An Alcatel service agreement brings your company the assurance of 7x24 no-excuses technical
support. You’ll also receive regular software updates to maintain and maximize your Alcatel product’s
features and functionality and on-site hardware replacement through our global network of highly
qualified service delivery partners. Additionally, with 24-hour-a-day access to Alcatel’s Service and
Support web page, you’ll be able to view and update any case (open or closed) that you have reported
to Alcatel’s technical support, open a new case or access helpful release notes, technical bulletins, and
manuals. For more information on Alcatel’s Service Programs, see our web page at
www.ind.alcatel.com, call us at 1-800-995-2696, or email us at support@ind.alcatel.com.
This manual documents Release 4.5 Voice over IP (VoIP) hardware and software.
The functionality described in this manual is subject to change without notice.
Alcatel® and the Alcatel logo are registered trademarks of Alcatel. Xylan®, OmniSwitch®, PizzaSwitch® and
OmniStack® are registered trademarks of Alcatel Internetworking, Inc.
AutoTracker™, OmniAccess™, OmniCore™, Omni Switch/Router™, OmniVista™, PizzaPort™, PolicyView™,
RouterView™, SwitchManager™, SwitchStart™, VoiceView™, WANView™, WebView™, X-Cell™, X-Vision™
and the Xylan logo are trademarks of Alcatel Internetworking, Inc.
SM
All-In-One
of their respective companies.
is a service mark of Alcatel Internetworking, Inc. All other brand and product names are trademarks
26801 West Agoura Road
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info@ind.alcatel.com
US Customer Support–(800) 995-2696
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Internet–http://eservice.ind.alcatel.com
Cautions
FCC Compliance
digital device pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable
protection against harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses, and can radiate radio frequency energy and, if not
installed and used in accordance with the instructions in this guide, may cause interference to radio
communications. Operation of this equipment in a residential area is likely to cause interference, in
which case the user will be required to correct the interference at his own expense.
The user is cautioned that changes and modifications made to the equipment without approval of the
manufacturer could void the user’s authority to operate this equipment. It is suggested that the user
use only shielded and grounded cables to ensure compliance with FCC Rules.
This equipment does not exceed Class A limits per radio noise emissions for digital apparatus, set out
in the Radio Interference Regulation of the Canadian Department of Communications.
Avis de conformité aux normes du ministére des Communications du Canada
Cet équipement ne dépasse pas les limites de Classe A d’émission de bruits radioélectriques pour les
appareils numériques, telles que prescrites par le Réglement sur le brouillage radioélectrique établi
par le ministére des Communications du Canada.
Lithium Batteries Caution
incorrectly replaced. Replace the battery only with the same or equivalent type of battery
recommended by the manufacturer. Dispose of used batteries according to the manufacturer’s
instructions. The manufacturer’s instructions are as follows:
Return the module with the Lithium battery to Alcatel. The Lithium battery will
be replaced at Alcatel’s factory.
: This equipment has been tested and found to comply with the limits for Class A
: There is a danger of explosion if the Lithium battery in your chassis is
This chapter describes Alcatel’s H.323 Voice over IP (VoIP) gateway and how telephone or
fax calls can be programmed to automatically go through either an enterprise’s Virtual Private
Network (VPN) via the gateway, and/or the Public Switched Telephone Network (PSTN), a
circuit-switched public telephone network that consists of all the interconnected calling
networks in the world.
Alcatel’s H.323 VoIP gateway provides transparent, seamless delivery and connection of local
and long distance, inbound and outbound telephone calls initiated through standard PSTN
North American (T1), European (E1), and Euro ISDN (Integrated Services Digital Network)
digital telephone transmission lines. For specific details on the precise types of calls handled,
see Chapter 2, “VoIP Daughtercards” and Chapter 5, “VoIP Commands.”
As with standard T1, E1, ISDN (Euro) telephone service, VoIP calls can be transmitted fullduplex (simultaneously in both directions). Likewise, Alcatel’s H.323 VoIP gateway digitizes
phone or fax call signals and uses one of these call transmission services, depending on the
type of call, to channel the calls, i.e., carry information to a destination point in the VoIP
network. Depending on the configuration of the VoIP network, the calls may also go through
the PSTN. For more details on the digitizing process, see
Signal Processing
This chapter provides general background information on VoIP networks, clients, gateways
and gatekeepers, and includes a list of key features provided specifically by Alcatel’s H.323
VoIP network. A VoIP call scenario is illustrated and described along with details on the technologies used in VoIP to explain how calls can be placed over IP. Elements of converged
VoIP network are also shown and described, and significant telephone and data communications technologies are explained in relation to the VoIP gateway. Technical standards for the
most prominent technologies used in Voice over IP are briefly summarized at the end of this
chapter, since as a whole, H.323 ITU standards define the major components using VoIP technologies in network-based VoIP communications systems.
on page 1-12 for more details.
Payload Packetization and Digital
VoIP Networks
Alcatel’s H.323 VoIP gateway for packet-switched IP networks combines the speed, versatility
and low cost of IP telephony with standard telephone features for enterprises in North American and Europe (this necessarily entails other continents or countries, such as Mexico, that
may have the same requirements). These networks are referred to as VoIP networks. Because
data networks as such usually operate with extra carrying capacity (bandwidth), most IP
networks are easily able to accommodate voice/fax traffic once the VoIP network is set up.
The Internet Protocol (IP) is used mainly because it is supported over many layer 1 and 2
network technologies including Ethernet (10, 100, 1000 Mbps), Token Ring, FDDI and Frame
Relay to name a few, including leased lines and satellites. Nearly every router, frame relay
device, and network switch used today supports the Internet Protocol. IP delivers any transport media used between local and wide area networks.
Enterprise IP networks consist of local area networks (LANs) installed at corporate offices
often joined together by corporate wide area networks (WANs). Usually the local area
networks support IP on various types of standard data communications technologies such as
Ethernet, Token Ring, ATM (Asynchronous Transfer Mode) and FDDI (Fiber Distributed Data
Interface).
Page 1-1
Introduction
Getting Started with VoIP
Wide area networks are used to support IP connections over leased lines, public frame relay,
ATM, satellite and ISDN. At each branch office location, enterprises use routers to connect the
remote LANs to the IP WAN. When used with Alcatel’s H.323 VoIP gateway, these Virtual
Private Networks, or VPNs, allow a portion of the Public Switched Telephone Network to be
managed and used by the enterprise.
Alcatel’s H.323 VoIP gateway provides the Voice over IP network capabilities by means of
either digital or analog VoIP daughtercards installed in the switch. The VoIP daughtercards
specifically enable enterprises to control the call routing capabilities of their own organizations by using a portion of the PSTN.
Once an enterprise network is ready to provide VoIP using Alcatel’s H.323 VoIP gateways,
Network Administrators can begin setting up VoIP networks by installing and configuring the
appropriate voice switching daughtercard(s). As a whole, Alcatel’s VoIP H.323 gateways can
be scaled from a minimum of two voice channels per switch to a maximum of 120 voice
channels per switch. Switches with the greatest VoIP scalability will use voice switching
modules (VSXs) in Omni Switch/Routers. See Chapter 2, “VoIP Daughtercards,” for further
details on supported configurations and scalability.
Initially, an Alcatel VoIP network dialing scheme (AVNDS) must also be selected and
deployed via a text-based configuration boot file, wherein each daughtercard must be
assigned a unique IP address among other unique gateway identifiers. From that point, operational parameters such as channel and port types can be set using the command line interface (CLI) configuration tool. Comparable text-based (ASCII) configuration boot files may also
be quickly generated to configure multiple VoIP-enabled switches with similar requirements.
Also, stored in the
vsmboot.asc
files are voice coding parameters which are pre-configured
and kept in profiles. Coding Profiles are configured directly to the components, and define
which operational VoIP characteristics will be used, and then implemented according to the
instructions contained in the profiles. Coding Profiles consist of general caller information,
voice and fax transmission, coding/decoding settings. Preferred Coding Profiles can be automatically selected based upon payload requirements. Coding Profiles are configured at the
channel level. VoIP configurations for VoIP callers are established by setting up profiles and
then assigning the profiles to each individual H.323 VoIP gateway or daughtercard. Profiles
can be created, modified, copied and deleted using one of the available configuration tools. It
should be known that in most circumstances, the default settings for the Coding Profiles are
sufficient.
Additional parameters that require configuration include calling Destinations and Network
Numbering Schemes, the latter being comprised primarily of Numbering Plans, Phone Groups
and Hunt Methods. Altogether, use of these parameters enable VoIP networks to translate IP
addresses from telephone numbers, and allow communications between the VoIP branch
offices to be configurable. For more details, see Alcatel VoIP Network Dialing Schemes
(AVNDS) on page 1-15.
See Chapter 5, “VoIP Commands,” for details on using these CLI commands once the H.323
VoIP gateway is configured; refer to this chapter as well if using an optional third-party gatekeeper (server) i.e., NT100 RADVision, on a PC for example, as some additional network
parameters will need to be set.
For details on configuring the AVNDS, see “Chapter 3, “Network Dialing Schemes” and Chapter 5, “VoIP Commands.” For details on installing the cards and setting up VoIP H.323 Gateways, see Chapter 4, “Setup and Installation.”
Page 1-2
♦
Introduction
Alcatel’s H.323 VoIP Gateway Key Features
Alcatel’s H.323 VoIP gateways, which connect voice and data networks, minimize call
complexity and dependency on leased telephone lines by allowing enterprises more control
over their own call processing. Alcatel’s H.323 VoIP gateway is used to transport digitized
voice conversations over IP local area networks, which are then sent over wide are networks
using such protocols as Frame Relay or ATM. All VoIP daughtercards are compatible with the
Alcatel OmniAccess 512 and Omni Switch/Router. As shown below, the following features of
Alcatel’s H.323 VoIP gateway are supported in this release.
• Digital T1/E1 voice and fax transport over IP networks
• T1 and E1 telephony interface links to digital Private Branch Exchanges (PBXs) via digital
• Foreign Exchange Station (FXS) telephony Loop interface via analog VoIP daughtercard
with (FXS) grand-daughtercard (variations includes FX Office— FXO).
• H.323 Network Call Control Gateway (establishes Local Area Network (LAN) terminal links;
performs call setup and voice translation functions; provides communications procedures
between LANs)
• Voice Codecs: Pulse Code Modulation (G.711), Internet Speech (G.723.1), Standard Tele-
phone Quality (G.729A), Realtime Fax over IP (Fax T.38).
• Non-Voice Signal Monitoring, Detection and Transmission Protocols:
•Dual Tone Multi Frequency/Modem Fax Relay
•Fax Transparency and Fax Relay
•Modem Transparency and Modem Relay
• PSTN Fallback via Deadman Relay Switch
•The H.323 VoIP gateway is capable of providing PSTN fallback for VoIP calls in the
event of a power failure in the VoIP network by means of a Deadman relay switch on
the digital VoIP daughtercards. For more information on the Deadman switch, see
Chapter 2, VoIP Daughtercards.”
• Echo and jitter controls on digital VoIP daughtercards.
• Pre-configured, modifiable AVNDS (Alcatel VoIP Network Dialing Schemes) with corresponding text-based (ASCII) configuration boot files (
vsmboot.asc
files).
• VoIP Text-based Command Line Interface (CLI) configuration tool.
Note ♦
When used separately, the terms E1 and ETSI both
entail European PRI and BRI interfaces. E1 ETSI used
together as one term refers specifically to Euro PRI.
Page 1-3
VoIP Telephone Calls
VoIP Telephone Calls
H.323 VoIP telephone calls, which can carry either voice, facsimile, or modem transmissions
over IP networks, are switched to the packet-based network and connected to the calling
destination (an IP device) via a unique IP address and local/remote dialing plan (actually two
Alcatel VoIP Network Dialing Schemes rolled into one). The numerical IP address, also serving to identify calls intended for VoIP networks, is determined and translated from a destination telephone number in a phone directory database while it is being entered, and the call is
in progress. (It should be noted here that callers do not need to remember the IP addresses,
only the called party or destination phone number). See Chapter 3, “Network Dialing
Schemes,” for more information on the AVNDS.
H.323 VoIP telephone calls are transparent so callers don’t have to worry about any special
procedures, except being aware of a dialing plan that may require them to dial a prefix, such
as 7, before a call can be placed across the VoIP network. This would be similar to current
dialing plans requiring callers to dial 9 before an office call can be placed (9 is the prefix
most often used by PBXs to access the PSTN).
VoIP calls initiated from standard telephone handsets after a preset number of digits are
dialed, for example, can be immediately transmitted using IP data networks whereby digital
or analog signals, meant to set up connections for carrying information, are intercepted by
Alcatel’s H.323 VoIP gateways in the network. These gateways translate the phone numbers
into IP addresses, convert the information to digital packet form, and then deliver the calls
over the network and the PSTN as shown below.
A VoIP Call Scenario
Once a VoIP network is set up a typical VoIP call scenario might go something like this.
Local Telephone Number
Remote Telephone Number
VPN
Step
Call setup begins
1
Source
IP Address
Dial Tone
VoIP H.323 Gateway
PBX #1
VoIP H.323 Gateway
PBX #2
Destination
IP Address
Page 1-4
Dial
Destination
Number
VoIP Call Scenario — Step 1: Call Setup
VoIP Telephone Calls
Local Telephone Number
Step
Session setup with remote gateway
2
Source
IP Address
VoIP H.323 Gateway
VPN
Dialed digits
translated to
IP address
VoIP H.323 Gateway
PBX #1
VoIP Call Scenario — Step 2: Call Progress
Remote Telephone Number
Destination
IP Address
PBX #2
As the caller dials, the H.323 VoIP gateway collects the dialed digits and then ultimately translates the digits using a pre-configured Numbering Plan and Phone Group into the IP address.
A VoIP session is then initiated with the remote gateway (when gatekeeper not used).
After the gateways determine that the VoIP call can be placed across the IP network, the gateways negotiate call capabilities using preconfigured coding profiles, and then optionally strip
before sending the extension digits from the local to the remote gateway where they are
delivered either to the phone, PBX, or keyset. The call can be processed as either a local or
long distance call depending on how the remote gateway is configured. A ringing or busy
signal is transmitted to the caller once the call is connected. If the call is answered, the gateway sends the voice or fax transmissions. If the wide area network is unavailable, calls may
not go through, in which case callers receive a busy signal.
When a caller hangs up the receiver, the VoIP call session is terminated. Multiple gateway
trunks may be used for all calls except those initiated from keysets which must go directly to
the gateway.
Page 1-5
VoIP Telephone Calls
Local Telephone Number
Step
3
Source
IP Address
VoIP H.323 Gateway
Remote Telephone Number
VPN
Call setup completed
VoIP H.323 Gateway
PBX #1
IP call
VoIP Call Scenario — Step 3: Call Setup Completed
PBX #2
(message
returned)
Destination
IP Address
Phone
Rings
Local Telephone Number
Step
4
Source
IP Address
VoIP H.323 Gateway
VoIP Call Scenario — Step 4: Remote Call Answered as VoIP Call
PBX #1
VPN
Call answered
VoIP H.323 Gateway
IP call
Remote Telephone Number
(message
returned)
Destination
IP Address
PBX #2
Page 1-6
♦
Elements of a Converged Network
Elements of a Converged Network
Alcatel’s H.323 VoIP gateway is based on a complex, dual-technology infrastructure taken
from what have been in the past two fairly distinct industries — namely, Telecommunications
(a.k.a. Telephony) and Data Communications. It converges voice and data into enterprise,
Internet Service Providers (ISPs), and carrier networks to provide various levels of VoIP
services using intelligent switches in order to generate long-term cost reductions for telephone services between sites.
The standard or key elements of a converged H.323 VoIP network are described below and
shown in the illustration
Circuit-Switched PSTN
representation of all the various devices that may be used in a VoIP network and how they
may interconnected. VoIP network interoperability is based on ITU H.323 network call control
standards and multiple vocoder support. See also abbreviated
tion Union (ITU) Standards
By means of either digital or analog VoIP daughtercards installed in Alcatel switches, the basic
elements required for providing enterprise H.323 VoIP gateways in packet-switched IP
networks are readily accommodated, including the client, the gateway, and the gatekeeper as
described.
Elements of Converged Voice/Data Packet-Switched VoIP Network and
on page 1-9. This illustration is intended to provide a sample, visual
International Telecommunica-
on page 1-17.
VoIP H.323 Client
The Client is the device initiating and/or receiving the call. This can be a standard telephone
handset or some other H.323 VoIP-capable device in an IP network.
VoIP H.323 Gateway
Alcatel’s H.323 VoIP Gateway is the device used to make the transition from the packetized
voice network to a circuit-switched network, e.g., PSTN, and back. Functionally, the enterprise VoIP gateway is comprised of voice to IP network converter components, .e.g, DSPs, on
the voice switching daughtercards. In VoIP, the process for call placement is the same as in a
service provider system except that the gateway is accessed from Customer Premise Equipment (CPE) instead of from a local service provider, e.g. CLEC (Certified Local Exchange
Carrier).
Note ♦
PBX and Key Systems setup, installation and configuration procedures are beyond the scope of this manual.
Gateway devices intercept then direct electric signals between networked devices. With VoIP,
gateways translate transmission formats between voice CPE and H.323 IP network call control
endpoints and terminals, including communications procedures between gateways. They also
translate between codecs, perform call setup/teardown on LANs and on circuit-switched telephone networks. Gateways are entrance and exit points into VoIP networks that without
hardwiring perform code and protocol conversions, as well as signal filtering.
VoIP gateways contain a user-definable phone directory database of phone number to IP
address mappings; this is called an Alcatel VoIP Network Dialing Scheme (AVNDS). See Chapter 3, “Network Dialing Schemes,” for details. Modifications to the local phone directory database are downloaded through the IP network to the switch, and may be accessed using the
VoIP configuration interface. The phone directory database is built as the VoIP network is
configured, and is contained in the VoIP configuration boot file (
plans, phone groups and destinations as part of the AVNDS comprise a portion of the phone
directory database used by Alcatel’s H.323 VoIP gateway.
vsmboot.asc
). Numbering
Page 1-7
Elements of a Converged Network
Gateways are considered H.323 terminals or H.323 endpoints in H.323 IP networks. Terminals are also the endpoints where telephone lines connect to network circuits. Terminals
provide real time, two-way communications for local area network (LAN) endpoint destinations. All terminals as such must support voice communications and H.245 in-band call
controls to use and negotiate channels. See also abbreviated International Telecommunica-
tion Union (ITU) Standards
VoIP H.323 Gatekeeper (Optional)
The H.323 Gatekeeper (server or workstation) is the device that verifies client VoIP privileges
and translates telephone numbers into IP addresses. It should be noted that H.323 gatekeepers are not required to use Alcatel’s H.323 VoIP Gateway. In lieu of an H.323 VoIP gatekeeper, Alcatel’s H.323 VoIP gateway uses its patent-pending Alcatel VoIP Network Dialing
Scheme (AVNDS) to perform IP address translations.
Gatekeeper setup, installation and configuration
procedures are beyond the scope of this manual.
on page 1-17.
Note ♦
♦
Alcatel recommends and has tested extensively use
of Alcatel’s H.323 VoIP gateway with the NT100
RADVision Gatekeeper.
Gatekeeper devices identify, track and control traffic flowing through them, and perform
other functions such as gateway registration, admission and bandwidth controls.
Page 1-8
VOICE
Gatekeeper
Elements of a Converged Network
LAN
Clients
Central Site
BRI
Telephone
PBX #1
4400
Ethernet
OmniPCX
H.245
VoIP H.323 Gateway
E
1
B
R
I
Digital
Packet-Switched
VoIP Network
WAN
Euro
ISDN
ISDN
PSTN
Ethernet
Ethernet
VPN
WAN
VoIP H.323 Gateway
T
1
WAN
Circuit-Switched
NO. AMER.
PSTN
H.245
Microsoft
NetMeeting
IP Address
Remote Site
PSTN
Fallback
(Deadman
Relay Switch)
Client
Key System
POTS/
PSTN
Analog
VoIP H.323 Gateway
FXO/
F
XS
Elements of Converged Voice/Data Packet-Switched VoIP Network
and Circuit-Switched PSTN
Page 1-9
H.323 VoIP Gateway Voice and Convergence Features
H.323 VoIP Gateway Voice and Convergence Features
As shown below, the main functions handled by Alcatel’s H.323 VoIP Gateway include the
following:
•Telephony Signaling
— used to communicate with the PSTN or Customer Premises
Equipment (CPE).
•Payload Packetization and Digital Signal Processing (via DSP)
— converts PCM voice
packets from circuit-switched network to H.323 packets on IP network and the
reverse.
•H.323 Network Call Control
— handles H.245 and H.225 packet processing, e.g.,
connect, disconnect.
•Alcatel VoIP Network Dialing Schemes (AVNDS)
— handles conversions between
phone numbers and IP address of H.323 devices.
•Network
Switch Backplane Interface — connects H.323 VoIP gateways to switch and
ultimately to IP network.
These functions can generally be divided into either voice or convergence features, based on
the controls they provide over VoIP in the switch. For the most part, the voice features
include separate controls for signaling and for voice interoperability, whereas, the convergence features encompass H.323 call control and voice/data interoperability via the use of
AVNDS in IP networks.
Alarms
Alarms
PSTN,
PCX/PBX,
BRI phone
Alarms
Signaling
Voice
Ports
Payload
Telephony Signaling
(Digital or Analog)
DTMF
(Digit
Collector)
Daughtercard Activation
Payload Packetization
(Voice, Fax, Modem)
Configuration
Alcatel
Voice Network
Dialing Schemes
(AVNDS)
VoIP Network
Call Control
(H.323)
VoIP Daughtercards and Enterprise VoIP Features
Control
Packets
Network
Switch
Backplane
Interface
RTP
Payload
Packets
Switch
Bus
Page 1-10
H.323 VoIP Gateway Voice and Convergence Features
Signaling Control and Voice Interoperability (Voice Features)
The ability to accommodate voice traffic using VoIP switches installed in data networks is
achieved by means of signaling controls and voice interoperability features. The VoIP signaling control and voice interoperability functions includes telephony signaling and payload
packetization as described.
Telephony Signaling
Telephony signaling is used for signaling with telephone equipment, e.g., PBX, via the telephony interface, as well as to control the communication signaling between the H.323 VoIP
gateway and the Customer Premises Equipment (CPE). It detects the presence of new calls,
collects dialed digit information (telephone number in some form or another) entered by the
caller to route a call via an AVNDS to its destination point, and is also used to detect the end
of calls (off hook).
Telephony signaling provides call progress supervision by generating supervisory and call
progress tones, as well as DTMF (Dual Tone Multi Frequency) tones for outbound calls. It
also provides DSP (Digital Signal Processor) interfacing control, and transfer of PCM-based
voice packets to and from the DSP subsystem or DIMM (DSP Interface Management Module).
It coordinates with the DSPs to select voice coders (codecs) at startup when a particular
vocoder is needed. When a call is received, telephony signaling is responsible for opening
channels and PCM data streams to the DSPs to process the voice data.
The signaling controls provided by the Telephony Signaling functions includes the following:
•Call Progress Tone and Tone Detection
converts them into tones or other signaling events, e.g., answer or busy signals.
•Dialing Timers
wink start, or how long to wait for another digit.
•E&M Signaling (Common, Wink Start, Immediate Start and Delay Start)
attribute settings or parameters to match CPE. (Available only on VSD-T1.)
•Foreign Exchange Station (FXS) Loop Start
ters to match CPE.
•Foreign Exchange Office (FXO) Loop Start
ters to match CPE.
•Caller ID
between first and second ring.
The telephony signaling configuration options for telephone signaling interfaces, e.g., ring
delay and cadence (ringing rhythm), are assigned to the physical ports on the daughtercard,
including the T1 and E1 line specifications. All options are defined at the channel level.
Parameters for telephony signaling and VoIP network preferences are pre-configured in textbased configuration files referred to as
boot files and subsequently assigned to a daughtercard and/or its components. For more
details on setting these parameters, see also Chapter 5, “VoIP Commands.”
— used to time incoming signaling events, e.g., how long to wait for a
— looks for Caller ID information, e.g., calling party telephone number
vsmboot.asc
— detects individual in-band frequencies and
— customizes
— customizes attribute settings or parame-
— customizes attribute settings or parame-
files. The parameters are stored in the
Page 1-11
H.323 VoIP Gateway Voice and Convergence Features
Payload Packetization and Digital Signal Processing
Payload packetization is responsible for conversion between time-continuous telephony
(analog or digital payload) at the telephony interface and Real Time Protocol (RTP) packets
on the data network interface. It supports voice compression, echo cancellation, Fax and
DTMF Relay (demodulation/modulation), modem data transport (up to 14400 baud), voice
activity detection and comfort noise generation, as well as packet arrival de-jittering.
Physically, the payload packetization function is implemented on the DSPs (DIMMs), with
control and configuration on the Motorola MPC860 processor. Configuration is performed
through the
vsmboot.asc
file on the switch. Upon VoIP daughtercard activation, the configuration is transferred from the switch to the daughtercard. See Chapter 4, “Setup and Installation,” for more information.
The controls for voice interoperability provided by the payload packetization functions
include the following:
•Codecs
(see also Coding Profiles -- H.323 Call Capabilities) — provides encoding/
decoding of H.323 packets.
•Voice Echo Cancellers
•Fax or Modem over IP
•Voice Activity and
— reduces echo on voice conversations.
— allows fax/modem calls to be transmitted via H.323.
Silence Detection — detects voice conversation (or lack thereof) to
reduce H.323 bandwidth requirements.
•Comfort Noise and Jitter Buffer
— generates slight background noise (white noise) on
the voice conversation, so callers do not think the connection has failed.
Digital Signal Processors, or DSPs as they are more commonly known, are math-intensive
coprocessors used to convert and manipulate information, especially in telecommunications
systems (systems that transmit all types of data including voice and video). They are also
programmable chips, well-suited for VoIP as DSPs have the ability not only to convert but to
compress analog signals into various digital formats, i.e., perform digital signal processing.
Although DSPs do not have any direct analog input/output since they are actually digital
devices, they can accept digitized analog data rather than raw analog signals. As a result,
DSPs are used in the digital and analog VoIP daughtercards developed by Alcatel to bring
switch-enabled VoIP to enterprises; however, before the digitized and compressed voice
signals can be delivered as “voice data” in a VoIP network, they must be packetized into
H.323 packets.
Packetized voice is digitized voice compressed into finite bit stream of IP packets, that carry
the “voice payload” between remote and distant locations, across the IP network and make
processing VoIP calls in IP networks possible. Once compressed and packetized, periodic
delays (jitter) to make the call sound smoother must be imposed on the transmission of these
packetized “voice” conversations to mimic “real time voice” (resonating by nature in continuous “analog” waveform). DSPs are used further to reduce the delays from conversion and
compression to ensure quality voice communications without affecting the real time voice
processing and compression that occurs simultaneously.
To transmit the compressed data (digitized voice) across the IP network, the Real Time Protocol (RTP) is used. RTP streamlines and then transports voice packets, including interactive
multimedia packets over IP, although it does so without any guarantees or quality of service
provisioning.
♦
Note
♦
H.323 VoIP telephone calls automatically receive the
highest priority in the VoIP network via the Quality of
Service ToS bit. For more information, see the switch
manual.
Page 1-12
H.323 VoIP Gateway Voice and Convergence Features
Voice packet transmissions, or the “payload,” are expedited by engaging the User Datagram
Protocol (UDP) for faster delivery, packets which by necessity include the IP network call
transport header information. Resultant jitter caused by delays imposed on the payload packets upon arrival to their destinations is also handled by the DSPs.
Layer 2
Header
UDP is needed by RTP to keep pace with “Real Time Voice” but lacks controls and error
checking capability.
DSPs can monitor calls in progress, detect voice activity and handle echo cancellation (the
filtering of unwanted transmission signals as specified in ITU algorithm standards G.160 and
G.126); comfort level background (white) noise can also be generated on either the transmitting or receiving end.
Since digital signal processing affects nearly every operation in VoIP, numerous DSPs are
incorporated adjacent to the supporting MPC860 CPU signaling controller in the voice switching daughtercards (normally used with voice switching modules), comprising the core of
Alcatel’s enterprise VoIP on the call processing end. The DSPs and the Motorola MPC860
controlling processor work in unison to support the various protocols and interfaces that
implement the enterprise VoIP telephony functions contained in software on the voice switching daughtercards. In a nutshell, the DSPs are the voice processors, and the MPC860 controller is the data communications processor on the daughtercards. Altogether, the above
components provide T1, E1 and ISDN voice and data synthesis processing, with scalable
versions of each bringing enterprises any-to-any switching functionality that now, with enterprise VoIP, includes least-cost call routing for VoIP Virtual Private Networks (VPNs).
IP
Header
UDP
Header
Voice Packet Transmission
RTP
Header
Voice/Fax
Data Payload
Signal Recognition
Initially, digital signal processing involves DSP detection of an array of voice signaling types
using Channel Associated Signaling (CAS) repetitive circuit-state signaling protocols (for T1
and E1 lines). Many forms of call signaling exist to set up and end calls, most of which result
in the ringing of a phone or connection of a fax machine. These forms entail newer line
signaling methods that use digital pulses (PCM, or Pulse Code Modulation), analog touchtones such as DTMF (Dual Tone Multiple Frequency), and other much older analog signals in
all their assortments, including but not limited to: Ear & Mouth (E&M), Loop Start, Ground
Start, Foreign Exchange Subscriber (FXS) and Wink Start. Each signaling method was developed through the years by the telephone industry to provide Plain Old Telephone Service
(POTS).
E&M signaling, of which there are five interface types, is the most widely used method for
connecting calls to PBXs, telephone switching systems which use channelized T1 or E1 lines
to transmit signals and multiplex digitized voice. T1 robbed bit signaling is an example of
narrow or in-band signaling — where signaling tones are passed along the same circuit as
someone’s voice.
ISDN (Integrated Services Digital Network), on the other hand, is another type of signaling
wherein voice transmissions are digitized then placed on separate broad or out-of-band channels (so signaling tones are not passed along the same circuit as someone’s voice). This
prevents signaling or other intrusions into the calls, and usually provides faster transmission.
ISDN is a common protocol in the Common Channel Signaling (CCS) network architecture
used for exchanging information between out-of-band signaling networks and telecommuni-
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H.323 VoIP Gateway Voice and Convergence Features
cations nodes in the network. ISDN does not use T1 (or DS-1) robbed bit signaling, where
bits are taken from voice data to carry signaling. Alcatel H.323 gateways support in-band and
out-of-band signaling.
Encoding
Once signaling types are determined they are analyzed and converted by the appropriate DSP
voice coder (vocoder) into digital signals, which are ultimately converted and expanded back
(re-modulated) into real voice. More specifically, after signal recognition and analysis, DSPs
convert (encode) the amplitude of incoming analog signals into digital form using codecs, or
CODer/DECoders.
The basic encoding schemes, or companding methods in use today, are for PCM which
“encodes” analog signals into digital signals. Although the PCM companding methods used for
T1, which follows Mu-Law, and E1 which follows A-Law, differ mainly in their algorithms,
their purpose is much the same. (Companding is a contraction for compression and expansion.) However, A-Law and Mu-Law are incompatible. They use different methods, for example, to sample analog signals.
Next, the digitally encoded signals are compressed using industry standard vocoders. These
are devices that use speech compression/decompression algorithms to analyze and convert
analog waveforms into digital signals and reduce related bandwidth requirements.
Vocoder
G.711
G.723.1
G.729
G.729a
Compressed
Real Time
Protocol
Encapsulation
Analog
DSP
Digitizer
Digital
Voice/Data Encoding and Call Compression
Compression
The appropriate vocoder used for VoIP calls is then negotiated by the H.323 VoIP gateway
prior to call placement. As an added bonus, but with some variations in protocols, the same
DSP technology that is used for voice compression also works with fax modems. (For that
reason, it can be assumed that references to voice signal packets inherently include “fax”
packets.)
The codecs and vocoders used in enterprise VoIP adhere to the ITU recommendations that
fall under the H.323 IP network call control umbrella of interoperability standards for multimedia communications over packet-switched local area networks (part of the Series H Recommendations for Audiovisual and Multimedia Systems). The ITU’s H.323 suite of specifications
includes the H.245 in-band call control specifications. For the signaling vocoders (G.711, PCM;
G.723.1, Internet Speech; G.729/G.729a; Standard Telephone Speech), the algorithms in the
Series G Recommendations for Transmission Systems and Media, Digital Systems and
Networks are used.
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H.323 VoIP Gateway Voice and Convergence Features
H.323 Call Control and Network Interoperability (Convergence Features)
The ability to accommodate voice (also fax and modem) traffic compressed into data form via
payload packetization for transport across data networks is achieved through the use of H.323
call controls and Alcatel VoIP Network Dialing Schemes (AVNDS) as described.
H.323 Network Call Control
H.323 network call controls are responsible for the procedures and protocols necessary to
establish/tear down VoIP calls across the IP network. The VoIP gateway implements the
H.323 network call control standards, which include the following:
•The H.225/Q.931 protocol that performs call establishment and tear down by establishing a reliable call signaling channel.
•The H.245 protocol that establishes a reliable H.245 in-band channel for communications between all endpoints or terminals, i.e., gateways, for capability exchange and
other messages.
•The registration, admission and status (RAS) protocol that creates a RAS channel to
carry RAS messages between an endpoint and a gatekeeper.
•The H.323 IP network call control standards that support multimedia communications
over local area networks.
The H.323 network call controls and capability provided by the H.323 call control functions
includes the following:
•H.323 gateway (VoIP Switch)
•H.323 gatekeeper (e.g., RADVision Server) (Discovery, Configuration and Operation)
The voice network configuration options include general network information, H.323, H.225,
and H.245 configuration settings, gateway, gatekeeper and registration parameters, and Real
Time Transport Protocol (RTP) session parameters that must be specified for IP network
communications. Voice network call control parameters are configured at the VoIP daughtercard level.
Alcatel VoIP Network Dialing Schemes (AVNDS)
The AVNDS are responsible for the operations and configuration of the VoIP daughtercard
and/or voice switching module, e.g, VSX. AVNDS are implemented on the Motorola MPC860
processor, and the switch.
AVNDS are responsible for providing the interface to configure and maintain all VoIP daughtercards (H.323 gateways) on the entire VoIP network. Additionally, both standard packet
Management Information Bases (MIBs) and proprietary voice packet MIBs are supported.
The AVNDS are used to store information contained in VSM (Voice Switching Module) configuration boot files (
larly the following:
vsmboot.asc) concerning the configuration of the VoIP network, particu-
•Destinations (H.323 endpoints, H.323 local channel destinations)
•Phone Groups (e.g., strip digits and extensions)
•Numbering Plans (hunt methods and hunt groups)
Page 1-15
H.323 VoIP Gateway Voice and Convergence Features
The AVNDS handle inbound/outbound calls routing to/from the VoIP network and local
ports. AVNDS are also used to set up calls and translate IP addresses to telephone numbers,
and can be used with, or in lieu of, H.323 VoIP gatekeepers.
VoIP configuration boot files and profiles simplify VoIP configuration of Alcatel’s H.323 VoIP
gateways (VoIP daughtercards) by using sets of pre-configured parameters that can be
assigned to the various manageable components. Various configuration elements, e.g.,
profiles, have a user-defined name associated with it. VoIP daughtercard configurations are
stored in the switch.
Destinations which consist of remote network and local calling gateways, including H.323
gatekeepers, allow Network Administrators to configure a destination IP address and its
specific protocol. Local channel destinations are considered subdestinations. Destinations,
which are appended to hunt methods, are configured at the daughtercard level.
Phone groups are used to indicate what telephone numbers are available. They also define
digits to be stripped and forwarded. Phone groups are configured at the daughtercard level.
Voice numbering plans use hunt methods to arrange telephone lines so that when calls come
into the network they will ring in a certain order. For example, to use PSTN fallback, all
phone groups must be set up with the last group element indicating the local destination or
gateway to fall back on when a call cannot be placed over the VoIP network.
Hunt methods in voice numbering plans are configured at the daughtercard level. Hunt methods dictate what to do if the first line tried is busy, i.e., hunt methods are used to track down
lines in a certain order until an available line is located. Phone line destinations can be
grouped as desired in user-defined groups, such as by divisions or departments, location, or
some other meaningful grouping.
For information on setting up and using the AVNDS (Alcatel VoIP Network Dialing Schemes),
see Chapter 3, “Network Dialing Schemes,” Chapter 4, “Setup and Installation,” and Chapter 5,
“VoIP Commands.”
Switch Backplane Interface
The switch backplane interface is responsible for the payload packet transport, and VoIP
daughtercard management message transport between the VoIP daughtercard and the host
switch. Physically, the interface consists of a 100-pin connector between the VoIP daughtercard and the motherboard. All functions of the H.323 VoIP gateway are implemented on the
MPC860 controllers on the daughtercards, the VSX (OSR configurations only) and the switch.
Page 1-16
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