depends on the geographical location where the IP phone is connected. When the IP phone is
moved, the voice VLAN may change. Manually introducing the new voice VLAN location is not the
ideal solution.
Some vendors have opted for IP phones locating the new voice VLAN from the LAN switch, using
proprietary location protocols. This is not a standard implementation, and may lead to being tied to
one vendor.
Note: In the Alcatel-Lucent networking infrastructure, the switch identifies that the device is an IP
phone (through the MAC address, for example) and transparently places it in the correct VLAN
without requiring action from the IP phone.
Automatic VLAN Assignment (AVA), based on standard procedures, enables IP phones to locate their
voice VLAN ID on the network. With AVA, Alcatel-Lucent OmniPCX Enterprise Communication Server
IP telephones can run on any vendors networking infrastructure.
3.6.1 Different VLAN assignment methods

Automatic VLAN assignment
October 2013 Offer
To simplify the VLAN management, Alcatel-Lucent has implemented a mechanism to affect
automatically the voice VLAN to IP Phones: Automatic VLAN Assignment. This mechanism is based on
double DHCP request
Dynamic VLAN
Most IP terminals support 802.1q tagging, meaning that all traffic originating from the IP phone is
tagged with and 802.1q header allowing the voice traffic to be separated from other traffic types and
placed into dedicated and isolated IP broadcast domains. Many Ethernet switches also have the
ability to identify traffic based on MAC addresses or other unique characteristics. In this way, Alcatel-
Lucent IP phones could be identified by their MAC mask (00:80:9F:xx:xx:xx) and automatically
isolated from other LAN traffic.
VLAN tagging on IP Touch
If the Switch is not able to use Dynamic VLAN, the IP phone must explicitly tag and use IEEE 802.1q
(DHCP, TFTP, signaling and voice are tagged). The explicit tag is managed from the IP Phone.
The VLAN number can be assigned to Alcatel-Lucent 8/9 series sets in three different ways:
Statically on the set
Dynamically through 802.1 AB/ LLDP MED
Dynamically through DHCP
If different values are received or configured on the set, the following priority rule apply.
If the set initializes in dynamic mode, the following priority rule applies:
1 AVA through LLDP
2 AVA through DHCP
3 VLAN configured statically on the set
If the set initializes in static mode, the following priority rule applies:
1 VLAN configured statically on the set
2 AVA through LLDP
3.6.2 Automatic VLAN Assignment (AVA) through 802.1 AB/ LLDP MED
AVA through 802.1 AB/ LLDP MED allows to assign the VLAN number as soon as the Alcatel-Lucent
8/9 series set connects to the switch, before IP initialization.
AVA through LLDP is available whether the set initializes in static or dynamic mode.
If needed (for example due to compatibility issue with the switch), AVA through LLDP can be
deactivated on the set.
AVA through LLDP is not available for the PC connected on the VLAN port of the set.
3.6.3 Automatic VLAN Assignment (AVA) through DHCP

The DHCP protocol allows to receive both voice VLAN ID and IP configuration (AVA uses the vendor
specific information DHCP option 43).
Therefore, AVA can be supported on the OmniPCX Enterprise embedded DHCP server but also on any
external DHCP server.
October 2013 Offer
The AVA process is as follows:
When the IP phone is connected, it requests a VLAN ID through a DHCP request.
The DHCP server responds to the IP phone DHCP request. Based on the value of the "data"
VLAN ID received (ipaddr field completed by the virtual router), the DHCP server sends
the IP phone the "voice" VLAN ID associated with the "data" VLAN ID received.
o With an Alcatel-Lucent OmniPCX Enterprise Communication Server CS DHCP server,
the IP address sent is identical for all VLAN ID requests. This address is configured
specifically for this purpose in sub network parameters
o With an external DHCP server, the IP address sent is one of the addresses available in
the configured range
The IP phone sends another DHCP request to complete IP configuration (this request is
tagged, the VLAN ID received previously is used).
3.6.3.1 Prerequisites
AVA compatibility
October 2013 Offer
AVA is compatible with switches supporting untagged and 802.1Q tagged frames on the same port
and with routers supporting DHCP relay.
Network design restrictions
When non Alcatel-Lucent switches are used, there must be at least one source IP network per
assigned voice VLAN. The VLAN request (using DHCP frames) comes from an IP sub network and
only one VLAN can be configured and assigned per IP range.
DHCP compatibility
Alcatel-Lucent has validated some of the more used DHCP servers on the market, such as: WIN 2003,
ISC DHCP server, used by all the UNIX/LINUX DHCP servers, and the Alcatel-Lucent Vital QIP DHCP
server.
3.7 VoIP supervision

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.).
They are delivered in real-time and can be sent to an external machine for further supervision quality.
See the AAPP program for more information.
Note: See the document on the OmniVista 8770 management platform for more information
October 2013 Offer
4. General call routing
The Alcatel-Lucent OmniPCX Enterprise Communication Server uses advanced routing algorithms to
distribute incoming calls according to the varying needs of a standalone or network configuration. Call
distribution is based on entity and routing tables and is ideally suited to direct inward dialing (DID)
call distribution.
The general call distribution algorithm is described in the following diagram:
Incoming DID calls

Incoming DID (direct inward dialed) calls can be routed to one of the following entities: a directory
number representing an attendant group, an attendant station (may or may not belong to the same
group), a logical entity, a designated phone, or a hunt group. A time-based routing table is
associated with the attendant group or station and with the entity. This enables call management
according to vector user availability. If a phone/hunt group is not answered or is busy, the call is
overflowed to a logical entity.
Incoming NDID calls
NDID (non-direct inward dialed) calls can be routed to a trunk or to a trunk group associated with a
directory number representing an attendant group, an attendant station (may or may not belong to
the same group), a logical entity, a designated set, or a hunt group. A trunk group may be also
routed directly to a logical entity.
4.1 Class of Service (CoS)

All attendant groups, devices, and entities are assigned a traffic routing class of service (COS)
category. Different call flow management scenarios and priorities may be defined based on the class
of traffic routing COS. The various traffic routing classes of services are:
Attendant group class of traffic
o Attendant group directory number
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o Non-dialing trunks (public/private) group callback
o Attendant group transfer
Individual attendant traffic COS
o Attendant directory number
o Non-dialing trunks (public/private)
o Chained calls
o Charging callbacks
o Busy phone on-hook for transfer with announcement
o Attendant callback
o Attendant transfer
o Trunk callback on mishandling
Entity traffic COS
o Entity call (internal/DID)
o Non-dialing trunk groups/trunks (public/private)
o General attendant call
o Entity callback
o Unanswered DID call
o Wrong DID number (entity 0)
o Trunk callback on mishandling
4.2 Routing table

In the following table, the four horizontal entries (Day, Fwd1, Fwd2, and Night) define the routing
vector destination according to the system status. Vertical entries from top to bottom (Group 1-3,
Attendant 4-5, Night phone 1) indicate the available vector destinations resulting from call overflow
(busy/no answer, attendant out-of-service). The lowest vertical entry field (Night set 1) is the final
possible vector destination. If previous routing attempts fail, this is how the call is routed.
October 2013 Offer
4.2.1 Time and Class of Service (COS) tables
Each routing table is associated with a time and a class of service (COS) table. The routing tables
are:
A weekly table enabling the system status to be changed automatically (maximum of four
changes a day) or enabling the attendant group/station to perform status changes.
A time dependent class of service table granting/restricting access to external lines. The first
figures of the dialed number are either authorized or unauthorized for a specific class of
service.
4.2.2 Routing in the network

A group of attendants belongs to a node. Several attendant groups can co-exist simultaneously on
the network. An entity can be distributed over several nodes. The node handling the incoming trunks
or trunk groups (source node) performs the routing. A status change for a network entity is
automatically sent to all concerned nodes.
4.2.3 Night service

An attendant, attendant group, and entity can have either a common night service phone or their
own night service anywhere in the network.
4.3 Call restriction for alarms and emergencies

Some businesses and organizations find it important during a crisis to be able to restrict certain
employees from making outside calls. For example, during a pollution alert in a chemical plant,
managers may want to restrict employees from being able to make calls to prevent premature or
incorrect news leaks. For these types of situations, Alcatel-Lucent offers call restriction features.
Call restriction features allow the gradual restriction of telephone calls between the Alcatel-Lucent
OmniPCX Enterprise Communication Server and the outside world at times of heightened sensitivity
and confidentiality. Each network of OmniPCX systems maintains its own alarm level that can be
increased dynamically via the physical alarm equipment, via an attendant console, or via the
management interface. Only calls with an alarm class of service greater than or equal to the general
alarm level can be established.
There are two types of call restriction, exclusive of each other: Ivory Tower Tap and minimize.
Ivory Tower Tap
Disconnects external calls in-progress
Blocks new outgoing external calls
Rejects incoming external calls
These steps completely isolate calls, a state referred to as: Ivory Tower Tap.
Minimize
Blocks new outgoing external calls
Minimize is a functional sub-set of Ivory Tower Tap.
The type of call restriction is determined by management during system installation.
Call restriction management
October 2013 Offer
The alarm level can be managed from the:
Alarm equipment by dialing the call restriction prefix
Attendant console
OmniVista 8770 management platform
System limits
Total number of alarm levels is 11 (0 10 inclusive).
Total number of call restriction classes of service: 11 (0 10 inclusive).
4.4 Priority calls Multi-Level Precedence and Pre-emption (MLPP)

In a crisis situation, selected calls must get through, whatever the situation (busy lines or not enough
idle resources available). In such cases, the voice communication system provides two types of
functionality:
Voice call priority
Seizure of resources already in use by others
These two features are part of the Multi-Level Precedence and Pre-emption (MLPP) service.
The MLPP service provides prioritized call handling service in two different ways:
Precedence involves assigning a priority to a call.
Pre-emption involves seizing resources already in use by a lower precedence call user if there
is a shortage of idle resources.
Implicit and explicit priority
The priority assigned to a call is always the calling partys priority.
By default, the implicit priority is used without any activation at the time of call setup.
Dialing the explicit priority prefix before dialing the called partys number activates the explicit
priority.
Pre-emption
Priority calls with an activation level equal to 3 pre-empt communications that are already under way
(established or not) according to the rules and in the following order of preference:
Calls with activation = 0
by selecting the one with the lowest level of priority
Calls with activation = 1 or 3
by selecting the one with the lowest level of priority, if the call priority level is greater than
that of this call
Pre-emption is applied in call progression and on busy telephones.
Standard compliance
The MLPP implementation is compliant with the I 253.3 CCITT standard.
Product limits
Feature available for Standard Users
Number of priority levels for each user: 16
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4.5 Call accounting (metering)
To optimize, control, and manage telecommunications expenses the Alcatel-Lucent OmniPCX
Enterprise Communication Server monitors and stores a call accounting log for all calls. The data is
stored by a call accounting proxy function that can be accessed and processed with an external
Alcatel-Lucent management application or a third party call accounting application.
The Alcatel-Lucent OmniPCX Enterprise Communication Server generates a ticket or Call Detail
Record (CDR) for each call. This ticket is generated from:
Information from the network operator/carrier
A carrier cost simulation generated from:
o The number dialed
o Call duration
o The date and time of the call carrier
o Tariffs tables
Tickets (Call Detail Records) are processed by:
o Financial Report, an Alcatel-Lucent OmniPCX Enterprise Communication Server internal
application
o Real-time output to a printer
o Real-time output on Ethernet, which allows an external application integrating
telephone costs to be created
October 2013 Offer
5. Business Telephony routing for branch offices
Business Contact Standard Edition is a packaged solution offering an intelligent business call routing for branch
offices, using standard telephony, upstream of a contact center. Based on the OmniPCX Enterprise, Business
Contact Standard Edition was created for services companies organized in branches that wish to personalize contact
with their customers and offer a consistent welcome across their entire organization. Customers are sent straight
through to their preferred advisor or routed to the next best available profile. Irrespective of where the customer
call comes from, it will always reach the right advisor, rather than a contact center or an IVR. This direct contact
with the right person can radically reduce abandoned calls estimated to be up to 20% and accounting for 4 million
loss per year in financial organizations, according to Gartner.
By always connecting the customer to the right advisor, irrespective of where the customer is calling from, Business
Contact drastically reduces lost calls and enhances business calls and advisors activity insight through call tracking.
This directly impacts direct touch activities by creating long-lasting customer-advisor relationships.
As it is using the powerful Business Contact Engine, further customization is easily possible from Alcatel-Lucent
Professional Services (upon request), to adapt to the customer organizational and business structure, job profiles,
opening hours, and more. Business Contact Standard Edition is easy and fast to deploy; it includes predefined
routing strategies for 50 Business Contact users.
Branch
Office
Preferred
assistants
Personal 1
advisor
busy*
Visit Card
2
Direct call
Customer 3 Available
specialists
Customer calls his personal
advisor which responds if available. * Busy / No answer / OOS
If he is unavailable, then route to
any one in the branch office according
to business rules & skills Route to
1 routing towards an assistant
2 routing towards a job profile

Contact Attendant Voicemail
Two choice menu or center
3
Final route (e.g. Contact Center)
Key features
Business Contact Unified engine with built-in predefined strategies:
- Configurable to adapt to job profiles
- Integrated with calendar for branch office opening hours management
- Adaptable to organization structure (branch office)
Complements the OmniPCX Enterprise business telephony services
Centralized performance measurement through call detail records
Existing strategies can easily be replicated into new business units or branches
High availability (optional)
Customizations to meet specific needs (optional)
October 2013 Offer
Benefits
Ensure unified welcome across a distributed business organization
Easy to manage and use
Fits perfectly in your existing Alcatel-Lucent IP architectures
Improves customer satisfaction by drastically reducing lost calls
Helps better measure advisors performance
Prerequisites
Hardware
Quad core processor 2,0 GHz, 4 GB of RAM, 80 GB of hard disk space
The hardware platform is not provided
Check that your platform is a Red Hat Enterprise Linux certified platform
Only 32-bit and 64-bit editions of RHEL are supported
Licenses
Red Hat Enterprise Linux R4 u8 or R6
Latest supported OmniPCX Enterprise release
Required OmniPCX Enterprise elements (not included)
- 1x RSI Routing Services Interface Server
- 1x TSAPI server
Limits and restrictions

No CTI monitoring of the Business Contact Standard Edition users and thus no screen pop-up or CTI integration
available in R1.0
Please refer to the Technical Datasheet posted on the Enterprise Business Portal for more details on the
application prerequisites, limits and restrictions.
October 2013 Offer
6. Integrated accounting
Decreasing telecommunication costs and providing better service to customers are constant concerns
for every company. This is just as true for telephone usage as for any other expense; telephone costs
must be controlled and efficiency measured. The OmniVista 8770 offers a way to analyze their
telecommunication costs as well as billing for the different cost centers or users, to all companies
operating on one or several sites.
The telephone application generates an accounting call record or ticket for any call (whether
outgoing or incoming, private or public). The accounting call record contains information on the
corresponding call (duration, date, services used, number called/dialed, network topology, etc.) The
integrated accounting application collects records for various processing operations to be applied
according to user requirements. Processing operations are performed by:
Internal applications: for example, Financial Reports or Hotel/Hospital Management.
(detailed elsewhere in this document)
External applications: for example, an accounting program using a dedicated database
(OmniVista 8770, detailed elsewhere in this document) or other remote interface.
The Alcatel-Lucent OmniPCX Office Communication Server integrated accounting application offers
the following main features:
External accounting
Call Record details
Traffic analysis
Financial reporting
October 2013 Offer
Application configuration enables the user to:
1 View of user counters (on some types of user sets and attendant stations).
2 Produce real-time output either to a V24 port (optional) or an Ethernet connection (optional).
3 Printout traffic analysis records (optional).
4 Printout financial reports (optional).
6.1 Configuring the application

The cost of a call may be calculated either by using the number of charge units transmitted by the
carrier or by the internal configuration of accounting based on call duration and destination.
6.2 User counters view

The attendant or the user (subscriber) may view the OmniPCX Enterprise Communication Server
internal counters.
Three counters are assigned to each user:
Total number of charge units/cost since the last reset
October 2013 Offer
Number of charge units/cost of the last call
Number of outgoing calls since the last reset
The unit used by counters for cost or charge units is configured in system management. It is the
same for all sets in the installation.
The attendant can view all user counters and reset individual counters or all counters. Users with a
set with a display (Alcatel 4034, 4023, 4012, and 4035) can view their own counters:
Total charge units/cost since last reset
Number of charge units/cost of the last call and date of the last reset (only on Alcatel 4034,
4023, 4035 sets)
The user accesses this feature via a prefix.
6.3 Real time records printing

Records are printed out by a printer connected to the serial port of the Communication Server or
output via an Ethernet connection.
A filter can be used to limit the number of records printed, according to:
Call type: incoming, outgoing, private, public, priority, data, etc.
User type: justified (charged), not charged
6.3.1 Ethernet
Selecting record output via Ethernet means that records cannot be sent or received via a V24 link.
This transmission support has a buffer capacity for 2,500 records and requires a client application
which is compliant with OmniPCX Enterprise Communication Server protocol to be developed. The
transmitted records contain all available information.
6.3.2 V24 support

If the records are printed for users not charged, the following information is masked:
Justified (charged) user number
The number of the set that transferred the call to the correct user
Number dialed
User name
The record contains less data than on an Ethernet link and is printed as a text file in one of two
formats:
Reduced format (1 line)
Extended format (2 lines)
The information contained in these two formats is:
Fields Extended format Reduced format
Justified (charged) user directory number X X
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Fields Extended format Reduced format
Justified (charged) user name X
Project number (business account code) or PIN (1)

X
Call end date X
Justified (charged) cost center X X
External carrier used for the call X
Trunk group number X
Trunk node number X
Justified (charged) user node number X
Call type (voice, data) X X
Directory number of the set that transferred the X X

call
Trunk number X
Number sent/received (2)

X X
Call end time X X
Call duration X X
Number of charge units or call cost X X
Project (business account) number (0 if not X X

applicable)
Call type (personal/professional) X X
Internal PCX services used during the call (3)

X
Call type (incoming, outgoing, private, etc.) X X

Notes:
(1) (2) Masking
PIN may be masked for personal calls. The final digits of transmitted numbers may be masked. The
number of digits masked may be configured according to call type: personal calls, business (project)
calls, normal calls.
(3) Internal services that may be used include VPN, ISVPN, ARS, speed dialing, remote charge, text
messages, etc.
October 2013 Offer
6.4 Printing traffic analysis records (optional)
One or more user types or specific call types may be analyzed via a real-time printout of records for
the concerned calls. Printout is via a printer connected to the V24 port of the PCX. Filtering may be
configured and modified by the attendant or done in system management. Filtering is performed
according to the following criteria:
By user (10 users maximum)
By number of charge units or cost threshold
By duration threshold
By trunk group (2 maximum)
By dialing (prefix or number transmitted) (10 maximum)
By cost center (2 maximum)
By project (business account) number (10 maximum)
The above criteria may be combined to refine analysis.
The record is output at the end of the call. If the printer is not available, messages are stored in a
buffer (max. size, 500 messages) and an incident is generated by the OmniPCX. When this threshold
is exceeded, subsequent messages are lost.
Records are identical to those output in real time on a V24 port. The user may select one of two
formats, reduced or extended.
Storage buffer
If the output support is not available, messages are stored in a buffer with a maximum capacity of
500 messages. When this capacity is exceeded, the OmniPCX Enterprise generates an incident report.
6.5 Printing financial reports (optional)

Financial reports use the accounting records contained in the SQL database on the OmniPCX hard
drive to provide cost distribution for a specified period by:
User
Cost center
Trunk and trunk group
Project (Business account) number
Only calls with a non-zero number of charge units are considered in the financial report.
Output on a V24 port of the OmniPCX is automatically triggered at the end of the selected period of
analysis. The period of analysis may be configured on the attendant station or in system
management.
Generating a financial report deletes data stored previously in the database (used only to print these
reports).
Records used by external accounting applications are maintained.
This feature cannot be used in an OmniPCX Enterprise Hospitality configuration.
A time period is determined by configuring a start and end date. The time period that follows is
automatically determined by the previous period. If the date is the same, the number of months
October 2013 Offer
between the two dates is used to calculate the new time period. Otherwise, the number of days in
the previous period is used to determine the dates for the new period.
Current period New period
01/01/2005-02/01/2005 02/01/2005-03/01/2006
01/01/2005-01/10/2005 01/10/2005-01/19/2005
Financial reports are divided into four sections:
1 User counters
For each user, the counters are:
Number of calls
Total call cost
Cost per area (five areas may be specified in system management, each representing one or
more public translator areas)
Personal call costs
Note: If the records of users not charged are saved in the database to generate financial reports, the
number of the justified (charged) users is masked when the report is saved or printed.
2 Trunk counters
For each trunk, the counters are:
Number of calls
Total call cost
Total cost per trunk group
3 Project (business account) counters
For each project (business account) number, the counters are:
Number of calls
Total call cost
Total call duration
4 Global counters
For the entire installation (system):
Number of calls
Total cost
Cost for area 1
Cost for area 2
Cost for area 3
Cost for area 4
Cost for area 5
Personal call costs
October 2013 Offer
7. OmniPCX Record
The Alcatel-Lucent
Lucent OmniPCX RECORD Suite offers small and medium medium-sized
sized businesses a complete
tracking solution for customer interactions through uniqu
uniquee call recording, screen capture and
coaching capabilities. Flexible, simple to install and user
user-friendly,
friendly, it seamlessly integrates into both
new and existing OpenTouch Server erver environments.
Key features
Modular offer
- RECORD: rich recording facilities of inbound and
outbound calls in multiple audio formats (MP3,
GSM6.10, WAV)
- SCREEN CAPTURE: complete user desktop activity capture
- SILENT MONITOR: remote and discrete monitoring
- QUALITY MONITOR:: evaluation of recorded employees and instant coaching sessions
Architecture
- Call recording in SIP (declared as SIP SEPLOS)
SEPLOS),, VoIP, analog, digital and mixed environments
- Web-enabled
enabled architecture that makes it easy to locate and use recordings
- Multiple language web interface
- Support of multi-node environmentss
- High availability
- Support of virtualization
- Open integrations through the Application Programming Interface (API)
Compliance
- Records encryption
- Login authentication via Radiuss server
- Compliance with Thales encryption
- In line with Payment Cards Industry (PCI) requirements for call recording systems
Benefits
Fulfills every need with a modular offer made up of four modules (Record, Capture, Silent Monitor and Quality
Monitor)
Accelerates
ccelerates resolution of customer issues
Minimizes risk of disputes through complete tracking of customer interactions
Boosts customer satisfaction and loyalty
Enhances employee productivity thanks to monitoring and coaching
ENT_MLE_015926 ed 01Chapter
Chapter 03: System Services 38/75 Alcatel-Lucent
Lucent Enterprise Communication Solutions
October 2013 Offer
Improves quality assessment based on actual customer-employee interaction recordings
Transparently integrates with Alcatel-Lucent Communication Servers
Minimal hardware requirement (single server)
Prerequisites
IPDRLINK or DRLINK Recording licenses
For VoIP systems, QoS must be implemented
Dedicated server with Windows 2008 Web Server Edition R2 (64bit) in English with permanent licenses
OmniPCX Enterprise or OpenTouch (standard users): latest supported release (please refer to the product
Release Lifecycle Policy document)
Limits and restrictions

400 units (400 simultaneous call recording units in IPDR-Link architectures, or 270 in DR-link architectures or up
to 100 simultaneous screen capture sessions)
Up to 400 Quality Monitor clients
Maximum of 5 simultaneous supervisor silent monitoring sessions
The OmniPCX RECORD can be installed on a virtual server; however, when installed in a virtual environment,
some limitations apply as follows:
- The solution will only support a maximum of 140 simultaneous call recording units in IPDR-Link architectures
or 90 simultaneous non IP
- The IP Attendant is not supported
IP Attendant limitations:
- Only supported sets are 4059EE
- Screen Capture not available
- Record On Demand will not work with IP Attendant Console handsets
- Recording filters can only be set using Caller or Called number
- High Availability Warm Standby is not offered for IP Attendant Recording
- Hold Time is not recorded when an IP Attendant call is put on hold
- Encryption not offered for IP Attendant
One single OmniPCX RECORD (or other voice recorder) per Alcatel-Lucent OmniPCX Enterprise
More details
Please refer to the Technical Datasheet posted on the Enterprise Business Portal for more details on the
application prerequisites, limits and restrictions. Technical requirements are described into details in the
"OmniPCX RECORD - Hardware & Software Specification.pdf document available on the Enterprise
Business Portal (under Customer support/Technical Support/Software Download/OmniPCX Record Suite
folder).
Please refer to the Services Applications Compatibility Matrix available on the Alcatel-Lucent
Enterprise Business Portal for all questions about software compatibility.
October 2013 Offer
8. Integrated voice guide services
8.1 Integrated voice guides

The Alcatel-Lucent OmniPCX Enterprise Communication Server includes recordings of human voice
guides to provide users with information on the various options available to them during a telephone
call. Voice guides give verbal information about the current status of the phone and offer direct
access to appropriate features. These step-by-step instructions are available on any phone, ensuring
easy use of all features and helping to avoid errors. Voice guides operate as dialing tones: the user
can start dialing at any time.
Voice guides are available in all languages as a standard feature (with a female or male voice,
depending on customer choice). They can be broadcasted to internal and external users alike.
Customization of voice guides, as well as creation of specific voice guides is made available to the
customer as a standard feature. You may record your own voice guides, either mimicking the
standard recordings and/or providing any information you deem relevant. Recordings can take place
in a professional studio, but you can greatly reduce your costs by using the Alcatel-Lucent Audio
Station application to record your own messages with a high professional sound quality or using an
Alcatel-Lucent 8/9 series phone. Customized files are easily integrated to the system to reflect your
company image.
A standard (free of rights) music-on-hold file is included on the voice guide CD-ROM delivered with
the Alcatel-Lucent OmniPCX Enterprise Communication Server. You may use this file, customize it
with voice messages, or play the music file of your choice, unless you prefer to connect another
device to the server.
The voice guide feature also includes: wake up calls, off-hook messages and greetings.
Pre-recorded standard messages are available to greet incoming calls. According to configuration,
when the Alcatel-Lucent OmniPCX Enterprise takes an incoming call, a pre-programmed message is
played before the caller is diverted to a directory number.
Greeting messages are stored in the system (maximum 254 messages)
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Messages are selected according to user requirements
The chosen greetings can apply to individual phones or to all extension phones, or can be
dependent on the recipients status
For more information on specific greeting use, see the automated attendant section in the document
on attendants.
8.2 Voice Guides

An economic and effective method of avoiding client dissatisfaction in any business environment is
the possibility to interact instantly with a caller.
The Alcatel-Lucent OmniPCX Enterprise Communication Server voice guides services are available to
increase and maintain communication.
From outside the enterprise, voice guides can be used to encourage the caller to:
Stay on line or call back later
Refine their search within an enterprise
Offer an interactive action
Within the enterprise, the system voice guides offer appropriate information for users analog, digital
and DECT telephones or external users (via AA or DISA). Voice prompts offer the following types of
information on:
Context-sensitive feature codes (suffixes and prefixes)
Telephone status
An activated service
A voice guide offers relevant information at specific times. The voice guide is made up of one or more
voice messages, which are relevant messages recorded in different languages.
The voice message services that can be offered via the Alcatel-Lucent OmniPCX Enterprise
Communication Server include:
Static: cannot be modified by the user:
Generic: compatible with generic numbering plans and offered in at least 6 different
languages for all countries.
Note 2: A static message file contains one or more messages.
Standard: adapted to a country with special requirements and/or a numbering plan differing
from the generic numbering plan.
Specific: to an area or company.
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Dynamic: modifiable, recorded by users and/or managers via a phone set or the AAS
application.
Note 3: A dynamic message file only contains one message.
External: music-on-hold (in most cases) played by a device external to the OmniPCX.
8.2.1 Chained voice guides

Some voice guides are composed of several messages. These are referred to as "chained" voice
guides. There are three different types of chained voice guide:
1 Variable chained voice guides: composed of several messages (8 maximum), of which at
least one varies.
Example 1: When programming an appointment reminder, the user programs the time at which he
wants to be called on his set. The set then confirms "Your request for an appointment at XX hour(s)
YY minute(s) has been recorded (where "XX" and "YY" are variables).
2 Static chained voice guides: composed of several non variable messages played in a pre-
configured order (sequence).
3 Composite chained voice guides: composed of an internal message (static or dynamic)
followed by an external voice message.
Example 2: The message: "Welcome to Alcatel-Lucent, please hold the line, we're trying to connect
you" is played before real-time broadcast of a radio station.
8.2.2 Voice messages

Voice messages:
Provide high quality recreation and playback of the spoken word using binary information
contained in the memory module
Voice messages in different languages are delivered on a CD-ROM along with a user-friendly
software transfer tool
Feature dialing codes (prefixes and suffixes) are implemented using a standard or customized
dialing (numbering) plan. The voice messages installed on the COM Server by the transfer
tool adhere to the country or company dialing plan
After installation, voice messages are stored on the COM Server hard disk, downloaded to the
media gateway memory, and played on the telephone by the media gateway
Voice messages can consist of one simple message or a combination of several messages (e.g., a
combination of a greeting message and a short musical excerpt).
8.2.3 Message examples
Thank you - Your request has been recorded; you may now hang up.
Please dial your password or personal code.
The number you have dialed is not authorized; for more information, please ask the system
manager. Thank you.
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Please dial the phone number you require.
Your calls will be forwarded. You can still make a call. To cancel forwarding, please dial....
You have a voice mail message.
You can forward to a partys voice mail by dialing 7.
You may cancel automatic callback by dialing 6.
You may request automatic callback by dialing 5.
You may request barge-in by dialing 4 or automatic callback by dialing 5.
You may request automatic callback by dialing 5 or camp-on by dialing 6.
You may request barge-in by dialing 4, automatic callback by dialing 5, or camp-on by dialing 6.
You may request barge-in by dialing 4.
Please dial the project account code.
Please enter the required time of day (alarm or appointment reminder call).
You may camp-on by dialing 6.
The person you have called is unavailable; please call back later. Thank you.
While your extension is in do not disturb mode, you can still make a call. To cancel do not disturb,
please dial...
This is your reminder call (alarm or appointment reminder).
The extension you have called is protected against barge-in.
Please do not hang up; your party is being paged (paging).
You may page the person you have called by dialing 7.
Your party is being paged; you may hang up (paging).
8.2.4 Main characteristics

Characteristics include:
Voice guide and display languages can be selected by the user on Alcatel-Lucent phones from
those available in the installation
Modification of the user language automatically synchronizes voice prompts and display as
well as the Alcatel-Lucent 4645 e-VA voice mail
The use of voice guides is optional and can be assigned by each user individually
Depending on the type of broadcast selected by management, the user will hear the voice
guide from the start or while it is being played. If playback is set from the start of the voice
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guide, each time a user requests this guide and a broadcast channel is free, the message is
played from the start
If the user requests a guide that is not playing and all guide broadcast channels are busy, the
user hears the backup tone for the guide requested
Voice guides cover different features (all described in other documents):
Telephone services
Access restrictions
DISA
Hotel, hospital
CCD
Integrated voice mail
Integrated automated attendant
8.2.5 Customized voice messages

Alcatel-Lucent offers the ability to manage voice guides professionally on site according to individual
customers' needs. Customized voice guides can be recorded on or off site and easily integrated into
the OmniPCX using the Alcatel-Lucent Audio Station (AAS), an adjunct system that offers professional
quality results:
Input can be digital, CD-audio, or CD-ROM
Original recording can be done by a professional studio
8.3 Internal Music-on-hold

This internal component plays messages and music to callers on hold.
Characteristics include:
The message is recorded in the voice guide module
The content of the announcement can be customized
A default Alcatel-Lucent music-on-hold recording is included on the voice guide CD-ROM
A centralized configuration in a large network can require specific music on hold (MOH) for each
branch office (e.g. for small bank agencies).
As with other resources, most of these MOHs are centralized in central sites and can be used by
remote IP Phones hosted in branch offices without a Media Gateway.
MOH features are stored in the GPA, GA or GD boards. Each board supports one memory with the
capacity for dynamic guides of 16 minutes (GPA2) or 8 minutes (GD/GA).
The duration can be extended in large multi-site configurations where it is possible to configure
several thousand branches per Communication Server with a specific MOH for each branch.
System optimization can be increased for the following durations per board:
GPA2: 64 minutes (16 in dynamic memory + 48 in static memory)
GD/GA: 40 minutes (shared with voice guides)
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8.4 External Music-on-hold
Messages and music for callers on hold can be played from an external device (tape recorder, radio,
CD player, etc.)
8.4.1 Distinctive features

Distinctive features include:
Customization of the recorded message
Variation of the duration of the announcement according to the capability of the external
device
8.4.2 Connection
A standard telephone pair connects the external device to an analog board or to the AFU
daughterboard socket on the GD (Gateway Driver) board.
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9. IP Telephony domains and related features
The Alcatel-Lucent OmniPCX Enterprise Communication Server offers several features:
IP connectivity, Signalling Control and Media Management for several IP Devices and Media
Gateways based on IPV4 and IPV6 ready for future proof
Enabling to cope with inherent bandwidth constraints on the WAN between remote sites
Offering defense mechanisms in case of IP link breakdown between a remote site and the
central site where the Com Server is located
Enabling to take into account the geographical location of a remote site
Providing Routing mechanism and Rules for both TDM and IP connectivities also management
and registration for SIP Proxy and translation of E.164 into IP.
These features are based on IP telephony domains, which enable to partition a node according to IP
address of IP devices.
An IP domain corresponds to one or several ranges of IP addresses. An IP telephony domain contains
IP devices whose IP addresses belong to ranges of IP addresses defined for this domain. An IP
telephony domain also contains non-IP devices connected to the media gateways belonging to this
domain.
Typically, an IP telephony domain corresponds to a remote site.
Note: The IP telephony domain feature is a "local" concept. In other words, IP domains are local to
each Communication Server. An IP domain cannot be defined across two or more IP communication
servers.
9.1 Call Admission Control (CAC) and codec selection

The original key role of IP telephony domains is to prevent voice congestion in the WAN by limiting
the number of calls on the WAN and by imposing the use of compression.
IP telephony domains allow to take into account the available bandwidth by limiting the number of
inter-domains communications, whereas there is no limit on intra-domain communications. The
limitation of inter-domain communication is "Call Admission Control" (CAC). CAC applies to all IP
communications, whether they involve a media gateway, IP phone, H.323 terminal, or SIP terminal.
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IP telephony domains allow the possibility to apply different voice compression/digitalization
algorithms for inter-domain and intra-domain communications. Typically, internal calls (i.e., limited to
a single location) are based on the G.711 uncompressed PCM standard, but WAN calls outside the
location use the G.723.1 or G.729/A compression algorithm.
CAC can take into account fax communications in the limitation of inter-domain communication: a
distinction is made between the voice flow rates and fax flow rates. IP domains mixing IP phones and
fax machines must be separated in two IP domains:
A primary IP domain (D1) including IP phones and IP boards not used for fax communications
A secondary IP domain (D2) including
o IP boards used for fax communications
o Other IP devices for which the CAC must apply in the same way as for fax machines
CAC allows to:
Define the total bandwidth available for fax and voice communications
Define the bandwidth for fax communications
Take into account the ratio between voice flow rate and fax flow rate in order to optimize the
use of the available bandwidth
This is useful when voice communications use a compression algorithm (e.g. G.723/G.729) whereas
fax communications use G.711. In this manner, CAC allows to limit the number of voice
communications when fax communications are already established, in order to guarantee a sufficient
bandwidth for existing communications. In the same way, CAC allows to reject fax communications
when the existing bandwidth is insufficient.
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Controlling the bandwidth is possible for other devices (for example, OmniTouch 8600 MIC Desktop).
9.2 Defense mechanisms

Defense mechanisms include:
The backup signaling link, which enables to set up a signaling link with the Com Server
through the Public network in case of IP link breakdown.
The Passive Communication Server, which acts as a Com Server for one or several remote
sites when the Com Server is not reachable, either because of the Com Server breakdown or
IP link breakdown.
IP phone survivability, which enable IP phone in a remote site to work (with limited services)
when the IP link with the Com Server is down or the Com Server itself is down. In
survivability mode, an IP phone uses the backup signaling link or connects to a Passive
Communication Server or to a SIP proxy server: in the latter case, the IP phone switches to
SIP mode.
For more information on these mechanisms, refer to module More Information - Architectural
Reliability.
9.3 Multi-time zone

The Alcatel-Lucent OmniPCX Enterprise Communication Server system allows to set up local time
zones for each IP domain. This enables sets to display the real time of the site. The date format may
be either in the order day-month-year, or in the order year-month-day.
This data is updated by ISDN or NTP and changes automatically with Daylight Saving Times. It is
displayed on mobile or fixed digital sets (whether IP or TDM) and attendant sets (for example,
Alcatel-Lucent 4059 IP attendant console). It applies to all user time-related features such as wake-
up and appointment calls, unanswered call time, deposit time for text messages, charging
(accounting tickets, according to the correct time zone), and is also updated in these features'
corresponding voice guides. It is also compatible with DECT and MIPT roaming.
The Alcatel-Lucent OmniTouch 8440 Messaging Software records messages with the local date and
time of the receiving set. The voice mail allows mailbox owners to select their present time zone from
a Web interface. This operation guarantees that the voice mail announces the date and time of
messages received by the mailbox owners according to their time zone.
Restrictions:
If an IP domain includes SIP or S0 sets:
SIP sets display their own date and time
S0 sets display the system date and time
9.4 Multi-country
IP telephony domains can be used to associate specific countries to IP devices. In addition, the
Alcatel-Lucent OmniPCX Enterprise Communication Server supports configurations where the remote
sites are located in different countries.
The multi-country mode is used for:
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Tones setup. The tone generation definition of the corresponding country is sent by the Com
Server to the Media Gateway at Media Gateway initialization. For IP Phones, the Com Server
sends the tone generation definition of the corresponding country to the set.
When a tone is sent to a device, the Com Server checks the country declaration of this piece
of equipment and sends the corresponding tone parameters in a specific tone message.
Call back service. An External Call Back Translation Table is used for incoming calls.It converts
the number of the external caller presented by the public carrier.The number thus translated
is the exact number a user/system must redial in case of callback.
The translated number is:
Displayed as caller number on digital sets with a display
When applicable, added to the list of unanswered ISDN calls on a digital set with a
display
Used to call back the calling set
Stored in a call log
In case of multi-country configuration, A call can enter the system via a trunk group located in
one country and be internally routed to the called set located in another country. Regardless
of called set location, the calling number displayed must be the number to dial to call the
caller back, in other words:
The outgoing prefix must be the prefix used in the country of the called set
If the calling set is located in a foreign country, the international access prefix and the
national code must been added
Common Hardware boards initialization and other pieces of equipment initialization (IP
devices and devices linked to a Common Hardware board).
The multi-country mode affects:
Calls to a forwarded set not located in the same country as the caller. The calling number is
built with the directory number of the forwarded set
Calls to a remote extension (for example, mobile phone) not located in the same country as
the caller. The calling number is built with the directory number of the remote extension
combined with additional parameters
9.5 Calling line identification (CLI) per IP Domain

IP telephony domains enable to define a specific installation number corresponding to a local number
of the remote site. This number is mainly used as CLI number for outgoing calls from non-DID sets.
This is particularly useful for outgoing emergency calls. It prevents the emergency center from being
misled by a CLI number which would not correspond to the real location of the calling party.
For emergency numbers, a specific CLI per IP domain, allows emergency services to call back the
user (or any other individual in the same geographical location) having called them in the first place.
This feature is available for PSTN, H.323, and SIP interfaces.
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10. SIP and web services
Alcatel-Lucent has chosen SIP as one of the corner stones of its standards-based developments for
enterprises interactive 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:
o It belongs to the web protocol family: XML, HTML, MIME and others
o It is directed by the IETF standards body
SIP is 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 fueled 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:
o End-points (devices, stations, phone-sets, soft-clients)
o Trunking (private or public)
o Interactive media-based applications
Currently considered as the protocol 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 follows the industry standards process (IETF) and will
progressively enlarge the level of services for each application as the market advances.
10.1 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 analog phones. Until
the recent emergence of IP convergence in 2000-1, analog 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 analog 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.
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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 depends on a
number of market pre-requisites that must be addressed before SIP becomes a true and viable
solution:
The devices must deploy correctly minimum criteria within the SIP protocol: RFC3261,
RFC3262, RFC3264, RFC3265, RFC3515.
The devices need to support a minimum of IP infra services and compatibility (including
codecs, POE, DHCP, QoS tickets)
The communication system has to control the SIP endpoints across network topologies
(including IP Domains, Call Admission Control, redundancy, survivability)
The level of telephony services provided to SIP phones is dependent on the capabilities of the
set (for example, conferencing resources in set). Standard SIP Telephony features are still
limited today. Careful assessment of end-users requirements must therefore be done before
installing the SIP sets.
Manageability/serviceability tools must ensure smooth/economic deployment (including
domain names, proxy addresses, VLAN assignments)
Finally and most importantly, the economies of scale are triggered that allows true low-cost
prices to be achieved: ISPs/ASPs (operators/carriers) must finally launch mass-market SIP
10.2 SIP basics

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 like that of HyperText Transport Protocol (HTTP) messages. Elements of the
SIP world are identified by SIP Uniform Resource Locators (URLs) like 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, acknowledgment 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.
10.3 SIP and OpenTouch

In OpenTouch SIP can be used for more of the basic communication applications. These include:
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Multimedia Routing application controls:
o Core Switching and handling
o Building rules for a user to route their calls to the endpoints or to another user
o User interaction for setting routing rules
User Session Manager to manage:
o Multiple device ownership
o Session handover between the devices
SIP Basic communication applications for basic and standard SIP endpoints. These include:
o Make, take, clear, reject and deflect calls
o CLIP/CLIR
o Multi-endpoints forking for user centricity, Multi-lines
o Hold/retrieve communication and enquiry communication
o Attendant and unattended transfer
o Immediate/no response/ on busy/ on busy or no response forwarding managed by TUI
o Call-backs for Busy or no answer calls
o Lock and unlock for each endpoint
o Message Waiting Indicator (MWI)
o Get and deposit message in Voice Mail from internal or external calls
o Peer to Peer Video between endpoints that support audio/video connections
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11. Session Initiation Protocol (SIP) networking
11.1 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.
11.2 Alcatel-Lucent and SIP
11.2.1 SIP and Alcatel-Lucent OmniPCX Enterprise Communication Server

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 (including IP Touch, H.323 trunks)
This implementation prepares the communication application architecture, to provide SIP enabled
applications to various sets and devices (whether SIP or not). This concerns notably:
Alcatel-Lucent OmniTouch 8400 Instant Communications Suite
GVP IP (Genesys Voice Portal)
Microsoft Exchange 2007
Selected Alcatel-Lucent Application Partner Program (AAPP) products such as Thomson ST
2022 & 2030, Ascom i75 and Teledex
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 (for example, 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 reregistering of SIP end points after a failure).
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11.3 Supported SIP standards
The Alcatel-Lucent OmniPCX Enterprise Communication Server is based on the latest standards. In
relation to SIP, the following RFCs are supported (they can be found on the IETF site at
http://www.ietf.org):
1889: RTP: A Transport Protocol for Real-Time Applications
2327: SDP: Session Description Protocol
2782: A DNS RR for specifying the location of services (DNS SRV)
2822: Internet Message Format
2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
2976: The SIP INFO method
3261: SIP: Session Initiation Protocol
3262: Reliability of Provisional Responses in SIP (PRACK)
3263: SIP: Locating SIP Servers
3264: An Offer / Answer model with SDP
3265: SIP-Specific Event Notification
3311: The Session Initiation Protocol (SIP) UPDATE Method
3323: Privacy Mechanism for the Session Initiation Protocol (SIP)
3324: short-term requirements for network asserted identity
3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within
Trusted Networks
3326: The Reason Header Field for the Session Initiation Protocol (SIP)
3458: Message Context for internet Mail
3515: The Session Initiation Protocol (SIP) refer method
3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap
Signaling to the Session Initiation Protocol (SIP)
3725: Third Party Call Control (3pcc): flow I
3842: A Message Summary and Message Waiting Indication Event Package for the Session
Initiation Protocol (SIP)
3891/3892: The Session Initiation Protocol (SIP) 'Replaces' Header/ Referred-By Mechanism
4028: Session Timers in the Session Initiation Protocol
4244: Extension to the Session Initiation Protocol (SIP) for Request History Information
4497: Inter-working between SIP and QSIG
4733: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
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11.4 Integration of SIP end-points in the Alcatel-Lucent OmniPCX
Enterprise Communication server network
11.4.1 Integration of SIP end-points in the PCX network

By implementing a complete SIP gateway/server solution, the Alcatel-Lucent OmniPCX Enterprise
Communication Server enables SIP end-points to be integrated to the PCX environment.
SIP end-points (that is, devices), connected to the Alcatel-Lucent OmniPCX Enterprise Communication
Server, operate as integrated SIP sets and can:
Fully communicate with devices controlled by the Alcatel-Lucent OmniPCX Enterprise
Communication Server (for example, TDM and IP sets)
Benefit from a large range of business telephony features available on the Alcatel-Lucent
OmniPCX Enterprise Communication Server (using the appropriate prefix/suffix on the SIP
set)
Be monitored by CSTA services and provide call events, call control processes and user
telephony presence (for example, make call, release call, busy set, free set)
11.4.1.1 Compatible SIP end-points

The following products can operate as integrated SIP end-points with an OmniPCX Enterprise:
Alcatel-Lucent IP Touch 4008/4018 phone Extended Edition sets operating in SIP mode and
communicating via the SIP interface of the OmniPCX Enterprise.
Partner SIP entry-level sets.
After careful examination and evaluation of potential suppliers, Alcatel-Lucent has chosen
Thomson and Ascom as strategic partners in the field of professional SIP sets, whose product
line is currently residential oriented, but evolves and matures in time to become a reliable and
deployable enterprise IP device. The following sets were approved:
o Thomson ST 2022/2030 sets
o Teledex SIP sets (dedicated to the hospitality market segment)
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 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
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Etc.
The following product guideline indicates the advantages and visibility of Alcatel-Lucents support of
SIP proposed by Thomson, an alternative SIP solution proposed by Alcatel-Lucent, and the end-to-
end (proprietary) technology.
This product guideline is designed to allow Alcatel-Lucents customers to weigh the consequences and
potential strategic impact on their core-enterprise communication strategy.
Entry level IP Alcatel-Lucent IP Alcatel-Lucent IP
sets Thomson ST 2030 Touch 4008/4018 EE Touch 4008/4018 EE
(SIP) (SIP mode) (Alcatel-Lucent
proprietary mode)
Basic features
Mono-
Multiline (5) Multiline (2) Multiline (6)
line/multiline
Hands-
LS Full hands free Full hands free
free/loudspeaker
CLIP name and

Yes Yes Yes
number
Hold & broker call Yes Yes Yes
Transfer Yes Yes Yes
Call waiting No Yes Yes
Call forwarding Yes Yes Yes
3-party conference Yes Yes (local) Yes
Voice-mail services
Yes + record, menu, VG,
(access, MWI, Yes Yes

forward)
Advanced features See below See below All
Infra support
PoE: 802.3af +
Yes Class 0 Yes Class 2 Yes Class 2
option
DHCP/TFTP Yes Yes Yes
QoS
Yes Yes Yes
802.1p/DiffServ
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proprietary mode)
Automatic Voice Yes (Thomson configuration Yes (802.1x, LLDP-MED, Yes (802.1x, LLDP-MED,
VLAN Assignment file) AVA) AVA)
Codec selection No Yes Yes
Codec: G711,
Yes Yes Yes
G723.1, G729A
Wideband ready No No No
Authentication
HTTP digest auth. HTTP digest auth. IPSec
(application level)
802.1X No Yes Yes
Communication
No No Yes
encryption
Com Servers
redundancy (level Yes Yes Yes
2)
Com Server spatial

redundancy (Level Yes (with DNS) Yes Yes
3)
Call Admission
Yes Yes Yes
Control (CAC)
Media Gateway
No No Yes
signaling backup
Passive Com
Yes Yes Yes
Server (PCS)
System integration
Barring, CDR, ARS Yes Yes
with OmniPCX
Management
integration with Partial Yes Yes
OmniPCX
DTMF: RFC 2833, Yes Yes Yes
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proprietary mode)
G711 in band
11.4.2 SIP end-point configuration and registration

Unlike sets that are stimuli terminals completely controlled by the system, SIP sets are intelligent
end-points that require item (local) management. Generally, the SIP sets have integrated Web server
management tools. These tools are required to manage SIP sets on an individual basis.
To operate as integrated SIP sets and benefit from advanced telephony features, so far reserved for
standard internal Alcatel-Lucent devices, SIP devices must follow a process including:
The SIP device declaration in the Alcatel-Lucent OmniPCX Enterprise Communication Server
database. This operation automatically triggers the SIP device registration in the SIP
dictionary (within the Alcatel-Lucent OmniPCX Enterprise Communication Server).
The SIP device commissioning.
o For partner SIP entry-level sets (Thomson ST 2022/2030), the SIP parameters
configuration can be fully performed in the set itself or downloaded from the SIP
Manager application embedded in the OmniVista 8770
o For Alcatel-Lucent IP Touch 4008/4018 phone Extended Edition sets, the SIP
parameters configuration includes:
SIP mode activation, either from the set itself or a configuration file
downloaded from a server
SIP configuration, using configuration files downloaded from a server
These configuration files are built with the SIP Manager application embedded in
the OmniVista 8770.
The SIP device registration (after device reboot) in the registrar server of the Alcatel-Lucent
OmniPCX Enterprise Communication Server. By registering, the PCX database links the IP
address of a SIP device to a directory number, and provides:
o 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 PCX
o A user name and first name in the PCX phone book
o A Class of Service (COS) for barring/restriction and accounting
o A voice mailbox including waiting messages notification
Example:
A SIP user called Bob has the following SIP address:
5000@sip.company.com
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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. This number must 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 bob@sip.company.com. This identity can also be used for calling the device/user.
First, Bob is registered on his/her SIP set (by the administrator or by user-programming according to
the set capabilities). The address is sip:5000@sip.company.com.
The SIP set 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.
11.4.3 SIP end-point deployment on a large scale

The SIP Manager application, embedded in the OmniVista 8770, provides a graphical user interface
enabling SIP devices' commissioning and SIP devices' daily maintenance. This application allows to
deploy quickly a large quantity of SIP devices.
The SIP Manager application operates in
association with a dedicated server
named Device Management (or DM).
The SIP Manager application enables,
notably:
To configure SIP users and SIP
devices
To assign one or several SIP
devices to a SIP user
To make configuration data
accessible to SIP devices for their
commissioning. Configuration
data are registered on the server
and downloaded by SIP devices
via TFTP, HTTP, or HTTPS.
The SIP Manager is open to deploy
partner SIP entry-level sets (Thomson ST
2022/2030)
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11.4.4 Telephony services available for SIP end-points
The following telephony services are provided when SIP sets communicate with the Alcatel-Lucent
OmniPCX Enterprise Communication Server.
Important:
These are interconnected services supported by the Alcatel-Lucent OmniPCX Enterprise
Communication Server gateway, and not services provided by the call handling feature for SIP sets.
Therefore, these services can only be provided if the involved SIP set is able to support them.
Presentation of the service on the set is also independent of the Alcatel-Lucent OmniPCX Enterprise
Communication Server (display, etc.).
For example, certain SIP sets use their own resources to set up a conference (for example, the
OmniPCX conference bridge is not used when the user of the SIP set initiates a conference when the
conference bridge is provided by the SIP set).
11.4.4.1 Telephony services access

Accessing the Alcatel-Lucent OmniPCX Enterprise Communication Server telephony services is
provided by dialing the corresponding prefixes on the SIP set.
For Alcatel-Lucent IP Touch 4008/4018

phone Extended Edition sets registered
as SIP sets on the Alcatel-Lucent
OmniPCX Enterprise Communication
Server, their programmable keys are
automatically associated to a specific
telephony service.
A set of paper label is proposed for a
quick and easy identification of
associated keys.
11.4.4.2 Telephony services display

The SIP devices can display a high level of information relating to telephony services. When a
telephony service is activated on the SIP set, the Com Server returns a SIP message to the SIP
device providing its status for this telephony service. This level of services is supported by the partner
SIP entry-level sets (Thomson ST 2022/2030).
Example: If an appointment is programmed, the SIP set can display the following message:
Appointment at 12:00
Information displayed in the SIP message applies to the following telephony services:
Forward activation
Do not disturb activation
Remote extension deactivation (OmniPCX Integrated Cellular Client service)
Hunting group belonging
Appointment or wake-up programmed
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Lock activation
11.4.4.3 Basic telephony services

Automatic answer
Basic incoming and outgoing calls
Broker call
Callback request (using call log)
Call diversion on ringing
Camp-on
Consultation call
DTMF transparency
Registration with or without authentication
Speed dialing
Transfer (attended and unattended)
11.4.4.4 Advanced telephony services

Appointment reminder and wake-up call
Barge-in
Call announce
Call by name (depending on the SIP set capabilities)
Call forwarding (unconditional, on busy, on no answer)
Call hold
Call park
Do Not Disturb (DND)
Multiline
Night forwarding
11.4.4.5 Team work

Conferences (three-party, casual, mastered and meet-me conferences)
Pick-up or Hunting group
Supervision
11.4.4.6 Complementary services

Accounting
CLIP/CLIR (secret identity)
InfoCenter facilities
Hospitality
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Multi-tenancy
Twin sets
Voice mail access and notification
11.4.5 Basic SIP call via the PCX SIP proxy
11.4.5.1 SIP to SIP call

If the above example is used:
If another integrated SIP set dials the DN: 5000, the call is routed by the set to its "outbound" proxy,
which is the Alcatel-Lucent OmniPCX Enterprise Communication Server SIP proxy.
The SIP proxy locates the called SIP set and forwards the message accordingly.
11.4.5.2 Calls from other Alcatel-Lucent OmniPCX Enterprise Communication Server

devices to SIP end-points
Users of IP Touch sets 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 an integrated SIP set with a directory number 5000, an Alcatel-Lucent OmniPCX Enterprise
Communication Server user dials 5000. The SIP gateway then sends sip:5000@sip.company.com to
the SIP proxy.
IP Touch sets that have an alphanumeric keyboard/keypad can also use the call by name service
(phone book name) to call integrated SIP sets, since names of integrated SIP sets can be in the
OmniPCX phone book.
11.4.5.3 Calls from SIP end-points to other devices on Alcatel-Lucent OmniPCX

Enterprise Communication Server
Integrated SIP sets can call other devices by dialing sip: <directory number>@ <omnipcx-address>,
where the DN can be a set number or a public number. Some SIP sets provide speed dialing so those
users only dial the directory numbers.
Names of integrated SIP sets can be recorded in the OmniPCX phone book. These names can be
exported to the Alcatel-Lucent OmniVista 8770 LDAP directory. Therefore, SIP terminals with an 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 OmniVista 8770 LDAP directory.
11.4.6 Media considerations for SIP within the PCX
11.4.6.1 Type of media

Three situations can be identified:
1 Communications between integrated SIP sets controlled by the same proxy
Any type of media can be used according to the SIP device type. For example: softphones
equipped with the right peripheral equipment (including camera and handsets) allows users to
establish video conferencing.
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2 Communications between integrated SIP sets to/from other Alcatel-Lucent OmniPCX Enterprise
Communication Server devices
Voice media is supported including G711 A/, G723.1 or G729A.
This is also true for communications between integrated SIP sets and other SIP clients.
3 Communications to fax machines. Fax T38 Annex D is supported over IP.
11.4.6.2 RTP flow

Media over IP is transported in Real-time 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 integrated SIP set and non-SIP (Alcatel-Lucent User) set
Scenario 2: Communication between integrated SIP set and unknown SIP set
Scenario 3: Communication between non-SIP (Alcatel-Lucent User) set and unknown SIP set
Definition: Unknown SIP set is not integrated in Alcatel-Lucent OmniPCX Enterprise Communication
Server.
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11.4.7 Calls made to TDM trunks
An integrated SIP set 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 integrated SIP sets that are directed towards
TDM trunks.
Example: User Bob (directory number 5000) dials a PSTN number. User Bob's integrated SIP set 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.
11.4.8 Voice mail account

SIP integrated users can have an account on voice messaging systems supported on the Alcatel-
Lucent OmniPCX Enterprise Communication Server:
Alcatel-Lucent OmniTouch 8440 Messaging Software
Alcatel-Lucent 4635 Voice Messaging System
Alcatel-Lucent 4645 Voice Messaging System
Alcatel-Lucent OmniTouch 8400 Instant Communications Suite (My Messaging)
Integrated SIP sets can be forwarded to the OmniPCX voice messaging system. They receive a
message waiting indication. Navigation in a voice mail account from an Integrated SIP set is
performed via touch tone signals.
For Alcatel-Lucent IP Touch 4008/4018 phone Extended Edition sets operating in SIP
mode, voice mail access is carried out by the Message key. The Message LED flashes
when a new message is received
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11.4.9 Authentication verification
The SIP proxy can identify the integrated SIP set based on information received through the
exchanged messages (invite).
Inside a corporate network, SIP authentication (verification) is not required as both the 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 set 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 set and the proxy is authenticated:
With HTTP digest authentication, user login and password are encrypted (MD5 process). This is
based on shared keys authentication.
11.4.10 Keep-alive dialog

A keep-alive dialog can be established between the Alcatel-Lucent OmniPCX Enterprise
Communication Server and SIP sets. The keep-alive dialog, initiated by the SIP set, allows the
Alcatel-Lucent OmniPCX Enterprise Communication Server to check whether the SIP set is in service.
Periodically, the SIP set sends an OPTION message to the Com Server to indicate it is operational.
The keep-alive dialog requires that SIP sets can send OPTION messages to the Com Server.
The keep-alive dialog is useful:
To optimize overflow mechanisms
To maintain session in infrastructure (router, SBC, FW) NAT tables
To optimize GSM-WIFI switching in the MCE dual-mode solution (Alcatel-Lucent OmniTouch
8622 My Cellular Extension)
11.4.11 What SIP services are offered by an integrated SIP set?

From a technical point of view
1 Native SIP sets or a SIP application on a PC (e.g. My IC SIP softphone)
2 Small FXS gateways, SIP router OmniAccess, SIP router Teldat (Spanish market), SIP AudioCodes
MP-1xx series connecting analog devices, with one directory number per analog port.
For example, the same OmniPCX SIP Proxy sees a gateway with two analog accesses as two different
SIP devices (with two registrars). These gateways must be able to register.
The Alcatel-Lucent Application Partner Program has tested SIP AudioCodes MP-1xx, SIP router
OmniAccess and SIP router Teldat successfully. Therefore, connectivity of a SIP analog FXS gateway
can be supported on the Alcatel-Lucent OmniPCX Enterprise Communication Server.
Other gateways (T2/T1 to SIP gateways), which are seen as hunting groups, are not able to benefit
from the integrated SIP set services (see SIP trunking).
11.5 SIP trunking

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.
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Several SIP trunks groups can be created:
One main trunk group is created for the external SIP extensions and Alcatel-Lucent
OmniTouch 8400 Instant Communications Suite
Separate trunk groups are created for external gateways
SIP trunk capacity is an example of the evolution speed in SIP technology. Current capacities can
fulfill requirements for large-size installations, including installations with numerous remote sites.
If an installation including several remote sites, it is possible to configure a local SIP connection to
the PSTN for each remote site. This, for example, enables accounting tickets to include the SIP trunk
group number and to adapt call costs to each site.
The configuration of mini-SIP trunk groups instead of standard SIP trunk groups enables to limit the
number of trunks per trunk group, so that the total number of trunks remains lower than the
maximum authorized, even in case of many trunk groups.
11.5.1 Public SIP trunking

The communication server performs the SIP trunk group signaling.
When the Alcatel-Lucent OmniPCX Enterprise Communication Server 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.
TLS/SRTP protocols can deployed to protect SIP communication with external/ public SIP
Gateways. TLS protects SIP signaling and the SRTP protocol (two keys: one key is used to
encrypt sent voice flow and an other key is used to encrypt received flows) protects voice
flows. Thus, End-to-end encryption, between carrier and all OmniPCX Enterprise equipments
(IP Phone, Media Gateway, Com Server, ) can be proposed.
Note: end-to-end ABC subnetwork encryption is not yet possible.
The SBC is the access point to the carrier infrastructure
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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.
Call Admission control (CAC) is configured in the Alcatel-Lucent OmniPCX Enterprise Communication
Server 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 sets
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.
11.5.2 Domain name resolution

In order to route outgoing requests towards an external gateway, the domain name of the external
gateway or of the outbound proxy must be resolved. This is done with DNS.
The Alcatel-Lucent OmniPCX Enterprise Communication Server supports two types of name
resolution: DNS A and DNS SRV:
DNS A resolution enables the resolution of a name into one single IP address
DNS SRV resolution enables the resolution of a domain name into one or several domain
names, which are in turn resolved into IP addresses using DNS A requests. Each record in a
DNS SRV answer has a priority order, indicating the order in which requests must be sent. A
request must first be sent to the IP address corresponding to the highest priority level. If no
answer is obtained from this address, the request must then be sent to the IP address
corresponding to the record with second highest level, and so forth,
The DNS SRV implementation on the PCX complies with RFC 2782.
The type of resolution (DNS A or DNS SRV) to be used is selected for each external SIP gateway in its
configuration parameters. Two DNS server addresses may be configured for each SIP gateway.
Note: NAPTR is not implemented on the PCX. The protocol used (TCP or UDP) for SIP messages is
the one selected in the gateway configuration parameters.
11.5.3 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 and direct T38 for Fax communications.
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11.5.4 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 involves the SIP network to
perform the service. If yes, means that the service can also be activated on carrier side.
SIP-related Features User SIP public SIP Comments

side trunk side private
(NGN trunk
involved) side
Basic call Outgoing with Yes Yes Yes Block dialing for OG
number and name
Canonical form for public numbering
display
<sip:+33390677517@my_domain>
RFC 3261
Basic Call Incoming with Yes Yes Yes Canonical form for public numbering
DDI with number and
<sip:+33390677517@my_domain>
name display
Range of DDI numbers/users must be
provisioned at the NGN side
RFC 3261
Calling line/name Yes Yes Yes Provided there is compliance with RFC
identification 3323, 3324 & 3325 is ensured and the
presentation (CLIP/CNIP) Alcatel-Lucent OmniPCX Enterprise
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(NGN trunk
involved) side
Communication Server is considered as a
trusted element
Calling line identification Yes Yes Yes Provided there is compliance with RFC
restriction (CLIR) 3323, 3324 & 3325 is ensured and the
Alcatel-Lucent OmniPCX Enterprise
Communication Server is considered as a
trusted element
Connected line/name Yes Yes Yes

identification
presentation (COLP)
Connected line/name Yes Yes Yes

identification restriction
(COLR)
Call forwarding Yes No Yes User side: Performed by joining both half
unconditional (CFU) and calls within Alcatel-Lucent OmniPCX
on busy/No answer Enterprise Communication Server. RTP
(CFB/CFNR) 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 invite, call
Broker/Conference hold, and conference (Conference in the
Alcatel-Lucent OmniPCX Enterprise
Communication Server, not using SIP
conference packages)
Attended Transfer/Early Yes No Yes User side: Performed by joining both half
Attended transfer calls within Alcatel-Lucent OmniPCX
Enterprise Communication Server. RTP
optimized.
Private trunk: Provided there is compliance
with RFC 3515 (SIP Refer method)
DTMF overdialing Yes Yes Yes Provided there is compliance with RFC
2833
Fax support Yes Yes Yes T38 ITU-T annex D with G711 fallback
G711 transparent
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(NGN trunk
involved) side
Same integration with Yes No Features like DDI, differentiated ringing,

user and system call parking, overflow to attendant, charging,
handling features than overflow to ISDN
for ISDN trunks
SIP related Features SIP public SIP private Comments

trunk side trunk side
Authentication for Outgoing Yes Yes Provided there is compliance with RFC
calls/Incoming calls 2617 & RFC 1321
Dynamic registration of Yes Range of DDI numbers/users must be

OmniPCX SIP gateway to provisioned statically at the NGN side
Carrier SIP proxy (with or
without authentication)
Alternate SIP proxy through Yes Alcatel-Lucent OmniPCX Enterprise

ARS Communication Server side
SIP keep alives to SIP proxy Yes
Message Waiting Indication No Yes Provided there is compliance with RFC

3842
Several SIP trunks Yes Yes Alcatel-Lucent OmniPCX Enterprise

Communication Server side
Call Admission Control SIP Yes Yes IP domain CAC takes into account
SIP sets inside IP domains
Multi-codec support Yes Re-negotiation of codecs in case of

communication evolutions to select
the best quality codec and avoid
audio transcoding
Tel URI Yes RFC 3966 To support tel URI for

incoming calls in addition to sip URI:
IMS business trunking conformance
Registration on Outbound Yes Allows OXE to register to a second

proxy discovery carrier proxy as soon as the first one
is no more active : better availability
for public SIP trunk without losing
calls
P-early media Yes RFC 5009 - P-early media is for
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SIP related Features SIP public SIP private Comments
trunk side trunk side
policing early media (Ring Back
Tone). Avoid user to stay without
audio in ringing phase in some cases.
Authentication nonce caching Yes SIP authentication procedure

optimization to avoid challenging
systematically all SIP messages
SIP Trunking and G722 Yes G722 Wide Band Audio on SIP trunk .
compatibility Available for G722 capable sets not
IPMG / OXE MS
Law A-mu transcoding with Yes

SIP Trunking
Avoid deploying additional CS node to
offer a A-Mu law conversion. Both
codecs can be sent by the same
carrier to the same OXE
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.
11.6 Unknown SIP sets

Unknown SIP devices (that are not integrated SIP sets) 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).
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.
11.7 SIP communication operations at Communication server

changeover
SIP communications are maintained when a Communication Server changeover occurs. Depending on
the call origin and its state:
Initiating calls (that is, SIP calls being set up) are interrupted, but incoming calls from an
external trunking gateway (SIP trunking provided by the carrier) are routed to the relevant
entity
Ongoing calls (that is, established SIP calls) are maintained with a few exceptions, as for
example, calls established to an internal voice mail, which are released
When the state of a SIP communication changes, the main Communication Server systematically
updates the standby Communication Server with the relevant information.
Generally speaking, the SIP communications processing at Communication Server switchover is
similar to that of typical telephone communications. The standard restrictions also apply to SIP
communications.
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12. H.323 architecture
H.323 is used for:
Peer-to-peer networking between several Alcatel-Lucent OmniPCX Enterprise Communication
Server platforms (see section 3.1 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)
12.1 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
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:
o Direct and routed mode
o Connection to an external gatekeeper or to the integrated gatekeeper
H.323 annex D (for T38 fax)
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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.
12.2 Direct RTP and transit RTP 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 OmniTouch 8400 ICS or a SIP phone
The following diagram demonstrates flows with direct RTP.
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
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o INTIPA supports 60 H.323 signaling sessions
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 1: RTP direct in H.323 is supported between Alcatel-Lucent OmniPCX Enterprise Communication
Server systems interconnected with ABC and with other H.323 gateways/terminals.
Note 2: 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.
12.3 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
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)
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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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Who Should Use this Document?

1. Architectural reliability ............................................................................................................ 6

3. Additional IP services ............................................................................................................. 18

9. IP Telephony domains and related features ........................................................................... 46

1.1 Communication server redundancy (duplication)

1.1.1 IP rack server (Common hardware) and Appliance server

1.1.2 Com server changeover

1.1.3 Database consistency

1.2 Communication server survivability for OpenTouch users

1.3 Secure call processing with the passive communication server

Breakdown of IP network links:

1.5 Backup signaling for IP media gateways

1.6 IP Touch sets survivability

1.7 Business continuity of IP Touch in case of IP failure

1.8 Business continuity for out of service Digital/Analog devices

1.9 Fallback when SIP Voice Messaging is not reachable

1.10 Dual LAN attachment

1.11 Uninterruptible power supply

2.1 System time clock

2.2 NTP service on OmniPCX Enterprise components

2.2.1 NTP services

2.2.2 Synchronization method

2.2.3 NTP and current Alcatel-Lucent OmniPCX Enterprise Communication Server

3.1 Fax machines

3.2 Modem and data transparency

3.2.1 Modem transparency

3.2.2 Data transparency

3.4 Trivial File Transport Protocol (TFTP)

3.5 Dynamic Host Control Protocol (DHCP) server

3.6.1 Different VLAN assignment methods

3.6.3 Automatic VLAN Assignment (AVA) through DHCP

3.7 VoIP supervision

Incoming DID calls

4.1 Class of Service (CoS)

4.2 Routing table

4.2.2 Routing in the network

4.2.3 Night service

4.3 Call restriction for alarms and emergencies

4.4 Priority calls Multi-Level Precedence and Pre-emption (MLPP)

1 routing towards an assistant

2 routing towards a job profile

Limits and restrictions

6.1 Configuring the application

6.2 User counters view

6.3 Real time records printing

6.3.2 V24 support

Justified (charged) user directory number X X

Justified (charged) user name X

Project number (business account code) or PIN (1)

Call end date X

Justified (charged) cost center X X

External carrier used for the call X

Trunk group number X

Trunk node number X

Justified (charged) user node number X

Call type (voice, data) X X

Directory number of the set that transferred the X X

Number sent/received (2)

Call end time X X

Number of charge units or call cost X X

Project (business account) number (0 if not X X

Call type (personal/professional) X X