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Standard Offer
Chapter 03 System Services
April 2014 Offer - Ed.02
Ref.: ENT_MLE_015989
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Table of contents
1.
2.
1.3
1.4
Secure call processing for Conversation users with the passive communication server ................ 10
1.5
1.6
1.7
1.8
1.9
1.10
1.11
2.2
3.2
3.2.1
3.2.2
3.3
Framing ..................................................................................................................................... 18
3.4
3.5
3.6
3.6.1
3.6.2
3.6.3
3.7
4.
1.2
2.2.1
2.2.2
2.2.3
3.
4.2
4.2.1
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4.2.2
4.2.3
4.3
4.4
5.
6.
6.2
6.3
6.4
6.5
6.5.1
6.5.2
Ethernet ............................................................................................................................................................ 32
V24 support ...................................................................................................................................................... 32
6.6
6.7
7.
8.
8.2
8.2.1
8.2.2
8.2.3
8.2.4
8.2.5
8.3
8.4
8.4.1
8.4.2
9.
9.2
9.3
9.4
Multi-country ............................................................................................................................ 45
9.5
9.6
Restriction ................................................................................................................................. 46
10.
10.1
10.2
SIP basics................................................................................................................................... 48
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10.3
11.
11.1
11.2
11.2.1
11.3
11.4 Integration of SIP end-points in the Alcatel-Lucent OmniPCX Enterprise Communication server
network ............................................................................................................................................... 51
11.4.1
11.4.2
11.4.3
11.4.4
11.4.5
11.4.6
11.4.7
11.4.8
11.4.9
11.4.10
11.4.11
11.5
11.5.1
11.5.2
11.5.3
11.5.4
11.6
11.7
11.8
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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 available options and features.
Note: To avoid possible communication server performance distortion, the CPU hardware must be
identical for both the main and standby communication servers.
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.
All the following applications are able to address the dual IP addresses:
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Configuration information
CCD data
If the standby Communication Server becomes operational before the expiration of the
storage duration, the main Communication Server sends MAO commands to the standby
Communication Server which automatically updates its database. As a result, the two
databases become consistent.
If the standby Communication Server is still unreachable after the expiration of the
storage duration, the main Communication Server stops storing MAO commands and
deletes them. When the standby Communication Server becomes reachable, the two
databases must be consistent via a database cloning operation (or master copy
operation).
Automatic database synchronization is available for Communication Servers located on CS-2 boards
(Common Hardware), Appliance Servers and Blade Centers. This operating mode is identical in a
double main configuration.
Information handled by MAO commands are:
Configuration of set parameters (secret code, language, user name, keys, etc.). This
applies to digital and cordless sets
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Note: Data relating to CCA, OmniVista 8770, Alcatel-Lucent OmniTouch Unified Communications, and
OmniTouch Fax Server are not duplicated.
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access the PCS). The Domain Name (DN) resolution is performed with DNS A (DNS SRV is not
supported).
SIP registrations on the main Communication Server are not duplicated on the PCS. When the PCS
becomes active (Communication 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 IP Phones and/or Media Gateways within a region are defined for the PCS
All standalone features defined inside active call processing are maintained by the PCS
including OmniTouch Contact Centers.
IP phones and media gateway reboot and are under the OmniPCX Enterprise control
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 Conversation 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 Conversation
users.
In an OpenTouch configuration, the PCS provides the continuity of telephony services and the
Contact Center in case of:
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If the OpenTouch server crashes (configuration without 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).
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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.
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 Communication 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.
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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.
Load balancing
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Note: Only the UPS recommended by AlcatelLucent for the OmniPCX Enterprise are
supported.
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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.
Inasmuch 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).
Display of the NTP server chain, in order to know the primary time source
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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.
System commands
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 ITUT.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:
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).
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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
Framing (ms)
Bandwidth at IP level
(KBPS)
G711
64
10
96
20
80
30
74.6
10
40
20
24
30
18.6
40
16
30
17
G729A
G723
6.3
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An existing PC server
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.
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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.
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If different values are received or configured on the set, the following priority rules apply.
If the set initializes in dynamic mode, the following priority rules 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 can 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 issues 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.
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 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).
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3.6.3.1 Prerequisites
AVA compatibility
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 most common 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.
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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
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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.
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Attendant console
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Voice call priority
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:
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Visit Card
Personal
advisor
busy*
Direct call
Customer
Customer calls his personal
advisor which responds if available.
If he is unavailable, then route to
any one in the branch office according
to business rules & skills
1
Available
specialists
Route to
Contact
center
Attendant
Voicemail
Key features
Business Contact Unified engine with built-in predefined strategies:
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Benefits
Ensures a unified welcome across a distributed business organization
Is easy to manage and use
Fits perfectly in your existing (Alcatel-Lucent) IP architectures
Improves customer satisfaction by drastically reducing lost calls
Helps to measure efficiently the performance of advisors
Prerequisites
Hardware (not included)
Quad core processor 2,0 GHz, 4 GB of RAM, 80 GB of hard disk space
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 TSAPI server
Limits and restrictions
No CTI monitoring of the Business Contact Standard Edition users: no screen pop-up or CTI
integration is 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.
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6. Integrated accounting
6.1 Call accounting (metering) options
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 the
telecommunication costs as well as billing for the different cost centers or users, for 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
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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
6.5.1 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 the OmniPCX Enterprise Communication Server protocol. The transmitted
records contain all available information.
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Extended format
Reduced format
(1)
Trunk number
Number sent/received
Call duration
(2)
(3)
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X
Alcatel-Lucent Enterprise Communication Solutions
April 2014 Offer
Notes:
(1) (2) Masking
The 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) The internal services that may be used include VPN, ISVPN, ARS, speed dialing, remote charge,
text messages, etc.
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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
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
01/01/2014-02/01/2014
02/01/2014-03/01/2015
01/01/2014-01/10/2014
01/10/2014-01/19/2014
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
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Key features
Modular offer
RECORD: rich recording facilities of inbound and outbound calls in multiple audio
formats (MP3, GSM6.10, WAV)
Call recording in SIP (declared as SIP SEPLOS), VoIP, analog, digital and mixed
environments
Support of virtualization
Records 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 resolution of customer issues
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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.
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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 using the scripts of
the standard recordings and/or providing any additional relevant information. 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 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)
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
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For more information on specific greeting use, see the automated attendant section in the document
on attendants.
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.
Dynamic: modifiable, recorded by users and/or managers via a phone set or the AlcatelLucent Audio Station (AAS) application.
External: music-on-hold (in most cases) played by a device external to the OmniPCX.
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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
3
Static chained voice guides: composed of several non-variable messages played in a preconfigured order (sequence).
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.
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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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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.
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(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:
Tones setup. The tone generation definition of the corresponding country is sent by the
Communication Server to the Media Gateway at Media Gateway initialization. For IP Phones,
the Communication Server sends the tone generation definition of the corresponding country
to the set.
When a tone is sent to a device, the Communication 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 complete number to use to
return the call, 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 as well as 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
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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
9.6 Restriction
It is not possible to mix trunk groups using two different laws (A or ) on a same node. Therefore it
is not possible to build a node containing sites in countries using different laws.
Multi-time zone and multi-country features are not supported with the OpenTouch Business Edition.
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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
opting for SIP sets.
Manageability/serviceability tools must ensure smooth/economic deployment (including
domain names, proxy addresses, VLAN assignments)
Finally and most importantly, economies of scale are triggered and allow true low-cost prices
to be achieved: ISPs/ASPs (operators/carriers) must finally launch mass-market SIP
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SIP Basic
o
o
o
o
o
o
o
o
o
o
o
communication applications for basic and standard SIP endpoints. These include:
Make, take, clear, reject and deflect calls
CLIP/CLIR
Multi-endpoints forking for user centricity, Multi-lines
Hold/retrieve communication and consultation calls
Attendant and unattended (blind) transfer
Immediate/no response/ on busy/ on busy or no response call forwarding
managed by the Telephone User Interface (TUI)
Callbacks for calls on busy telephones or unanswered calls
Lock and unlock of each endpoint
Message Waiting Indicator (MWI)
Get and deposit message in voice mail from internal or external calls
Peer to peer video calls between endpoints that support audio/video connections
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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:
The 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 (for example, to find the IP address) in order to locate 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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11.4.1.1
Compatible SIP end-points
The following products can operate as integrated SIP end-points with an OmniPCX Enterprise:
Alcatel-Lucent IP Touch 8002/8012 sets (as well as 4008/4018 sets) operating in SIP mode
and communicating via the SIP interface of the OmniPCX Enterprise.
Alcatel-Lucent IP Touch 8082 sets operating in SIP mode for hospitality deployment. This
device can be deployed in Suite room, offering both SIP telephony and Web Hotel Services.
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
Etc.
11.4.2
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 include 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
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Example:
A SIP user called Bob has the following SIP address:
5000@sip.company.com
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 AlcatelLucent 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 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.
And, due to previous configuration, Bob has directory number 5000 in the Communication Server.
11.4.3
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 a
fast deployment of a large quantity of SIP devices.
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11.4.4
OmniVista
8770 NMS
(SIP Manager)
8002/8012
SIP Phones
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 can 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 using its
own conference bridge).
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 8002/8012 phone
Extended Edition sets registered as SIP
sets on the Alcatel-Lucent OmniPCX
Enterprise Communication Server,
programmable keys are automatically
associated to specific telephony
services.
A set of paper labels is offered for
quick and easy identification of
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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 Communication 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:
Call forwarding activation
Do not disturb activation
Remote extension deactivation (OmniPCX Integrated Cellular Client service)
Hunting group
Appointment or wake-up programmed
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 call forwarding
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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
Multi-tenancy
Twin sets
Voice mail access and notification
11.4.5
11.4.5.1
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11.4.5.3
Calls from SIP end-points to other devices on the Alcatel-Lucent OmniPCX
Enterprise Communication Server
Integrated SIP sets can call other devices by dialing sip: <directory number>@ <omnipcx-address>,
where the directory number can be a set number or a public number. Some SIP sets provide speed
dialing features, and their users only need to dial 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
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 adequate peripheral equipment (including camera and handsets) can start and
participate in video conferencing.
2 Communications between integrated SIP sets to/from other Alcatel-Lucent OmniPCX Enterprise
Communication Server devices
Voice media is supported, including the protocols 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 T.38 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.
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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: An unknown SIP set is a SIP set that has not been integrated to the Alcatel-Lucent
OmniPCX Enterprise Communication Server.
11.4.7
An integrated SIP set has an Alcatel-Lucent OmniPCX Enterprise Communication Server user profile,
with a specific class of service containing the barring/restriction feature.
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.
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11.4.8
SIP integrated users can have an account on voice messaging systems supported by the AlcatelLucent OmniPCX Enterprise Communication Server:
Alcatel-Lucent 4645 Voice Messaging System
Alcatel-Lucent OpenTouch Message Center
Alcatel-Lucent OpenTouch Multimedia Services
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 4008/4018 and 8002/8012 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.
11.4.9
Authentication verification
The SIP proxy can identify the integrated SIP set from the information received through the
exchanged messages (invite).
Within a corporate network, SIP authentication (verification) is not required as both the set and the
proxy are managed and controlled by system administration, 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
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11.4.11
11.5.1
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TLS/SRTP protocols can be deployed to protect SIP communication with external/ public SIP
Gateways. TLS protects SIP signaling and the SRTP protocol (there are two keys: one key is
used to encrypt sent voice flow and another key is used to encrypt received flows) protects
voice flows. This means that end-to-end encryption, between carrier and all OmniPCX
Enterprise devices (IP Phone, Media Gateway, Communication Server, ) can be offered.
Note: end-to-end ABC subnetwork encryption is not yet possible.
The SBC is the access point to the carrier infrastructure
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 class of service, connection class of service and entity are taken into account.
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. The bandwidth must be calculated to avoid deterioration
in voice quality.
11.5.2
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,
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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: Network Authority PoinTeR (NAPTR) is not implemented on the PCX. The protocol used (TCP or
UDP) for SIP messages is the protocol selected in 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 T.38 for Fax communications.
11.5.4
The tables in this section indicate the telephony features supported by the Alcatel-Lucent OmniPCX
Enterprise Communication Server with their current limitations.
In the following tables:
User side indicates if the service is available to a user on the OmniPCX side when
communicating with/through a SIP trunk
SIP public trunk side indicates that a service activated on the OmniPCX involves the SIP
network to perform the service. If yes, then the service can also be activated on the carrier
side.
SIP-related Features
User
side
SIP public
trunk side
(NGN
involved)
SIP
private
trunk
side
Comments
Yes
Yes
Yes
Yes
Yes
Yes
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SIP-related Features
User
side
SIP public
trunk side
(NGN
involved)
SIP
private
trunk
side
name display
Comments
<sip:+33390677517@my_domain>
Range of DDI numbers/users must be
provisioned at the NGN side
RFC 3261
Calling line/name
Yes
identification
presentation (CLIP/CNIP)
Yes
Yes
Yes
Yes
Yes
Connected line/name
identification
presentation (COLP)
Yes
Yes
Yes
Connected line/name
identification restriction
(COLR)
Yes
Yes
Yes
Call forwarding
unconditional (CFU) and
on busy/No answer
(CFB/CFNR)
Yes
No
Yes
Yes
Yes
Yes
Consultation/
Broker/Conference
Yes
Yes
Yes
Attended Transfer/Early
Attended transfer
Yes
No
Yes
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SIP-related Features
User
side
SIP public
trunk side
(NGN
involved)
SIP
private
trunk
side
Comments
Yes
Yes
Yes
Fax support
Yes
Yes
Yes
Yes
No
SIP public
trunk side
SIP private
trunk side
Comments
Yes
Yes
Dynamic registration of
OmniPCX SIP gateway to
Carrier SIP proxy (with or
without authentication)
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Multi-codec support
Yes
Tel URI
Yes
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SIP public
trunk side
SIP private
trunk side
Comments
Registration on Outbound
proxy discovery
Yes
P-early media
Yes
Yes
Yes
Avoid deploying additional CS node to
offer an 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.
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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.
END OF DOCUMENT
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