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A L C T E L

Alcatel OmniPCX Enterprise

SECTION

IP overview

Ed.03
A L C T E L

Alcatel OmniPCX Enterprise


Section - IP overview

Section - IP overview

SUMMARY

IP Overview
General information ......................................................................................... . 3
1. IP network overview ................................................................................ . 3
2. IP network and OmniPCX Enterprise....................................................... . 4
3. Speech processing on IP .......................................................................... . 5
4. Modem/fax/data communications........................................................... . 7
5. Quality of Service ..................................................................................... . 8
6. Connection of Call Server to Media Gateways ........................................ . 8
7. The H323 gateway function of the OmniPCX Enterprise ......................... . 9
8. Alcatel IP-Phones..................................................................................... .10
9. Configuring IP equipment........................................................................ .11
10. Summary table....................................................................................... .12
Management - QoS ........................................................................................... .13
1. Definitions................................................................................................ .13
2. Using the ToS / DiffServ field ................................................................... .14
3. 802.1p/Q tagging .................................................................................... .17
4. Summary .................................................................................................. .23
Management - Compression ............................................................................. .25
1. Introduction.............................................................................................. .25
2. Managing system parameters................................................................. .26
3. H323 Calls ................................................................................................ .27
4. VPN hop on IP.......................................................................................... .27
5. Negotiation mechanism........................................................................... .28
6. IP-Phones................................................................................................. .33
7. Inter-Media Gateway Call ....................................................................... .37
8. Bandwidth requirements according to the compression
algorithm ................................................................................................ .37
Management - Call Restrictions ........................................................................ .39
1. Call restrictions within the same node .................................................... .39
2. Restriction by management of IP boards ................................................ .39
3. Restricting calls by VPN overflow ............................................................ .40
Appendix - H323 standard................................................................................ .43
1. H323 Standard Presentation ................................................................... .43
2. Structure of the datagrams...................................................................... .48
3. Summary of H3XX standard characteristics ............................................ .53
4. Voice quality............................................................................................. .53
5. TCP/IP tools (present on a PC running Windows NT) .............................. .56

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Section - IP overview

Glossary............................................................................................................. .59
1. A ............................................................................................................... .59
2. B ............................................................................................................... .60
3. C ............................................................................................................... .61
4. D............................................................................................................... .62
5. E ............................................................................................................... .63
6. F ............................................................................................................... .64
7. G .............................................................................................................. .65
8. H............................................................................................................... .65
9. I ................................................................................................................ .66
10. J.............................................................................................................. .69
11. K ............................................................................................................. .69
12. L ............................................................................................................. .69
13. M ............................................................................................................ .70
14. N ............................................................................................................ .70
15. O ............................................................................................................ .71
16. P ............................................................................................................. .71
17. Q ............................................................................................................ .73
18. R ............................................................................................................. .73
19. S ............................................................................................................. .74
20. T ............................................................................................................. .75
21. U............................................................................................................. .76
22. V ............................................................................................................. .76
23. W ............................................................................................................ .77
24. X ............................................................................................................. .77
25. Y ............................................................................................................. .78
26. Z ............................................................................................................. .78

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Section - IP overview IP Overview
General information

44A01F01000A000AAEN

IP Overview
General information
Edition: 03
The OmniPCX Enterprise architecture has been designed:

- to render the communications control application (Call Server) independent from the physical
platform: the Call Server can be hosted by an S or L rack, an ACT shelf (as from R5.1) as well as
by a PC (appliance server).

- to use the IP network as a privileged means to support voice and signaling.


The advantage of an IP network is as follows:

- It requires the same support for both computer and telephone networks, which reduces investment
cost and simplifies installation process,

- It uses a standard protocol, which allows standard terminals to be connected (especially H323 and
SIP terminals).

1. IP network overview

On an ISDN network, voice is digitized and transported on a channel (or Time Slot) in the form of a
continuous flow of bits. The channel is reserved when the call is set up. Transfer rate is guaranteed.
On an IP network, voice is digitized and possibly compressed and then transformed into datagrams by
the encoder/decoder. These datagrams are then transmitted over the IP network. At reception, voice
is extracted from the messages and the digital flow reconstituted before being sent to the recipient.

Figure 1 : Transmission over ISDN and IP networks

ISDN network
Digital G711

Analog Cofidec Cofidec Analog

Digital G711
Digital G711

IP network
Coder Decoder

Framing Messages Frame extracted

The part of an IP network that is "inside a company is often supported by an Ethernet network. Ethernet
is a competitive type network and voice frames compete with the frames of other applications (file
transfer, etc.).

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IP Overview Section - IP overview
General information

The Ethernet network is not secure, data transfer time (delay) and integrity are not guaranteed. This is
also the case for the entire IP network, where losses and delay depend on congestion and the media
used.

The Ethernet network throughput may vary (10 Mb/s, 100 Mb/s, 1 Gb/s).

2. IP network and OmniPCX Enterprise

In the OmniPCX Enterprise, the IP network is used:

- to link the Call Server to the Media Gateway,

- to establish a link to voice over IP equipment: Alcatel IP-Phones, Alcatel PCMM, H323 terminals
and gateways, SIP terminals,

- to implement an ABC link over IP between two OmniPCX Enterprises or between one OmniPCX
Enterprise and one OmniPCX 4400 (release R4.2 minimum).

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Section - IP overview IP Overview
General information

Figure 2 : Example of IP network usage

H323 terminal SIP set

Call Server Call Server

Node 1 Node 2
IP network

OmniPCX Media OmniPCX Media


Gateway Gateway

e-Reflexe
ACT Media set
Gateway

There are two main standards used to manage voice over IP communications, both are supported by
the OmniPCX Enterprise:

- H323: used for ABC links over IP and for communications with H323 terminals and gateways,

- SIP (Session Invitation Protocol): used for communications with SIP phones or gateways.
H323 and SIP standards are different from the signaling exchange point of view. For voice transport,
both of them use the RTP (Real Time Protocol) and voice is coded using G711, G723 or G729 algo-
rithms (see below).

3. Speech processing on IP

To limit throughput between the sender and the receiver, the system performs the following processing
operations:

- speech compression,

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IP Overview Section - IP overview
General information

- silence suppression.

These operations can be enabled or disabled in management: see module IP Overview - Management
- Compression.

3.1. Speech compression

Compression is ensured by "Codecs", also referred to as "Audio codecs", supported by GD, GA and
INT-IP boards and by IP-Phones and PCMM equipment. These circuits are programmable; the mana-
ger configures the processing standard to be applied. Standards available include:

- G711, no compression, call throughput is then 56 kbps or 64 kbps, according to the case.
They accept the "A" or "µ" law. G711 is recommended when there is no bandwidth problem,
e.g. on a LAN.

- G723.1 and G723.1 Appendix A (with silence suppression), call throughput is then 6.3 kbps.
Audio quality is slightly below that of the public phone network.

- G729 Annex A and G729 Annex A & Annex B (with silence suppression), call throughput is then
8 kbps. Audio quality is similar to that of the phone network.

A default type of compression (G723.1 or G729A) is manage by the PCX. If the default compressor is
not appropriate for a specific access, G711 type processing (no compression) may be locally imposed
on this access.

3.2. Silence suppression

To reduce voice throughput (also referred to as "bandwidth") yet further, the Codecs have a voice activity
detector that allows no data to be transmitted during pauses (silences) in a conversation. This process
would result in total silence at the receiving end. This would be unpleasant for the listener. To avoid
this problem, they incorporate a "background noise generator" that gives the user the impression that
he is using a standard phone line. This "noise" is also referred to as "Comfort noise".

On average, silence suppression enables a coefficient of 2 to be gained in transmitted bit rate.

3.3. Transferring voice flows

Voice flows are transferred on the IP network, except when equipment is connected to the same Media
Gateway.

On OmniPCX Media Gateway, the interface with the IP network is ensured by the GA or GD boards
that uses the MCV8 or MCV24 daughterboard compressors.
On ACT Media Gateway, the interface with the IP network is ensured by the INT-IP A or INT-IP B boards
that uses the GIP4 or GIP6 daughterboard compressors.

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General information

Table 1 : Compressors on IP boards

Daughter boards
GD
OmniPCX Media - MCV24: 21 compressors max.
Gateway - MCV8: 7 compressors max.
GA
- 1 GIP6 board: 30 comp. max.
- 2 GIP6 boards: 60 comp. max.
INT-IP A - 1 GIP4 board: 30 comp. max.
ACT Media Gateway
INT-IP B - 2 GIP4 boards: 60 comp. max.
- 1 GIP6A board: 8 comp. max.
- 1 GIP4A board: 8 comp. max.
In certain cases, according to the equipment communicating, voice (RTP flow) does not transit through
Media Gateways. When two equipments use the same compression algorithm, the RTP flow can be
direct. Only the signaling will be processed by PCXs (see module Direct RTP in ABC network - Functional
description).

Node 1 Node 2

IP network

RTP flow via GA or Direct RTP


IP-Phone IP-Phone
GD board flow

In the above figure, voice flow between two IP-Phones do not transit through Media Gateways and is
directly routed between terminals.

4. Modem/fax/data communications

In addition to voice communications, the following is possible:

- fax Relay over IP (group 3 fax) communications, whose implementation may be:
• proprietary, with an inter OmniPCX 4400 or OmniPCX Enterprise configuration. Maximum
speed: 9600 bits/s,
• standard (H323 standard, T38 procedure): allows faxes over IP with non-OmniPCX
4400/OmniPCX Enterprise H323 gateways, such as the OmniPCX Office or from other
manufacturers (see module ABC links through IP - Management - § Fax parameters).

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IP Overview Section - IP overview
General information

- as from R5.1, transparent modem/fax/data communications (see module Modem and data
transparency over IP - General information):
• within the same node (or the same domain) for transparent modem/fax communication,
• over an entire ABC network for transparent data communications.

Note: fax relay and transparent modem/data communications cannot be performed simultaneously
on the same system. These latter communications must be authorized or prohibited in management.

5. Quality of Service

The IP network supports both voice and data equipment. This is why mechanisms have been developed
to give preference to voice frame transfer (voice frames must be transferred in real time) over data
frame transfer and to separate voice and data flows.

Such mechanisms are defined by the following standards:

- 802.1p: Specifies the marking of level 2 frames of the ISO model to enable IP equipment to
process them according to the desired priority. High priority frames (voice) are processed before
low level frames (data) level 2 equipment,

- 802.1Q: Allows a VLAN (Virtual LAN) number to be defined to separate voice flow from data flow.
The principle is to define two separate VLANs containing the voice over IP equipment on one hand
and data equipment on the other hand. Each VLAN domain represents a broadcast domain. For
example, a broadcast message transmitted by an IP-Phone is only broadcast within the IP-Phone
VLAN and will not be received by data equipment (PC) which is part of a different VLAN.

- Diffserv: Allows a priority level to be assigned to level 3 frames of the ISO model.

For more information on QoS management, see module IP Overview - Management - QoS.

6. Connection of Call Server to Media Gateways

In the case of a Call Server on CS board or on Appliance Server, the link to Media Gateways is over IP.
From the Call Server side, there is an inter-shelf link over IP, the Call Server being the main shelf (shelf
0) and the Media Gateway being the peripheral shelf.

On an OmniPCX Media Gateway, the link is established on the GD board (rack controller board).

On an ACT Media Gateway, the link is established on an INT-IP B board. There may be one or two
INT-IP B boards: one is the active board and the other is the backup board.

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General information

OmniPCX Main
Media INT-IP B
Gateway ACT Media
Backup Gateway
Call INT-IP B (M2 cabinet)
GD
Server

IP network
Remote IP link

A Call Server hosted on CPU5 or CPU6 board is linked with the main ACT boards via link C1 and a
group of shelves can be connected via a wired link (RT2, INTOF). There can also be remote ACT or
OmniPCX Media Gateways over IP (see module Topology - General information).

7. The H323 gateway function of the OmniPCX Enterprise

A gateway is used for the interconnection of two networks using different protocols e.g. an IP network
and the telephone network behind the PCX.

The OmniPCX Enterprise H323 gateway function allows:

- an IP link between two PCXs on a same ABC network to be set up,

- another H323 gateway to be reached (for example: the IP board of an OmniPCX Office or
the H323 gateway of another manufacturer),

- an H323 terminal to be reached.

Important:
- in R5.0, the OmniPCX Enterprise H323 gateway function support only voice and data
communications (proprietary fax and T38 fax). Data communications (modem, TA and minitel) are
not supported,
- as from R5.1, ABC links over IP support transparent data communications.

The H323 gateway function is ensured by GA, GD and INT-IP A boards. The H323v2 standard
applies to exchanges.

Compressors are provided by:

- MCV24 or MCV8 daughterboards installed on GA or GD boards,

- GIP4, GIP6, GIP4A, GIP6A daughterboards installed on INT-IP A boards.

IP-Phones and H323 terminals have their own compressors.

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IP Overview Section - IP overview
General information

Node 1 Node 2

OmniPCX Office
GD GD
Call Server IP board
UAI

IP trunk group to H323


ABC link H323 sig. gateways and terminals IP network
ABC-F2 sig.

Call Server

H323 terminal

H323 calls are made using the E164 numbering scheme (standard dialing). E164 number translation
into IP addresses is carried out:

- either internally in the PCX, via speed dialing numbers,

- or using a gatekeeper that may be external or internal to the PCX.

For more information, see the

- module ABC links through IP - General information on how to configure an IP link,

- module H.323: terminals, gateway, gatekeeper - General information on how to declare a H323
terminal and a gatekeeper.

8. Alcatel IP-Phones

Alcatel IP-Phones are:

- IP-Phones V2 (e-Reflexe sets):


• A4010-IP: see module 4010IP set (Easy e-Reflexes) - Hardware description.
• A4020-IP: see module 4020IP set (Premium e-Reflexes) - Hardware description.
• A4035-IP: see module 4035IP set (Advanced e-Reflexes) - Hardware description.

- IP-Phones V1 and V1S: association of an UA3G (4020 or 4035) set and a TSC-IP V1 or TSC-IP
V1S adapter (see the module TSC-IP V1 (4098RE) - Hardware description and the module TSC-IP
V1S (4098FRE) - Hardware description).

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General information

Table 2 : Alcatel IP-Phones

Type of IP-Phone Set name Ethernet access


A4022 (A4020+TSC-IP V1) 10 Mb/s
IP-Phones V1
A4037 (A4035+TSC-IP V1) 10 Mb/s
A4022F (A4020+TSC-IP V1S) 10/100 Mb/s
IP-Phones V1S
A4037F (A4035+TSC-IP V1S) 10/100 Mb/s
A4010-IP 10/100 Mb/s
IP-Phones V2 (e-Reflexe) A4020-IP 10/100 Mb/s
A4035-IP 10/100 Mb/s

IP-Phone and PCMM signaling is processed:

- either directly by the Call Server,

- or via a GD or a GA board of the same domain.

For more information on implementing Alcatel IP-Phones, see the module IP-Phones - General infor-
mation.

9. Configuring IP equipment

All IP equipment has IP settings: IP address, subnetwork address, router address.

These settings can be configured:

- dynamically: the settings are automatically assigned by a DHCP server: used, for example, by the
GD and INT-IP B boards,

- statically: for example, for a GA board, the settings are assigned by the board management on
the PCX.

The DHCP server is an equipment that defines IP address ranges that are available to equipment that
requests them, with other information (TFTP server address) used to initialize equipment.

The DHCP server can be:

- external: e. g. a DHCP server on a Windows PC (NT4 and 2000),

- internal: The Call Server has its own DHCP server that can be activated or not (see the module
DHCP Server - Functional description).

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General information

Table 3 : Type of configuration (dynamic/static) for the Alcatel IP equipment.

Dynamic configuration Static configuration


GD board Yes Yes (via V24, UA set or telnet)
GA board No Yes (via the Ethernet settings
management of the board)
INT-IP B Yes Yes (via V24)
INT-IP A No Yes
IP-Phones Yes Yes (via the supervisor menu of the
set when it is initialized)

10. Summary table

Table 4 : Implementing the different functions by the PCX components

OmniPCX Media
Call ACT Media Gateway
Gateway
Server
GA GD INT-IP A INT-IP B
H323 gateway (internode link and
x x x
link to H323 equipment)
SIP Gateway x
Signaling transfer for IP-Phones x x x x
IP compression for H323, SIP and
x x x
IP-Phones
Remote IP link x x x x
Transparent modem/data (as from
x x x x
R5.1)

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Section - IP overview IP Overview
Management - QoS

44A01F01000A160AAEN

IP Overview
Management
QoS
Edition: 03

1. Definitions

The Quality of Service on IP can be implemented at two levels:

- Level 2 QoS (Ethernet) or 802.1p/Q.

- Level 3 QoS (IP), using the ToS or DiffServ field.


The Quality of Service allows flows on the data network to be assigned a priority. QoS applies to
signaling (ABC signaling on hybrid link, IP-Phone UA signaling, H323 signaling between gateways)
and initialization (IP-Phone, GD and INT-IP B board download). it also applies to VoIP flows.

1.1. Level 2 QoS


Level 2 QoS was not initially planned in the Ethernet standard. When it is implemented, it must be
implemented on all network equipment. This is because a 4-byte field is added in the Ethernet frame
header, which renders it incompatible with the 802.3 or Ethernet standard.
This field consists of the TPI (Tag Protocol Identifier), on two bytes, with a value of 81 00 (hexa) to
indicate that 802.1Q is used and of the TCI (Tag Control Information) field, also on 2 bytes, which can
be broken down as follows:

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
User Priority CFI VID: VLAN Identifier

- The User Priority field, on 3 bits, is described in the 802.1p standard. Possible values of this
field are:

Priority Type of traffic


0 Best Effort
1 Background
2 Spare
3 Excellent effort
4 Controlled Load
5 Video
6 Voice
7 Network Control

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Management - QoS

The IEEE provides the following description for the seven traffic levels:
• 0: Best Effort traffic: traffic carried without constraint, as is the case today.
• 1: Background traffic: traffic liable to be carried on the network without any impact on the
users and applications.
• 2: Reserved for future use.
• 3: Excellent Effort traffic: traffic carried with optimum effort.
• 4: Controlled Load traffic: major applications traffic.
• 5: Video traffic: traffic requiring a transmission delay lower than 100 ms.
• 6: Voice traffic: traffic requiring a delay and jitter lower than 10 ms.
• 7: Network Control traffic: traffic required for network management.

- the CFI field (1 bit) indicates the presence of the RIF (Routing Information Field), used for Token
Ring frames transiting via an Ethernet network:
• 0 (default): RIF field absent.
• 1 : RIF field present.

- The VLAN Identifier field, on 12 bits (3 hexa digits), indicates VLAN No. Values 000, 001 and
FFF are reserved. A VID at 0x000 indicates that the TCI field contains only one priority data
item (only 802.1p is used).

1.2. Level 3 QoS (IP)


Level 3 QoS is planned from the start of the Ipv4 standard with ToS field, on 1 byte. This field, which
was not used for a long time because it was not actually needed, was replaced in the standard by the
DiffServ or DSCP (Differenciated Services Code Point) fields. This field only uses the 6 first bits of the
byte reserved for ToS:

0 1 2 3 4 5 6 7
DSCP Not used

2. Using the ToS / DiffServ field

The ToS/Diffserv field value is managed by MAO. For IP-Phones, IP and Call Server boards, manage-
ment is performed via Quality of Service categories.
For signaling on hybrid logical links over IP, the ToS value is managed separately.
For GD and INT-IP B boards, it is also possible to specify a priority level directly on the board. This is
used during board initialization to download binaries.

2.1. IP Quality of Service categories


The ToS field value is managed in the IP Quality of Service category.

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Management - QoS

Object name: IP > IP Quality Of Service COS


Attributes:

IP QoS COS ID : Enter IP QoS Category number (between 0 and 15).


TOS/diffServ : This value must be specified by the person in charge
of the customer network.
To calculate the value to be entered, the fact that the
TOS value is coded on the first three most significant
bits of the TOS field, whereas Diffserv occupies the first
six bits of this field, must be taken into consideration
(see the examples below).

- Example 1 (TOS): in client routers/switches, the TOS field is to be used for VoIP flows and the
value to be used is 5. This means that the "precedence" mechanism is used.
To obtain the value to be entered on the Call Server:
• Convert the hexadecimal value to binary: 5 → 101,
• As the precedence level is coded on the first 3 most significant bits of the TOS field, add three
zeros to the previous result, as follows: 101000,
• Convert 101000 into decimal → 40.
40 is thus the value to be entered in the TOS/diffServ attribute.

- Example 2 (Diffserv): in client routers/switches, the TOS field is to be used for VoIP flows and
the value to be used is B8. This means that the “Diffserv” mechanism is used.
To obtain the value to be entered on the Call Server:
• Convert the hexadecimal value to binary: B8 → 10111000,
• Since IP marking only uses the first 6 bits of the TOS field, remove the last 2 bits from the
previous result as follows: 101110,
• Convert 101110 into decimal → 46.
46 is thus the value to be entered in the TOS/diffServ attribute.
Note: Whether TOS or Diffserv is used, the recommended value is 46.

2.2. IP-Phones
On IP-Phones, the Quality of Service category is linked to the IP domain. The Quality of Service cate-
gory previously modified must therefore be assigned to the IP domain of the IP-Phone.
Object name: IP > IP Domain
Attributes:

IP Domain Number : Enter the IP domain number of the IP-Phone.


IP Quality of Service : Enter the number of the Quality of Service category previously
modified.

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Management - QoS

2.3. Boards
For INT-IP A, INT-IP B, Call Server, GA, and GD boards, the Quality of Service is managed in the
Ethernet parameters of the board.
Object name: Shelf > Boards > Ethernet Parameters
Attributes:

Shelf Address : Enter the shelf number.


Board Address : Enter the board number.
IP Quality of Service : Enter the number of the IP Quality Of Service COS
previously modified.

Note:
- For the Call Server, the Quality of Service category number is managed in the Ethernet parameters
of the virtual INT-IP A boards, located in shelf 19, in positions 1 and 2.
- For A4645 voice mail, the Quality of Service category number is managed on the Call Server in the
Ethernet parameters of the GD board, located in shelf 18, in position 0.
Warning: Changing the number of the IP Quality of service COS causes the board to be reset
automatically.

2.4. INT-IP B board


For voice calls and signaling, the values are managed via MAO as described above.
To download binaries, it is possible to specify a TOS/Diffserv value directly on the INT-IP B board when
it is started:

- Start the board as described in module ACT Media Gateway - Commissioning.

- Connect the 3BA 28112 cable to the front panel of the board.

- During the boot phase, press "Enter" to stop start-up and to access the IP configuration menu of the
board (this menu is used to specify the start mode of the board (static or dynamic), to configure
the IP parameters in static mode as well as the QoS parameters):

- Enter chqual to configure TOS/Diffserv (DSCP):


Config:chqual
UseDscp (0 no 1 yes)
(default 0) :1
Define Default DSCP parameter (between 0 and 63)
(default 0) :46
Use802 (0 no 1 yes)
(default 0):

- Save the configuration, then reset the board.

2.5. GD board
For voice calls and signaling, the values are managed via MAO as described above.

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Management - QoS

To download binaries, it is possible to specify a TOS/Diffserv value directly on the GD board before it
is started:

- Either, by using the mgconfig command, while logged onto the board with the root account:
Select 8. VLAN menu, then 6. Change Diffserv DSCP,
then reboot the board.

- Or, on one of the UA sets of the main shelf, if the GD board is started without an IP connection
cable to the LAN.

2.6. Signaling TOS on hybrid link over IP


Signaling on hybrid link over IP may include TOS information. TOS value is entered in the parameter
IP TOS Sig.
Object name: IP > IP Parameters
Attributes:

System Option : Select IP TOS Sig.


IP TOS Sig : Enter the TOS value (between 0 and 63) used for
signaling on hybrid links. The value to be entered is
the decimal value of the TOS coded on 6 bits.
Example: TOS value is 5:
- Converting this to binary gives 101
- On 6 bits, this gives 101000
- Converting this to decimal gives 40
Note: Default value: 46 (101110).
The meaning is:
- 3 most significant digits: Precedence (priority)=5
- LowDelay bit: set
- Throughput bit: set
- Reliability bit: not set

3. 802.1p/Q tagging

802.1p/Q includes two Quality of Service principles, at level 2 (Ethernet):

- The tagging of the VLAN No., or VLAN ID (802.1Q).

- The priority field associated with VLAN (802.1p).


802.1p/Q tagging is managed by the MAO. For IP-Phones, IP and Call Server boards, management
is performed via Quality of Service categories.
For hybrid link over IP signaling, 802.1p/Q tagging is managed separately.
For devices that retrieve their IP address via DHCP (IP-Phones, GD, INT-IP B), for the DHCP request to
be tagged with the correct VLAN number, VLAN No. and priority information is directly configurable
on the devices.

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From R5.1, a VLAN number may be configured in the parameters of each subnetwork on the DHCP
server: it is used by IP-Phones initializing via the DHCP server. This avoids having to manually configure
each set.

802.1p/Q configurations must be mutually consistent (i.e VLAN number must be the same).

The VLAN value must also be managed in the same way at the level of the switch port to which the
device is connected.

Figure 3 : Example

Router

Switch Switch

VLAN 3 VLAN 3 VLAN 4 VLAN 2 VLAN 2

IP-Phone GD Call Server PC1 PC2


VLAN 3 VLAN 3 VLAN 4 VLAN 2 VLAN 2

3.1. Call Server

- 802.1p/Q activation and VLAN number management are performed by netadmin.


Warning:
- Enabling or modifying VLAN number may result in loss of the network, this should therefore be
done via V24 port and not via telnet.
- If the Call Server is duplicated, enabling or modifying VLAN number must be done manually on
both Call Servers. The netadmin Copy Setup menu must not be used.
Run the command netadmin -m, select 14. 'VLan configuration', then 2. 'VLan feature'
VLan Setup update
=================
Do you want to activate 802.1q frame tagging (y/n default is 'n') ? y
Enter the VLan ID to use (0-4094, default is 0): 3

- Priority level management is performed via MAO in the Ethernet parameters of the virtual INT-IP A
boards, (shelf 19, couplers 1 and 2).

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Warning: If 802.1p/Q has not been activated by netadmin, priority level management via MAO
has no impact.

Operation

- Outgoing frames: when 802.1q tagging is enabled on the Call Server, outgoing frames are
tagged with the configured number.

- Incoming frames: all incoming frames are accepted, even if tagged with a different VLAN number
to that of the Call Server.

Figure 4 : Configuration example

Port 1 Port 2 Port 3 Port 4


Primary VLAN ID PVID1 PVID2 PVID2 PVID4

VLAN 1, 2, 4 VLAN 1, 2 VLAN 1, 2 VLAN 1, 4

Call Server Carte GD IP-Phone


VLAN 1 VLAN 2 VLAN 2 PC1

Frames leaving the Call Server have VLAN number set to 1, they are accepted by ports 2, 3 and 4 of
the switch.
Frames leaving telephone equipment (GD board, IP-Phone) have VLAN number set to 2, they are
accepted by ports 1, 2 and 3 of the switch.
Frames leaving the PC do not have a VLAN number. When they reach the switch, they are tagged with
VLAN 4 (primary port VLAN) and thus accepted by port 1.
Conclusion: The Call Server can dialog with telephone equipment and the PC, but telephone equip-
ment cannot dialog with the PC.

3.2. A4645 Voice Mail

- If the voice mail service is on a dedicated CPU: activate 802.1p/Q and manage the VLAN number
by netadmin on this CPU.
Run the command netadmin -m, select 14. 'VLan configuration', then 2. 'VLan feature'
VLan Setup update
=================
Do you want to activate 802.1q frame tagging (y/n default is 'n') ? y
Enter the VLan ID to use (0-4094, default is 0): 3

- Manage priority level on the Call Server via MAO in the Ethernet parameters of the GD board,
(shelf 18, coupler 0).

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Warning: If 802.1p/Q has not been activated by netadmin, priority level management via MAO
has no impact.

3.3. GD board

- VLAN number management must be performed directly on the GD board before board start-up:
• Either, by using the mgconfig command, while logged onto the board with the root account:
Select 8. VLAN menu, then 2. Change vlan id,
and, if necessary, 4. Change user priority.
then reboot the board.
• Or, on one of the UA sets of the main shelf, if the GD board is started without an IP connection
cable to the LAN.

- A priority level can be managed directly on the GD board, by using one of the two methods
described above.

- A priority level can also be managed via MAO: this level applies after board start-up for RTP/RTCP
flows, Call Server/GD signaling, IP-Phones GD and H323.

3.4. GA board

A Quality of Service category number must be managed at the level of the board Ethernet parameters,
and a VLAN number as well as a priority level in this category must be specified.

3.5. INT-IP B board

- VLAN number and, if necessary, priority level must be managed directly on the board when it
is started:
• Start the board as described in module ACT Media Gateway - Commissioning.
• Connect the 3BA 28112 cable to the front panel of the board.
• During the boot phase, press "Enter" to stop the startup and to access board IP configuration
menu:
• Type chqualto configure VLAN number and priority:
Config:chqual
UseDscp (0 no 1 yes) (default 0) :0
Use802 (0 no 1 yes) (default 0):1
Define Default Vlan ID parameter (between 0 and 4095) (default 0): 2
Define Default Priority parameter (between 0 and 7) (default 0): 5
Config:save

New Configuration saved.

WARNING : Coupler must be restarted to take in account new configuration


Do you want to restart coupler now? (y/n):y
Coupler will reset in 5 seconds !

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- A Quality of Service category number must also be managed at the level of board Ethernet
parameters and VLAN number and priority level in this category must be specified.

3.6. IP Phone

- VLAN number can be configured in two ways, depending on set startup mode:
a. Dynamic start: VLAN number can be directly assigned by the DHCP server at set initialization.
Note:
- VLAN number must be configured in each subnetwork's parameters when the DHCP server is
configured: See the module DHCP Server - Functional description - § Distributing VLAN ID by
DHCP and module DHCP Server - Management - § Creating the subnetwork.
- This VLAN configuration mode implies that switches accept untagged frames because, in
this case, the DHCP request is not tagged.
b. Static start: VLAN number must be assigned in the supervisor menu at set startup (only
possible on V1S and V2 sets):
To access the supervisor menu, at set initialization consecutively enter [i] or Menu, then #.
Select 2. IP Parameters, then 7. Default VLAN,.
By default, there is no managed VLAN, the display indicates: 1. tag disabled.
Enter 1, then enter VLAN number (0 to 4095), and confirm by typing # (to cancel, type *).
The display indicates, for example:

1. Tag (Enabled) 0. Back


2. VLAN id (0003)

Type 0 several times to exit the supervisor menu.


Note: In the set menu, priority is not managed since downloading does not have precedence
over voice frames.

- Priority level and VLAN number are managed via the MAO.

3.7. Configuration in the the MAO


802.1Q tagging is managed in the MAO via Quality of Service COS. This setting is used by the Call
Server, the GA boards and also by remote devices (IP-Phones, INT-IP B and GD boards), once they
have been initialized.

3.7.1. IP Quality of Service categories


Object name: IP > IP Quality Of Service COS
Attributes:

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IP QoS COS ID : Enter IP QoS Category number (between 0 and 15).


802.1Q Used : Select Yes to activate 802.1Q tagging.
802.1Q Priority : Enter the level of priority to be assigned to datagrams
using the access. Priority level is a number between 0
and 7. Priority 0 is the lowest priority. This value must
be specified by the person in charge of the customer
network.
VLAN ID : Enter the virtual LAN network identifier. Entering "0"
disables the VLAN mechanism. This value must be
specified by the person in charge of the customer
network.

3.7.2. IP-Phones
On IP-Phones, the Quality of Service category is linked to the IP domain. The Quality of Service cate-
gory previously modified must therefore be assigned to the IP domain of the IP-Phone.

3.7.3. Boards
For INT-IP A, INT-IP B, Call Server, GA, and GD boards, the Quality of Service is managed in the
Ethernet parameters of the board.
Note:
- For the Call Server, the Quality of Service category number is managed in the Ethernet parameters
of the virtual INT-IP A boards, located in shelf 19, in positions 1 and 2.
- For A4645 voice mail, the Quality of Service category number is managed on the Call Server in the
Ethernet parameters of the GD board, located in shelf 18, in position 0.
Warning: Changing the number of the IP Quality of service COS causes the board to be reset
automatically.

3.7.4. Signaling on hybrid link over IP


For signaling on hybrid logical links over IP, priority value is managed separately.
Object name: IP > IP Parameters
Attributes:

System Option : Select 802.1Q Prio Sig Hyb.


802.1Q Prio Sig Hyb : Enter level of priority, between 0 and 7 (default
value: 0).

3.8. Example of operation


For an example of VLAN use with IP-Phones, see the module IP-Phones - Management - VLAN and the
module IP-Phones - Management examples - VLAN.

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4. Summary

4.1. TOS/Diffserv Management

- Call Server: Manage a Quality of Service at the level of shelf 19, couplers 1 and 2. Specify in
this category the TOS/diffserv field (recommended value: 46).

- 4645 Voice mail: manage a Quality of Service at the level of shelf 18, coupler 0. Specify in
this category the TOS/diffserv field (recommended value: 46).

- GD Board:
• Configure TOS/Diffserv level via the mgconfig command (or via the UA set configuration menu).
• manage a Quality of Service category in board Ethernet parameters. Configure the
TOS/diffserv field for this category (recommended value: 46).

- GA Board: manage a Quality of Service category in board Ethernet parameters. Configure the
TOS/diffserv field for this category (recommended value: 46).

- INT-IP A Board: manage a Quality of Service category in board Ethernet parameters. Configure
the TOS/diffserv field for this category (recommended value: 46).

- INT-IP B Board:
• During the boot phase, configure the TOS/Diffserv level directly on the board, using the
chqual command.

• Manage a Quality of Service category in board Ethernet parameters and configure the
TOS/diffserv field for this category (recommended value: 46).

- IP-Phone: manage a Quality of Service category in the parameters of the domain assigned to
the board. Configure the TOS/diffserv field for this category (recommended value: 46).

Note: For an IP board, the Quality of Service category managed in the associated domain is not
taken into account. The category that counts is that specified in board Ethernet parameters.

4.2. 802.1p/Q management

- Call Server:
• Enable 802.1p/Q via netadmin.
• Specify a VLAN number via netadmin.
• Repeat the operation on the duplicated Call Server.
• Manage a Quality of Service category number in the Ethernet parameters of the INT-IP A virtual
boards (shelf 19, couplers 1 and 2), and specify the desired priority and VLAN number in
this category.

- 4645 Voice mail:

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• If the voice mail service is on a dedicated CPU, activate 802.1p/Q and specify a VLAN number
on this CPU via netadmin.
• On the Call Server, manage a Quality of Service category number at the level of the shelf 18,
coupler 0; specify the desired priority and VLAN number in this category.

- GD Board:
• Specify a VLAN number and, if required, a priority level using the mgconfig command (or via
the configuration menu on a UA set).
• Manage a Quality of Service category number at the level of board Ethernet parameters, and
specify a VLAN number and priority level in this category.

- GA Board:
• Manage a Quality of Service category number at the level of board Ethernet parameters, and
specify a VLAN number and priority level in this category.

- INT-IP A Board:
• Manage a Quality of Service category number at the level of board Ethernet parameters, and
specify VLAN number and priority level in this category.

- INT-IP B board
• During the boot phase, specify a VLAN number using the chqual command.
• Manage a Quality of Service category number at the level of board Ethernet parameters, and
specify a VLAN number and priority level in this category.

- IP-Phone:
• For IP-Phones in dynamic initialization mode: Specify VLAN number in DHCP server parameters.
• For IP-Phones in static initialization mode: at set initialization, specify a VLAN number using
the supervisor menu.
• Manage a Quality of Service category number at the level of set domain and specify VLAN
number and priority level in this category.

Note: For an IP board, the Quality of Service category managed in the associated domain is not
taken into account. The category that counts is that specified in board Ethernet parameters.

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Section - IP overview IP Overview
Management - Compression

44A01F01001A160AAEN

IP Overview
Management
Compression
Edition: 03

1. Introduction

Three coding algorithms can be used for voice transport on IP:

- An algorithm without compression: G711, call throughput is then 56 kbps or 64 kbps, according
to the case. Coding can be performed following “A” law or “µ” law. G711 is recommended when
there is no bandwidth problem, e.g. on a LAN.

- Two algorithms with compression:


• G723.1 and G723.1 Annex A (with silence suppression), call throughput is then 6.3 kbps.
Audio quality is slightly below that of the public phone network.
• G729 Annex A and G729 Annex A & Annex B (with silence suppression), call throughput is then
8 kbps. Audio quality is similar to that of the phone network.
Note: The throughputs indicated here are for voice coding alone. The bandwidth required for
transport is higher since each layer (UDP, IP, RTP) adds its own data. For example G711 requires 84.7
kbps bandwidth on Ethernet (for framing at 30 ms). See § 8 : Bandwidth requirements according to
the compression algorithm.
Depending on management, two or three of these algorithms are available on the OmniPCX Enter-
prise:

- G711, still available.

- The compression algorithm managed by default for the system: G723 or G729.

- If multi-algorithm is validated, the three algorithms are available.


For the various types of calls and equipment, the user can choose between G711 and the system
default algorithm.
The third algorithm (if the multi-algorithm is validated) is only used for calls with non OmniPCX En-
terprise H323 equipment, depending on the result of negotiation.
Note: The gateways use H323 protocol for inter-node calls over IP and for calls with non OmniPCX
Enterprise H323 equipment. However, for reasons of clarity, in the rest of this document, "H323 call”
is only used to refer to calls to non OmniPCX Enterprise H323 equipment. Inter-node calls over IP
are occasionally mentioned as a “VPN hop on IP”, as they are implemented by VPN hop.
Management includes a part at system level, plus specific management for each type of equipment or
link (H323 calls, IP-Phones, links on IP, etc.).

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2. Managing system parameters

The following parameters are managed for the entire system, they must be homogeneous on all
network nodes.

2.1. Default compression algorithm


Here, the manager configures the default compression type used for the entire node.
Object name: System > Other. System Param. > Compression Parameters
Attributes:

System Option : Select Compression type.


Compression Type : Select default compression type:
- G723: compression according to the G723.1
standard (6.3 kbit/s).
- G729: compression according to the G729 Annex
A standard (8 kbit/s).
Note: Compression type must be homogeneous on all network nodes.

2.2. Multi-algorithms
This parameter is used to make both the G723 and G729 algorithms simultaneously available on the
PCX. It is involved in building the list established for algorithm negotiation: if it set to "Yes", both the
G723 and G729 algorithms will be included in the list.
Object name: System > Other. System Param. > Compression Parameters
Attributes:

System Option : Select Multi. Algorithms for Compression.


Multi. Algorithms for : - No: only the default algorithm specified above is
Compression available.
- Yes: both G723 and G729 algorithms are
available.
Note: This parameter must be homogeneous on all network nodes.

2.3. Silence suppression (Voice Activity Detection)


Silence suppression allows the bandwidth used to be reduced. When silence suppression is requested,
the background noise generator is systematically enabled.
Silence suppression is enabled via two system options, one applies to calls coded in G711, the other
to calls coded in G723 or G729.
In the case of an H232 call, silence suppression involves negotiation between the caller and the called
party.

2.3.1. Silence suppression in G711


Object name: System > Other. System Param. > Compression Parameters

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Attributes:

System Option : Select VAD on G711.


VAD on G711 : - Yes: the silence suppression mechanism is
enabled.
- No: the silence suppression mechanism is
disabled.

Note: The value of this parameter must be homogeneous on all network nodes.
Warning: If the system has V2 IP-Phones (e-Reflexes) in G711, this parameter must be left at "No".

2.3.2. Silence suppression in G723 and G729


Object name: System > Other. System Param. > Compression Parameters
Attributes:

System Option : Select Voice Activity Detect on LIO Boards.


Voice Activity Detect on LIO Boards : - Yes: the silence suppression mechanism is
enabled.
- No: the silence suppression mechanism is
disabled.

Note: The value of this parameter must be homogeneous on all network nodes.

3. H323 Calls

An IP compression type is specified in the IP trunk group parameters. But this algorithm is not ne-
cessarily used for H323 calls. This is because, to ensure compatibility with the remote, the choice of
algorithm is the result of negotiation between the caller and called party. See § 5 : Negotiation me-
chanism.
Object name: Trunk Groups > Trunk Group

IP Compression Type : - Default: the system default algorithm is chosen


in priority.
- G711: the G711 coding algorithm is chosen in
priority.

4. VPN hop on IP

An IP compression type is specified in the VPN overflow parameters. But, as is the case with H323
calls, the algorithm which is effectively used is the result of negotiation between the caller and called
party. See § 5 : Negotiation mechanism.
Object name: Inter-Node Links > VPN Overflow

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Attributes:

Node X - Node Y : Enter the numbers of the VPN hop end nodes, starting with the
smallest (X<Y).
IP Compression Type : - Default: the system default algorithm is chosen in priority.
- G711: the G711 coding algorithm is chosen in priority.

Remark: VPN algorithm type is broadcast. If broadcasting is not set up, it must be managed in the
same way on each side of the link.

5. Negotiation mechanism

The negotiation mechanism is involved in H323 calls and VPN overflow on IP. It applies to the compres-
sion algorithm, to VAD, and to the quantization law (A or µ law). This negotiation varies depending
on operating mode:

- Fast Start: the Call Server performs negotiation.

- Slow Start: the gateway handles negotiation.


Fast Start is a function provided by H323v2. It allows faster signaling exchanges with fewer messages
and ensures selection of an identical algorithm at both ends of the link. The caller provides, in the
SETUP, the list of compression algorithms he is able to use, as well as VAD mode (with or without VAD)
or the quantization law (A or µ law).
On receiving the SETUP, the called party selects from the list received, the algorithm, law, or VAD mode
that will be used. The result of this negotiation is transmitted to the caller, usually in the ALERT or
CONNECT message (possibly in the CALL PROCEEDING or FACILITY message).

5.1. System options


Fast Start is systematically used for VPN overflow on IP (ABC link on IP). For H323 calls, Fast Start is
used if the H323 equipment allows this.

5.1.1. VPN overflow on IP


The H323 internode protocol Boolean is used to choose the protocol (proprietary or standard H323
V2) implemented for VPN overflow on IP. It has no impact on IP calls to H323 terminals.
It can be managed in R5.0 and is no longer used in R5.1. It must always be set to "Yes" (standard
H323 V2 protocol selected).
Object name: IP > IP Parameters
Attributes:

System Option : Select Inter-Node Protocol H323.


Inter-Node Protocol H323 : This parameter must be set to Yes.

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5.1.2. H323 Calls


The Fast Start parameter concerns the signaling protocol used between the PC, terminals and H323
gateway. It has no impact on VPN calls on IP.
This system option must be configured at "Yes", allowing H323 calls to be transmitted in Fast Start
mode. If the H323 terminal operates in Slow Start mode, the call will be transmitted as Slow Start.
Object name: IP > IP Parameters
Attributes:

System Option : Select Fast Start.


Fast Start : Select Yes.

5.2. Slow Start


For Slow Start, the gateway firmware handles compression algorithm negotiation. Call Handling per-
forms a simple check on the algorithm sent by the firmware. If this check is negative, the call will be
interrupted with the message: "INCOMPATIBLE DESTINATION".

5.3. Fast Start


The following sections describe the principle used to select the compression algorithm when Fast Start
is used.

5.3.1. Building the list


The list sent by the Call Server is built according to 4 parameters:

- Calling set algorithm, if it is a poor set 1,

- IP compression type parameter of the trunk group or VPN overflow.

- Compression type system option.

- Multi. Algorithms for Compression.


The list is built according to one of the following three cases:

1. The call is from a poor set: the following are listed in order:
• Poor set algorithm: G723 or G729.
• If different, trunk group algorithm (for calls to non OmniPCX Enterprise equipment) or
VPN hop algorithm.
• The complementary algorithm to the first algorithm, if multi-algorithm is set to "Yes".

2. The IP compression type of the trunk group or the VPN overflow is G711: the following are
listed in order:

1 The term poor set refers to a set belonging or linked to a board belonging to a poor domain.

The term poor domain is used to refer to a domain with a compressed type Extra domain coding
algorithm and a Max. No. of connections/domain of less than 10 000.

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• Trunk group algorithm (for calls to non OmniPCX Enterprise equipment) or VPN hop
algorithm: G711.
• The system algorithm: G723 or G729.
• The complementary algorithm to the second algorithm, if multi-algorithm is set to "Yes".

3. The IP compression type of the trunk group or the VPN overflow is Default: the following are
listed in order:
• The system algorithm: G723 or G729.
• The complementary algorithm to the first algorithm, if multi-algorithm is set to "Yes".
In this case, G711 is not listed.

Table 5 : Building the list of algorithms for case 1

Trunk group or
Multi. Algorithms
Poor set algorithm VPN overflow IP List sent
for Compression
compression type
G723 G711 No G729 G723
G723 G711 Yes G723 G711 G729
G723 Default No G723
G723 Default Yes G723 G729
G729 G711 No G729 G711
G729 G711 Yes G729 G711 G723
G729 Default No G729
G729 Default Yes G729 G723

In this case, insertion of the set algorithm at the head of the list is to avoid the call being made in
partial direct RTP mode.

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Example:

Domain 1
Intra-domain coding: G711
Extra-domain coding: G723

UA set
Direct RTP flow G723
WAN
IP network

Node 1 VPN hop


algo: G711

Domain 0
Intra-domain coding: G711 Node 2
Extra-domain coding: G711

In the above example, the UA set on node 1 calls an IP set on node 2.


The UA set is attached to domain 1, a poor domain. Thus the corresponding extra-domain algorithm
is presented at the head of the list. Thus, although VPN hop algorithm is G711, the call will be
made in G723 mode.

Table 6 : Building the list of algorithms for cases 2 and 3

Trunk group or
System Multi. Algorithms
VPN overflow IP List sent
Compression type for Compression
compression type
G711 G723 No G711 G723
G711 G723 Yes G711 G723 G729
G711 G729 No G729 G723
G711 G729 Yes G711 G729 G723
Default G723 No G723
Default G723 Yes G723 G729
Default G729 No G729
Default G729 Yes G729 G723

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5.3.2. Outgoing call


The Call Server sends a list of available algorithms to the IP board. This list is built as described in §
5.3.1 : Building the list.

5.3.3. Incoming call


The data to be considered for algorithm selection is:

- The list of algorithms that can be used (and sent) by the caller: referred to as the received list,

- The list of algorithms available on the OmniPCX Enterprise, built as described in § 5.3.1 :
Building the list: referred to as the Enterprise list.
The selection principle is as follows:

- The first algorithm in the received list that is also contained in the Enterprise list is selected.

- If there is no algorithm common to both lists and if G711 is included in the received list, G711 is
selected.

- If there is no algorithm common to both lists and if G711 is not included in the received list, the
call fails.

Table 7 : Examples

Received list Enterprise list Algorithm selected


G729 G723 G711 G723 G729 G729
G729 G723 G711 G723 G723
G723 G729 G729 G723 G723
G711 G723 G723 G723
G711 G723 G711 G723 G711
G711 G723 G729 G711
G711 G723 G729 G711
G723 G729 Failure
G729 G711 G723 Failure

5.3.4. Negotiating VAD and the quantization law


The VAD mode and law selected are those of the calling node, unless the called party does not support
VAD, in which case the call is made without VAD.
Example: The called node is configured with VAD, but the calling terminal does not support VAD. In
this case, the call is made without VAD.
Note that a difference between the calling and called node settings is not enough to prevent setup of
direct flow and cannot result in call release.

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5.4. Call using VPN hops


For a call using two VPN hops, negotiation is performed for each hop. If the result is different, the call
is still made in direct RTP mode and the compressed algorithm is selected. This is still possible due to
the homogeneity of the system algorithm on the entire network.

Node 1 Node 2 Node 3


System algo: G723 System algo: G723 System algo: G723

VPN hop algo: G723 VPN hop algo : G711

IP network G723 RTP flow

5.5. Call using a VPN hop and an IP trunk group


For a call using a VPN hop and an IP trunk group, negotiation is performed for each VPN hop and IP
trunk group.
If the two end devices do not have a common algorithm, two compressors must be allocated to ensure
conversion between the two algorithms. The call is set up in partial network direct RTP.
This case may occur with 4645 voice mail (that only supports G711) or with an H323 terminal: see
module Direct RTP in ABC network - Functional description - § Partial direct RTP.

6. IP-Phones

The compression algorithm configured for an IP-Phone is applied for local calls and public network
calls set up from the set if it is a poor set. For other network calls, the set algorithm is not applied, the
algorithm is selected by the negotiation mechanism described above (see § 5.3.1 : Building the list).

1. At IP-Phone level, specify whether:


• The IP-Phone follows the general directive for the associated IP domain.
• For this specific IP-Phone, the administrator bars use of compression.
• For this specific IP-Phone, the administrator makes use of compression mandatory. In this case,
the type of compression specified at PCX level is used.

2. At IP domain level, specify for all IP-Phones in the domain whether compression is to be used or
not, or whether the settings made at IP set parameters level are to be applied.

3. At IP set parameters level, specify whether compression is to be used or not for all IP-Phones.

4. On the PCX, compression type is specified for the entire PCX: G723 or G729.
In the case of an intra-domain call, the algorithm applied for the IP-Phone is determined as indicated
in the figure below.

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Figure 5 : Compression for an intra-domain call

Users
IP Domain
Coding algorithm:
INTRA-domain
- Default coding algorithm:
IP phone
parameters
Default coding
- Default
algorithm:

- Without
- Without compression: System
- Without G711
compression: parameters
compression:
G711 G711 - With compression Compression
type:
- With
- G723
compression
- With
compression - G729

In the case of an extra-domain call, IP-Phone settings are not involved. Only domain settings apply,
as indicated in the figure below.

Figure 6 : Compression for an extra-domain call

IP Domain

EXTRA-domain
coding algorithm:
IP phone
parameters
Default coding
- Default
algorithm:
- Without compression: System
- Without G711 parameters
compression:
G711 - With compression Compression
type:
- With
- G723
compression

- G729

Algorithm selection: once the algorithm of the two domains (or of the two IP-Phones) has been
determined (according to the principle presented in the above two figures), the algorithm selected will
be that which uses the least bandwidth. If one of the two algorithms is G723 (or G729), the call will
be made in G723 (or G729). If both algorithms are G711, the call will be made in G711. See § 6.5
: Example.

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6.1. IP Phone

Object name: Users > TSC IP User

Attributes:

Directory Number : Displays IP-Phone subscriber's directory number.


Voice Coding Algorithm : This parameter is used for intra-domain calls. Possible
values are:
- Default: the IP Domain object parameter is used.
- With Compression: the compression type specified
in system parameters is used.
- Without Compression: no compression, coding
complies with the G711 standard.
IP Domain Number : Displays terminal domain number.

6.2. IP domain

Object name: IP > IP Domain

Attributes:

IP Domain Number : Enter the number of the IP domain.


Intra-domain Coding : This parameter is used for IP-Phones with their coding algorithm
Algorithm set to Default Possible values are:
- Default: the Default Voice Coding Algorithm parameter
of the IP Phones Parameters object is used.
- With Compression: the compression type specified in system
parameters is used.
- Without Compression: no compression, coding complies
with the G711 standard.
Extra-domain Coding : This parameter is used for extra-domain calls, in which case,
Algorithm IP-Phone settings are not used. Possible values are:
- Default: the Default Voice Coding Algorithm parameter
of the IP Phones Parameters object is used.
- With Compression: the compression type defined by the
system parameter is used.
- Without Compression: no compression, coding complies
with the G711 standard.

6.3. IP Phones

Object name: IP > IP Phones Parameters

Attributes:

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Parameter : Select: Default Voice Coding Algorithm.


Default Voice Coding : - Without Compression: by default, IP-Phones do not use
Algorithm compression (transmission complies with the G711 standard).
- With Compression: the Compression Type system
parameter is used.

6.4. On the PCX

See § 2.1 : Default compression algorithm.

6.5. Example

A remote Media Gateway is accessible via a link whose bandwidth is restricted. To meet this constraint,
call compression on this link is mandatory: the extra-domain compression algorithm is set to With
Compression.

For intra-domain calls (that use the LAN), voice quality is given preference: the intra-domain compres-
sion algorithm is set to Without Compression.

The system default algorithm is G723.

In the figure below:

- The call between B and E is compressed: the extra-domain algorithm has precedence over set
settings.

- The call between D and E is not compressed.

- The call between C and D is compressed because one of the two sets is configured with
compression.

Domain 0 Domain 1
Intra-domain coding: G711 Intra-domain coding: G711
Extra-domain coding: G723 Extra-domain coding: G723

G723 C (With
CS compression)

IP network

G723 G711

A (Default) B (Default) E (Without


D (Default)
compression)

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Management - Compression

7. Inter-Media Gateway Call

If the two Media Gateways belong to the same domain, the Intra-domain Coding Algorithm pa-
rameter is used.

If the two Media Gateways belong to two different domains, the Extra-domain Coding Algorithm
parameter is used. If the two domains are managed with a different algorithm, the call is made with
compression.

8. Bandwidth requirements according to the compression algorithm

The table below lists bandwidth requirements according to algorithm, framing and network type.

Framing is the transmission period of voice packets on the network. It is always 30 ms for G723. It
can be 20 or 30 ms for G729 and G711. Selection is performed using the following parameter.

Object name: IP > IP Parameters

Attributes:

System Option : Select Framing VOIP.


Framing VOIP : Select 20ms or 30ms. This is the transmission
interval for voice packets on the IP network.
This parameter only concerns compressed calls in
G711 or G729.

Table 8 : Bandwidth requirements

G723.1
Codec G 729A G 711
(MP-MLQ)
Bit rate 6.4 kb/s 8 kb/s 64 kb/s
Packet transmission period 30 ms 20 ms 30 ms 20 ms 30 ms
(framing)
RTP payload size (in bytes) 24 20 30 160 240
IP frame size (in bytes)2 64 60 70 200 280
Bandwidth at IP level 17.1 kb/s 24 kb/s 18.7kb/s 80 kb/s 74.7kb/s
Bandwidth at Ethernet level 27.2 kb/s 39.2 kb/s 28.8kb/s 95.2 kb/s 84.7kb/s
(Full Duplex Media)3

2 =RTP payload +RTP(12)+ UDP(8)+IP(20)


3 =IP Frame + MAC (14) + CRC (4) + preamble (8) + inter-frame silence (12)

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G723.1
Codec G 729A G 711
(MP-MLQ)
Bandwidth at WAN level 19.2 kb/s 27.2 kb/s 20.8 kb/s 83.2 kb/s 76.8 kb/s
without CRTP4
Bandwidth at WAN level with 9.1 kb/s 12 kb/s 10.7 kb/s 68 kb/s 66.7 kb/s
2-octet CRTP.45

4 8 bytes Layer 2 overhead (= maximum for PPP, MLPPP, FRF.12, HDLC)


5 CRTP: Compression for RTP header, the IP/UDP/RTP headers are restricted to 2 bytes.

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Management - Call Restrictions

44A01F01002A160AAEN

IP Overview
Management
Call Restrictions
Edition: 03

Several parameters are used to restrict voice calls on IP in order to adapt to the network's bandwidth.

1. Call restrictions within the same node

Object name: IP Domain, attribute : Domain Max Voice Connection


This parameter is used to restrict the number of calls between devices of the same node.
Example: a remote Media Gateway is accessible via a 128 kbit/s allocated link. In order to maintain
a proper quality of the calls, their number is limited to 2 on the link. To do this:

- in domain 1 parameters, extra-domain calls are restricted to 2.

- in domain 0 parameters, the parameter is left at -1 (no check)

Figure 7 : Restricting the number of extra-domain calls

Domain 0 Domain 1
max calls : 2
max calls : -1
Media Gateway

CS CS

IP Network

Allocated links
128 kbit/s
2 calls max
Call
IP-Phones IP-Phones
prohibited

2. Restriction by management of IP boards

Object name: Shelf > Board, attributes: No. of Compressors for Gateway and Nb Compres-
sors for IP Devices
These two parameters are used to assign certain compressors of a board to the Gateway H323 function
and others to calls to IP-Phones and Media Gateway.

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Example: to take into account the bandwidth of the link between nodes 1 and 2, the number of calls
is limited to 5. On the boards which support the logical link, the maximum number of compressors
for the Gateway function is managed at 5.

Figure 8 : Assigning compressors to IP boards

Node 1 Node 2
Nb calls for Gateway : 5
Nb calls for Gateway : 5 Nb calls for IP-Phones :2
Nb calls for IP-Phones :19

CS
CS

Calls to IP-Phones and Calls to IP-Phones and


Media Gateway : 19 max Media Gateway : 2 max

ABC log. link on IP


5 calls max

IP-Phones IP-Phones

Restrictions:

- if the direct RTP in network is validated, the allocation of compressors is dynamic, the restriction
of calls on the logical link is therefore no longer possible with these two parameters. In this
case, the sum of the two parameters defines the sum total of compressors that can be used
simultaneously on this board.

- If the same board supports the links to several nodes, the restriction can only be applied globally
at the start of a node and not according to the direction.
Warning: The time intervals of the IP trunk group T0/T1/T2 access must never be modified.

3. Restricting calls by VPN overflow

Object name: Inter-Nodes Links > VPN Overflow, attribute : Maximum number of IP calls
See also module VPN overflow - Functional description - § Limiting VPN calls.
Managing the maximum number of IP calls on a VPN hop is used to restrict the number of calls between
two given nodes. Indeed, inter-node calls are performed by VPN overflow.
Example: with 3 nodes interconnected by ABC links on IP.
Between the node 1 network and the node 2 network, the IP link is restricted to 5 calls.
Between the node 1 network and the node 3 network, the IP link is restricted to 10 calls.
There is no direct IP link between the IP networks of nodes 2 and 3.
To take into account these constaints, the number of calls on the logical links is restricted, by managing
a maximum number of calls by VPN hop, as indicated in the figure below.

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Management - Call Restrictions

Figure 9 : Restricting calls by VPN overflow

Node 1 Node 2
Max Nb VPN hop 1-2 : 3 Max Nb VPN hop 1-2 : 3
Max Nb VPN hop 1-3 : 8 Max Nb VPN hop 2-3 : 2
IP link 5 log. link 1-2 :
calls max 3 calls max
CS
CS

IP link 10
calls max log. link 2-3 : Node 3
2 calls max CS Max Nb VPN hop 1-3 : 8
log. link 1-3 : Max Nb VPN hop 2-3 : 2
8 calls max

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Section - IP overview IP Overview
Appendix - H323 standard

44A01F01000A980AAEN

IP Overview
Appendix
H323 standard
Edition: 03

1. H323 Standard Presentation

1.1. Generalities
H323 is part of the H32x series of standards dealing with video conferencing over different networks. It
is an extension of standard H320 (video conferencing on ISDN (Integrated Services Digital Network)),
as is standard H324 (video conferencing on PSTN (Public Switched Telephone Network), H323 being
the standard describing video conferencing on the LAN.
Standard H323 provides for voice/data convergence around IP protocol.
H323 is the internationally adopted standard (approved in 1996) for real-time multimedia communi-
cations (audio, video and data) over a network. H323 is adapted for voice transmission on IP
It provides a common protocol enabling communications software to operate together.
This standard was developed by the International Telecommunications Union (ITU) for networks not
guaranteeing a quality of service (QoS ): IP and IPX on Ethernet, Fast Ethernet and Token Ring.
This standard is intended for LAN and, in the future, for INTERNET.
It concerns call control, multimedia management, management of the bandwidth and the interfacing
between the LAN and the other networks.

1.2. H323 architecture (protocols and CODECs)

Control Data Audio Video Control

G7XX H26X
H225 H245 T120 RTCP

RTP

TCP UDP

IP

H323 can be implemented regardless of the physical medium and the layer 2 protocol used:

- WAN: ATM

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- LAN:
• FDDI (optical fibres),
• Ethernet,
• etc.
The H323 architecture comprises:

- Protocols: H225, H245, T120, RTP, RTCP,

- CODEC: G7XX, H26X.


H323 uses TCP for the following information:

- signalling,

- control,

- data.
The H323 protocols associated with TCP are:

- H225 handles signalling between 2 stations during connection establishment. This is a signalling
protocol based on Q931.

- H245 handles call control. This protocol handles the negotiation of channel opening and
communications parameter establishment. It determines whether the communication is audio or
video and/or data and which type of compression is to be used.

- T120 handles multimedia communications (real-time data and conferencing).


H323 uses UDP for the following information:

- Audio,

- Video.
The H323 protocols associated with UDP are:

- RTP (Real Time Protocol). By using RTP, H323 guarantees good transmission conditions for audio
and video flows. RTP adds a header to each packet. This header contains:
• time stamp,
• order number.
RTP allows to:
• eliminate any packets received twice,
• put the packets back in order,
• synchronize the audio, video and data.
RTP is associated with RTCP control protocol.

- RTCP (Real Time Control Protocol).

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The purpose of RTCP is to control the quality of service.


It handles frame order and informs the system if the connection is still active. It also provides a
transfer average.
The H323 protocols using UDP are also associated with flow CODECs. The latter represent two
separate families:
• Voice CODECs (G7XX),
• Video CODECs (H26X).
To introduce voice and/or video on an IP network, you must reduce the bandwidth that they occupy.
This is done by audio and/or video compression algorithms, which are supplied by the CODECs.
The audio CODECs are:

- G711: does not allow for optimal compression. It is used for transporting voice at a throughput
between 56 and 64 Kbps.
H323 imposes at least G711 compression even though G711 was essentially designed for a
constant throughput network (ISDN type) rather than a variable throughput one (e.g.: LAN).

- G723: The compression mode present in the G723 CODEC is used to transport voice at
throughputs of 5.3 Kbps or 6.4 Kbps, namely a throughput 10 times less than the one proposed
by G711.
Voice quality is variable and is slightly below that of public telephony.

- G729: This CODEC is the one which best meets the VoIP applications.
It uses slightly more bandwidth than G723 (8/13 Kbps). Voice quality is close to telephone quality.
Video CODECs are:

- H261: H261 is used to transport video at throughputs equal to or higher than 64 Kbps.

- H263: H261 is used to transport video at throughputs equal to or less than 64 Kbps.

1.3. Components defined by the H323 standard


The H323 standard defines 4 components for VoIP:

- the terminal,

- the Gateway,

- the Gatekeeper,

- The multipoint control unit or multipoint controller (MCUs - MC: Multipoint Controller, MP:
Multipoint Processor).

1.3.1. The terminal


The H323 terminal can be:

- a PC,

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- a telephone set,

- a dedicated videoconferencing terminal.


It must, at the least, implement the G711 compression standard.
For voice, the protocols used by H323 terminal are:

- H245 protocol,

- H225 signalling protocol (Q931) for communication establishment and halt.


There are two types of H323 terminal - a high quality one (used on LAN) and another optimized for
low bandwidths (G723 and H263).

1.3.2. The Gateway


This allows a terminal connected on an IP network (H323) to communicate with another on an ISDN
(H320) or STN (H324).
It undertakes signalling correspondence from Q931 to H225, controls signal correspondence and the
coherence between the different media (for example: correspondence of throughputs, audio transco-
ding, audio compression, multiplexing, etc.).
Application software: Time RTCP/UDP packets IP packets Interface with the media: e.g.
Digitalization Compression stamping: RTP allocation distribution Ethernet

PCM Ethernet
(e.g.: T2) Gateway

1.3.3. The Gatekeeper


The gatekeeper handles the following functions:

- Authorization administration: this is used to authorize making a call or not,

- Restrict bandwidth (if need be) and handle the traffic on the LAN.

- Management of user names,

- Translation of the H323 addresses of the users into IP addresses. The H323 addresses of the users
can be telephone number type or e-mail address type.
Its presence in an H323 network is not mandatory. However, if there is one, all the H323 terminals must
use it. For this, the H323 standard proposes a signalling type: RAS (Registration Admission Status).

1.3.4. The multipoint control unit or the multipoint controller


This allows the establishing of a 3-way or more conference, handling the negotiation between the
terminals according to the H245 protocol (media control protocol).

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1.3.5. The different cases

Three types of connection can be envisaged:

1. Network to Network
H323 Terminal H323 Terminal
e.g.: netmeeting e.g.: netmeeting

Intranet

2. Network to Telephone

H323
Terminal
e.g.: netmeeting

H323 Telephone
Intranet gateway network

3. Telephone to Telephone

H323 H323
gateway
Intranet gateway

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2. Structure of the datagrams

2.1. IPv4
0 31
Version Length Service type Total length
Identification Flags fragment re-allocation
Lifespan Protocol Header control
Source IP address
Destination IP address
Further IP options Jam
Data
..........

Description of fields

- Version
The version field allows the sender, recipient, router to agree about the structure of the datagram.
The machines reject the datagrams if the version of their IP software is different from the IP data-
grams received. The value is 4.

- Length
The length field defines the header of the IP datagram. The header fields are fixed except for the
IP options and stuffing fields.

- Type of service

0 7

Priority D T R Unused

• Priority
"Priority" indicates the priority of a datagram (0= normal priority, ..., 7= network supervision).
As a rule, the machines and routers ignore this field. It is used to ensure that control information
takes priority over data. This is used for congestion control.
• Bit "D"
indicates the type of routing. If D=1, then there will be a short routing delay.
• Bit "T"
indicates a high throughput request.
• Bit "R"
indicates a high reliability request.

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- Total length
The total length field defines the total length of the IP datagram (header plus data).

- Identification
The identification field is used to identify the fragment. It allows the recipient to reassemble the
fragment.

- Flags
• the first low-order bit is called the “non fragmentation bit”. If bit=0, then there will be no
fragmentation. An application can inhibit fragmentation to ensure the integrity of the datagram.
• the second bit is called “data to come”. If bit=1, then there are data to come.

- The move fragment


The move fragment field indicates the moving of the data transported as regards the initial data-
gram. Each fragment has the same structure as the initial datagram.

- Life cycle
The life cycle field indicates, in seconds, the maximum duration of datagram transit. The routers
decrement this counter. When a datagram is destroyed, an error message is generated (ICMP). This
field is used to avoid having a datagram loop endlessly over the network.

- Protocol
The protocol field indicates the type of higher layer protocol used (TCP=6; UDP=17; ICMP=1, etc.).

- Header control
The header control field ensures the integrity of the header content. The checksum is only carried
out on the header. This is used to reduce the processing time of the IP datagrams at router level.

- The source IP address and IP target address fields


complete the IP addresses of the machines.

- IP Options
Allows to make tests, upgrading.

- Stuffing
Allows the header to be multiple of 32.

- Data
Indicates the content of upper layers.

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2.2. IPv6
0 31

Version Flow label - QoS


Total length Next header Lifespan
Source IP address (128 bits)
Destination IP address (128 bits)

Description of fields

- Version
The version field allows the sender, recipient, router to agree about the structure of the datagram.
The machines reject the datagrams if the version of their IP software is different from the IP data-
grams received. The value is 6.

- Flow label
The flow label field (with priorities) replaces the IPv4 datagram type of service field. It increases its
possibilities by making possible acceptance in the routers of the quality of service. This field has
been extended to meet the constraints of real-time services.

- Total length
The total length field defines the total length of the IP datagram (header plus data).

- Next header
Extension possibilities for the headers and options exist via the next header field (8 bits). This field is
used to organize the information elements, indicating the protocol entity to be invoked in order to
process the next envelope. This field appears at the start of each option header. In IPv6, the options
are stored in additional headers located between the IPv6 header and the transport packet header.
There is a small number of header extensions, each one identified by a separate next header value.
An IPv6 datagram can contain a variable number of additional headers. Most of the options in the
headers are neither examined, nor processed by the intermediary routers. Unlike IPv4, the options
can have a random length; there is no limit on the size. Each additional header has a length which
is a multiple of 8 bytes.

- Life cycle
The life cycle field indicates, in seconds, the maximum duration of datagram transit. The routers
decrement this counter. When a datagram is destroyed, an error message is generated (ICMP). This
field is used to avoid having a datagram loop endlessly over the network.

- The source IP address and IP target address


complete the IP addresses of the machines.

2.3. RTP and RTCP


RTP: Real-time Transport Protocol (rfc 1889)
RTCP: RTP Control Protocol (rfc 1889)

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Voice is transported in the RTP/RTCP sessions above UDP. The RTP protocol is used to encapsulate the
data packets. The RTCP protocol is used to control line quality.
RTP provides a point-to-point delivery service for data with real-time characteristics (voice, video).

15 0
PT M CC X P Version
Sequence number
Time stamping (LSB)
Time stamping (MSB)
SSCR (LSB)
SSCR (MSB)
CSCR (LSB)
CSCR (MSB)
........

Description of fields

- PT
Payload Type. The PT (payload type) field identifies the format of the data field and determines how
it is interpreted by the application.

- G711 law µ 0
- G711 law A 8
- G723.1 4
- Noise Comfort 13
- G729A 18

- Bit "M"
Bit “M” is used to define specific events.

- CC
CSRC Count. the CSRC number identifies what follows the header. This number is greater than 1
if the RTP packet data contain data from several sources (in the conferencing case, for example).

- Bit "X"
Used for a future extension.
If the value is 1, this means that the header is followed by an extension header.

- Bit "P"
Bit “P” specifies the presence of a stuffing field at the end of the packet (not used at present).

- Version

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Identifies the actual version of the RTP header. This is version 2.

- Sequence Number
The sequence number field increases by 1 with each RTP packet sent. It can be used by the recipient
to detect lost packets and to restore the packet sequence.

- Time stamp
The time stamp field specifies the time when the RTP packet was formatted. When frame exchange
is constant, this field is used to synchronize the RTCP and RTP packets.

- SSCR
Source Synchronization. The SSCR (Synchronization Source) field defines the identification of the
source synchronization. It is a value chosen at random to distinguish the synchronization sources
within a same RTP session.

- CSCR
Contributing Source. The CSCR (Contributing Source) field informs the sources intervening for the
data field contained in this packet. The contributor number is given by the CC field.

In the first step, the fields used are the PT, sequence number and SSCR fields.

RTP is an IP-based protocol allowing transfer of data such as video or audio flows in real-time. RTP
provides the scheduling and timing devices required by multimedia transmissions.

RTCP is used with RTP to control the latter. Two types of RTCP packets can be generated in order to
monitor the QoS of a multimedia data transmission in real-time:

- RR (Receiver Report):
Return on the data reception quality.
• highest packet number received,
• number of packets lost,
• inter-arrival jitter,
• time stamps to calculate round-trip.

- SR (Sender Report):
Reception quality (RR) + transmission specific information:
• inter-media synchronization,
• cumulative packet counters,
• of bytes sent.

This protocol is therefore used to assess the QoS level of a transmission in both directions (transmis-
sion/reception). It is used to monitor the QoS for the VIP PC product.

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3. Summary of H3XX standard characteristics

ISO- B-ISDN B-ISDN


Packet
Network N-ISDN PSTN Ethernet
switched
IEEE 802.9 (ATM) (ATM)
Multimedia H.320 H.324 H.322 H.323 H.321 H.310
standard
Audio/voice G.711(M) G.723.1 G.711(M) G.711(M) G.711(M) MPEG1 (M)
G.722 G.729 G.722 G.722 G.722 G.711 (M)
G.728 G.728 G.728 G.728 G.722
G.723.1 G.728
G.729
Audio rates, 64 5.3–6.3 64 64 64 nx94
Mbps 48–64 8 48–64 48–64 48–64 64
16 16 16 16 48–64
5.3–6.3 16
8
Video H.261 (M) H.261 (M) H.261 (M) H.261 (M) H.261 (M) H.262 (M)
H.263 (M) H.263 (MPEG-2)
H.261 (M)
Data * T.120 T.120 T.120 T.120 T.120 T.120
Multiplex H.221 (M) H.223 (M) H.221 (M) H.225.0 (M) H.221 (M) H.222.0 (M)
H.222.1 (M)
Control H.242 (M) H.245 (M) H.242 (M) H.245 (M) H.242 (M) H.245 (M)
Signalling Q.931 Q.931 H.225.0 Q.931 Q.931
(Q.931)

(M) = Mandatory.

* For example, the Whiteboarding application.

4. Voice quality

4.1. Measurement parameters of voice quality

The voice quality on IP is measured by the following 5 transmission parameters:

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- Service Availability
Average availability of a service over a given period. This must be the maximum and as close as
possible to the "five nines" rule (99.999%) which is the telephony norm.

- Delay
Time between the transmission of a packet and its reception over the network. In the case of voice
and video interactive applications, the appearance of delays may give the impression that the system
is not responding.

- Jitter
Jitter presents the variation of the end-to-end transmission delay. High levels of jitter are not accep-
table for applications running in real-time.
This results in signal distortion, requiring the introduction of additional delays necessary for packet
re-assembly, thus making it very difficult to propose interactive sessions.

- Throughput
The throughput is the maximum transfer rate that can be attained between two points in a network.
It is not only limited by the physical infrastructure of the path followed (which nonetheless indicates a
theoretical limit), but also by the quantity of miscellaneous traffic sharing the common components
of the selected path.

- Reliability - Packet loss rate


This parameter is dependent on the design and configuration of the network used. In a poorly
configured system or one that works incorrectly, the packets may arrive at their destination in a
different order or even disappear altogether, thus entailing their retransmission. In the case of voice
and video applications, a failing in reliability may result in distortion in the signal received.

4.2. Voice quality techniques

There are five techniques for QoS implementation:

- Over Provisioning Bandwidth


Although in itself not strictly a QoS solution, increasing bandwidth allows the network to support
greater traffic loads. This method works out to be rather expensive.

- Bandwidth Conservation
This involves restricting total traffic so that the network operates more efficiently, by using techniques
such as IP multicast, data compression, allocation of bandwidth on demand, etc.

- Traffic Prioritization
Each packet sent over the network has an associated priority level used by the network equipment
responsible for transmission. This technique is founded on the fact that certain types of traffic are
deemed more important than others.

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- Static Resource Reservation


A pre-set percentage of the bandwidth of a network segment is reserved for defined types of traffic.
This solution is frequently used for transmission-delay sensitive protocols.

- Dynamic Resource Reservation


This approach requires the installation of a signalling protocol such as RSVP allowing an application
to reserve a given quantity of bandwidth and to specify the performance characteristics required for
a given flow.

4.3. Services Prioritization Standards


The quality of service mechanism consists in marking VoIP frames in order to give them priority over
such VoIP frames as data frames (client / server computer applications, file transfer...). Three standards
can be used.

4.3.1. 802.1p
The 802.1p standard is used to assign levels of priority to the ISO model layer 2 frames (Link). The
standard defines the following parameters:

- 802.1QPpriority: defines the priority level allocated to the datagram, 0 being the lowest priority
level, 7 being the highest priority level. Thus in the level 2 switching equipment, the datagrams
with the highest priority level are processed before the datagrams with the lowest priority level.

- 802.1QvLAN: defines virtual LAN domains. Thus the level 2 switching equipment disable
inter-vLAN communications. Thus closed user groups are created. Then the inter-vLAN
communications require routers configured as inter-vLAN gateways.

- 802.1Qsubnet: defines the sub-network type. This parameter is not used in the PCX.
Warning: Not all switching equipment process the 802.1Q standard. Some of them even ignore it
and consider this field when present as an error. The corresponding datagrams are then erased.

4.3.2. ToS (precedence)


This standard was defined in the 80s and had not been used until the recent appearance of congestion
at router level, which meant that priority levels had to be defined for IP frames.
The “precedence” level is coded on the first 3 bits of the TOS byte (value between 0 and 7, recommen-
ded value for voice: 5).

4.3.3. Diffserv (Differentiated Services)


Diffserv is a standard defined by the IETF (Internet Engineering Task Force). Diffserv works with traffic
conditioners located at the end of the network to indicate the need for each packet.
This is a layer 3 (Network) solution, which means it can work on the layer 2 equipment currently used,
while remaining adapted to the new layer 3 architectures.
The Diffserv level of priority is coded on the first 6 bits of the TOS byte (value between 0 and 63,
recommended value: 46).

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4.4. Quality of Service Contracts


The Quality of service contracts (Service Level Agreements - SLA) are established between network
service suppliers and their clients.
Each of the quality of service parameters is assessed and agreements reached to set a minimum ope-
rating framework. The supplier is then under the obligation to deploy the resources necessary to
guarantee that this minimum framework is met. If the requirements are not met, compensations may
be due.

4.5. Applications Performance


Part of the applications under development operate in client/server mode and their operation does not
necessarily require all the QoS characteristics insofar as they are not really multimedia applications.
The analysis of the conversations between a client and server over a LAN is used to determine different
durations characteristic of a client/server application. These durations must be less than the generally
accepted maximum durations:

- Total Round Trip (TRT)


Duration between transmission of a request by a client and reception of the response returned by
the server. TRTmax = 1.5s.
TRTmax = 1.5 s.

- Round-Trip (RT)
Duration between the transmission and reception of a network message. RTmax =0.5s
RTmax =0.5 s.

- Processing Time (Ps)


Duration of the processing of a request by the server - Duration between the reception of the request
and the transmission of the response. Psmax = 1s
Psmax = 1 s

4.6. Quality of service on OmniPCX Enterprise


The PCX transmits the Quality of service information given by the manager. The latter defines the
quality of service parameters in accordance with the client site network manager. If the client switching
equipment do not allow the presence of the 802.1Q field, it can be deleted from the datagrams.

5. TCP/IP tools (present on a PC running Windows NT)

Command Remark
ipconfig indicates the IP address of the machine, its subnetwork mask and the
address of the default gateway.
ping tests the connectivity to a remote machine.

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Command Remark
telnet used to connect to any port on a remote server. This means it is
possible to check the response of a service to the connection on
the corresponding port.
tracert indicates the route used by IP datagrams to reach a remote machine.
netstat displays all the connections established on a server.
route access to the IP routing tables of a machine.
arp access to the Ethernet IP addresses translation tables (MAC
addresses).

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44A01F01000A015AAEN

IP Overview
Glossary
Edition: 03

1. A

10Base T A variant of Ethernet, connecting stations via twisted pair cabling (shielded or
unshielded) wiring at 10Mbps.
10Base F A variant of Ethernet which runs on optical Fiber (mono-mode or umti-mode) at
10 Mbps.
100Base TX A variant of Ethernet which runs on Category 5 unshielded twisted pair wiring at
100Mbps. This is one version of Fast Ethernet.
100Base FX A variant of Ethernet which runs on optical Fiber (mono-mode or multi-mode) at
100 Mbps.
1000Base T A variant of Ethernet, connecting stations via twisted pair cabling at 1 Gigabit.
1000Base LX A variant of Ethernet, connecting stations via Long-wawe optical fiber at 1 Gigabit
1000Base SX A variant of Ethernet, connecting stations via Short-wawe optical fiber at 1 Gigabit
802.x The set of IEEE standards defining LAN protocols
AAL ATM Adaptation Layer Corresponds to a service-dependant sublayer of the
dala-link layer. The AAL accepts data from different applications and provides it to the ATM layer in
48 bytes payload segments. AALs differs depending on the service required : e.g Constant Bit Rate or
Variable Bit Rate. The ITU recommends four types of AALs : AAL1, AAL2, AAL3/4, AAL5.
AAL1 ATM adaptation layer 1 used for connection oriented, delay-sensitive service
requiring constant bit rate like voice.
ACELP Algebraic-Code-Excited Liner Pediction Analog to digital coding providing good
voice quality at only 8 kb/s. Has been standardized by the ITU in G729 as CS-ACELP.
Access Control Method This is the main distinguishing feature between different LAN technologies. It
regulates each workstation’s physical access to the cable (transmission medium), and determines the
order in which nodes gain access so that each user gets efficient service. Access methods include
token passing, which is used in token ring and FDDI, and Carrier Sense Multiple Access with Collision
Detection (CSMA/CD), which is employed by Ethernet and Fast Ethernet.
ACL Access Control Lists Rules for packet filters (typically routers) that define which
packets to pass and which to block.
Access Router A router that connects your network to the external Internet. Typically, this is your
first line of defense against attackers from the outside Internet. By enabling access control lists to
this router, you’ll be able to provide a level of protection for all of the hosts « behind » that router,
effectively making that network a DMZ instead of an unprotected external LAN.

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ADPCM Adaptative Differential Pulse Code Modulation Analog to digital coding providing
high-quality digital signals at 32 kb/s or 16 kb/s.
ADSL Asymmetric Digital Subscriber Line By using the latest technology in DSP, bitrates
of over 8 Mb/s (downstream) and 800 Kb/s (upstream) are possible over the existing telephone
network. The telephone traffic and the Internet traffic are separated by a filter allowing simultaneous
use of the telephone and the Internet service.
AH Authentication Header Part of the IPSec protocol suite. It is the header used in
IPSec-compliant IP packets to carry authentication data permitting verification of the sending party.
ANT ADSL Network Termination (ADSL Modem)
Application-Level Firewall A firewall system in which service is provided by processes that maintain
complete TCP connection state and sequencing. Application level firewalls often re-address traffic so
that outgoing traffic appears to have originated from the firewall, rather than the internal host.
ARP Address Resolution Protocol Used to translate an IP address to an ethernet
physical address.
ARPA : Advanced Research Project Agency
ASP Application Service Provider
ATM Asynchronous Transfer Mode A high speed, connection-oriented switching
and multiplexing technology for transmitting information across a WAN or a LAN. ATM divides
information into fixed-length cells of 53 bytes capable of transmitting different types of traffic
simultaneously, including voice, video, and data. Fixed-length cells allows cell processing to be done
by the hardware, thereby reducing transit delays.

2. B

Backbone LAN or WAN connectivity between subnets across a high-speed network. Fiber
optic cable is often used.
BACP Bandwith Allocation Control Protocol Is the associated control protocol for BAP.
Bandwith reservationProcess of assigning or allocating bandwith to users or applications in a network
based on priority in order to make the best use of available bandwith.
BAP Bandwith Allocation Protocol Can be used to manage the number of links in a
multi-link bundle. BAP defines datagrams to coordinate adding and removing individual links in
a multi-link bundle, as well as specifying which peer is responsible for various decisions regarding
managing bandwith during a multi-link connection.
BAS Broadband Access Server
Bastion Host A system that has been hardened to resist attack, and which is installed on a
network in such a way that it is expected to potentially come under attack. Bastion hosts are often
components of firewalls, or may be « outside » web servers or public access systems.

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BGP Border Gateway Protocol BGP4 is a replacement for older EGP Based on RFC
1771
BOD Bandwith on Demand
BOOTP Like DHCP, BOOTP provides an IP address to a client and also a file name
in order to boot with TFTP.
BRAS Broadband RAS (idem as BAS)
BRI Basic Rate Interface ISDN interface composed of two B channels and one D
channel. A throughput of up to 128Kb/s is possible.
Broadcast A service in which information is sent from a central source to multiple
destinations.

3. C

Cable modem Via the cable network, bitrates up to 10Mb/s (downstream) and 28..768 Kb/s
(upstream) are possible.
CAR Commited Access Rate CAR, a function of CISCO « switch routers », allows
static banwith management by limiting the amount of bandwith consumed on a link by any given
application . Provides a minimum or a maximum bandwith for a specific type of traffic flow.
CAS Channel Associated Signaling A type of in-band trunk signaling.
CBQ Class-Based Queuing Is a public-domain scheme which divides all user traffic
into categories and assigns bandwith to each class. The classes themselves can be established by
configuring CBQ by combinations of IP address, protocols such as TCP or UDP and ports that
represent the applications such as file transfer, Web access and so on.
CCP Compression Control Protocol
CCS Common Channel Signaling A type of out-of-band trunk signaling (for example,
using Primary Rate Interface) in which a control channel carries signaling for separate voice and data
channels. In CCS signaling is passed in messages.
CDDI Copper Distributed Data Interface
CELP Code –Excited Linear Prediction An analog to digital voice coding and
compression scheme used in transmission of voice over data networks.
CGI Common Gateway Interface
CHAP Challenge-Handshake Authentication Protocol Is a more secure procedure for
connecting to a system than the PAP. It avoids the password to be sent over the link as in PAP.
CIDR Classless Inter-Domain Routing Based on RFC 1519
CIR Commited Information Rate Average rate of information transfer a subscriber
(for example the network administrator) has stipulated for a Frame Relay PVC.

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Circuit Switching A method of communication whereby a circuit is held open and maintained
only while the sender and recipient are communicating. This is different from a dedicated circuit
which is held open regardless of wether data is being sent or not, and different from a datagram /
connectionless network, in which data flows without establishing a connection.
CLNS Connection Less Network Services This type of service allows information to be
transferred over a network without having to set up and end-to-end connection before information is
sent.
CODEC Coder-decoder Technique of transforming analog voice into a digital bitstream
and vice versa; also used to indicate the compression type (for example, G.729 CODEC).
CONS Connection Oriented Network Services A connection-oriented service in a
network is one in which a connection has to be set up between the source and destination before the
communication can proceed.
COPS Common Open Policy Services Being standardized by the IETF, aims to manage
multiple network equipments from a central policy server.
CoS Class of Service In the 802.1p specification, COS uses 3 bits in the LAN frame
header to assign seven priority levels to LAN frames. Cos levels can be mapped to IP type of service
(ToS) levels or supported in routers with a number of other mechanisms.
CPE Customer Premises Equipment Is the end-user’s home equipment. It can concern
residential users or corporate users.
CSMA/CD Carrier Sense Multiple Access with Collision Detection A contention-based
network access method in which any computer may attempt to communicate at any time. Since there
is no centralized force controlling the medium, a device must first sense wether or not the medium is
in use. If the medium is unused, the device then transmits. If two computers sense that a channel is
open and transmit at the same time, the result is a collision, after which there is a random pause
determined individually by each transmitting machine. Each machine then senses the line again and,
if it is available, retransmits.
CRC Cyclic Redundancy Check Error checking technique
CRM Customer Relationship Management
CRTP Compression for RTPheader CRTP is a hop by hop compression similar to TCP
header compression. CRTP reduces the IP/UDP/RTP header to 2-4 bytes.

4. D

Datagram A form of packet switching in which the packets that make up a conversation
do not all take the same path through the network, thus improving the robustness and security
of the network.
DES: Data Encryption StandardA symetric encryption algorithm. DES uses a 56-bit key. DES has the
advantage that is is easily implemented in hardware but its keyspace may not be large enough for
continued use.

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DHCP Dynamic Host Configuration Protocol DHCP offers dynamic configuration of IP


address and related information (subnet mask, default router,…). It is an extension of Bootp. DHCP
provides safe, reliable, and simple TCP/IP network configuration, prevents address conflicts, and help
conserve the use of IP addresses through centralized management of address.
Diffserv Differentiated Services Basically, the idea is to assign different priorities to
different flows based on their Quality of Service needs. To achieve this, the differentiated service
approach employs a small, well-defined set of building blocks from which a variety of services may
be built. In particular, a small bit-pattern in each IP packet, in the Ipv4 TOS (the Type Of Service byte
has been redefined as the DS byte) byte is used to mark a packet to receive a particular forwarding
treatment, or per-hop behaviour, at each network node.
DLC Data Link Control
DMZ Demilitarized Zone A demilitarized zone is a computer host or small network
inserted as a neutral zone between a company’s private network and the outside public network. It
prevents outside users from getting direct access to a server that has company datas. Typically, a
DMZ is an IP network segment that contains resources available to Internet users such as Web
servers and FTP servers.
DNS Domain Name Service Domain Name System Is the way that Internet Domain
names are located and translated into IP addresses. A domain name is a meaningful and easy-to
–remember « handle » for an Internet address. The Internet user’s PC contacts the DNS server
which is located at the ISP premises.
DSL Digital Subscriber Line A protocol that can carry digital signals at a higher rate
across twisted-pair cabling.
DSLAM DSL Access Multiplexer
DTE Data Terminal Equipment
DTMF Dual-Tone Multifrequency
Dual Homed Gateway A dual homed gateway is a system that has two or more network interfaces,
each of which is connected to a different network. In firewall configurations, a dual home gateway
usually acts to block or filter some or all of the traffic trying to pass between the networks.
DWDM Dense Wawelength Division Multiplexinhg

5. E

E1 Wide-area digital transmission scheme used predominantly in Europe that


carries data at a rate of 2.048 Mbps using 30 64-Kbps digital channels for voice or data, plus a
64-Kbps channel for signaling and a 64-Kbps channel for framing. E1 lines can be leased for
private use from common carriers.
E.164 The ITU-T recommendation for assignment of international telecommunication
numbering, which is an evolution of traditional telephone numbers.
EGP Exterior Gateway Protocol

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EIR Excess Information Rate


Encapsulation The technique used by layered protocols in which a layer adds header
information to the protocol data unit (PDU) from the layer above.
Encryption The process of converting information from an easily understandable format
(plain text) into apparent random gibberish (ciphertext) by the use of well-defined rules and
calculations known as algorithms or cipher to ensure the privacy and confidentiality of information.
The reverse process is decryption.
Ethernet The most common layer-two protocol used in LAN’s. Ethernet is a 10Mbps
CSMA/CD standard originally developped by Xerox to run on thick coaxial cabling. It has evolved
and now runs primarily on twisted pair cabling.
ESP Encapsulating Security Payload Payload format used in IPSec compliant IP
packets to carry encrypted and/or authenticated data, thereby preventing sniffing on the network
between communicating nodes.
ETSI European TelecommunicationS Standard Institute
Extranet An Extranet is an intranet which has been extended to include a company’s
suppliers, partners and customers. They will lay the foundations for a major expansion of electronic
commerce.

6. F

FAQ Frequently Asked Questions


FDDI Fiber Distributed Data Interface
FEC Forward Error Correction
Firewall A firewall is a set of related programs, located between the intranet and the
Internet, that protects the resources of a private network from users and/or from other networks.
This firewall grants people from the company access to the Internet, but prevents that people from
the Internet get access to the companies resources. In particular, a H323 firewall is very complex,
because it allows H323 connections to be made (signaling and RTP/RTCP connections) between a
component located on the private network and another one connected to the Internet or other
network, in order to provide for example VoIP public connection.
FoIP Fax over IP This is carrying faxes on IP Networks (in general T38)
FR Frame Relay An ITU standard for the interface to a public frame-switching
network designed to provide high-speed frame transmission with minimum delay across the
wide area. It operates at data-link layer level and handles multiple virtual circuits using HDLC
encapsulation between connected devices. Is used in public and private networks, gradually replacing
X25 and leased-line networks.
FRAD Frame Relay Access Device
Frame A variable-length layer-two protocol entity containing address and other control
information, plus data.

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FTP File Transfer Protocol Is a standard protocol within the TCP/IP protocol suite,
which is the simplest way to exchange files between computers.

7. G

Gatekeeper The gatekeeper is an optional element. However, if a gatekeeper is present, it is


mandatory that H323 endpoints (terminals, gateways, MCU) make use of the services offered by
gatekeepers. Basically, the gatekeeper is a kind of andpoint manager. The gatekeeper services
basically include address translation, admission control, bandwith control and zone management. It
can be also in charge of theH323 call signaling acting as a call signaling « proxy » for the terminals.
Gateway In an internetworking context, the gateway can provide many services, including
translation between signaling procedures or between codecs ; in such a case, it will perform call
set-up/release and call control on both the IP side and the other.
G.711 The recommendation G.711 from the ITU is based on PCM technique at a
sampling rate of 8kHz. The frequency bandwith that is used is 300Hz- 3.4 kHz. This is generally used
in speech coding to restrict captured bandwith to a factor where voice signals are mainly present.
Each sample is coded with 8 bits(in Europe) or with 7 bits(in the US), which produces respectively a
64kbps or 56kbps bit stream.
G.723.1 The G723.1 recommendation targets very low bit rates. G723.1 is a dual rate
coder (5.3kbps or 6.4kbps) based on ACELP for the low-rate coder and based on MP-MLQ for the
high rate coder. The bandwith is 3.1kHz in both cases. The lower bit rates has smaller quality than
the higher one but provides systems designers with additional flexibility.
G729a The G729a recommendation targets very low bit rate. This is one of the most
recent and promising codecs standardized by the ITU. This belongs to the G729 family. As such, this
is a competitor to G723.1. This is based on CS-ACELP and produces a 8 kbps bit stream from a 3.1
kHz bandwith. The bit rate is slightly higher than G723.1, but the delay is significantly lower.
GIF Graphics Interchange Format
GRE Generic Routing Encapsulation
GSM Global System for Mobile

8. H

H225 Performs the signalling for call control. Defines a much larger set of capabilities
than those used in systems concerned only with voice traffic. H225 itself uses messages defined in
H245 to establish and terminate individual logical channels for audio. H225.0 corresponds to the
RAS signaling function (see H323 RAS).
H235 Securisation and authentication of recording sequences for H323 Gatekeeper.
H323 An ITU standard for multimedia communication (voice, video and data) over
connectionless networks that do not provide a guaranteed quality of service such as IP based network.
It addresses call control, media management, and bandwith management for point-to-point and

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multipoint conferences. It refers to a set of other standards (H.245, H.225.0, and Q.931) to describe
its actual protocol.
H323 RAS Registration, admission, and status. The RAS signaling function performs
registration, admissions, bandwidth changes, status, and disengage procedures between the VoIP
gateway and the gatekeeper.
H450 This corresponds to the supplementary services associated to H323 version
2 (similar to QSIG).
HDLC High-Level Data Link Control A bit-oriented synchronous data-link layer protocol
developed by ISO. HDLC specifies a data encapsulation method on synchronous serial links using
frame characters and checksums.
HTML Hypertext Markup Language. A form of page description language used in
the World Wide Web.
HTTP Hypertext Transfer Protocol Is the set of rules for exchanging files (text, graphic
images, sound video, and other multimedia files) on the Web. Relative to the TCP/IP suite of
protocols, HTTP is an application protocol.
HTTPS Is a Web protocol developped by Netscape and built into many browsers
that encrypts and decrypts user page requests as well as the pages that are returned by the Web
server. HTTPS is really just the use of the Secure Socket Layer as a sublayer under the regular HTTP
application layer.
Hub The center of a star topology network or cabling system. Typically used in older
Ethernet and token ring networks. A device connected to a hub receives all the transmissions of all
other devices connected to that hub. Hubs are now being replaced in many cases by LAN switches.

9. I

IAD Integrated Access device


IAP Internet Access Provider The IAP is responsible for the access between the user
and the ISP. Towards the user, the IAP can use for example the PSTN or ADSL. Towards the ISP, the
typical network can be the PSTN or Frame Relay. When a user connects to an IAP, it is up to the
IAP to find out to which ISP the user belongs.
ICMP Internet Control Message Protocol Is the error and control message protocol used
by the Internet protocol family. In particular, ICMP manages the ECHO/REPLY message (ping).
IDRP Interdomain Routing Protocol
IEEE Institute of Electrical and Electronics Engineers
IEEE 802.1p An IEEE standard for prioritizing time-critical flows and filtering multicast traffic
to contain traffic in layer-two networks. The 802.1p header includes three bits for prioritization,
allowing for eight priorities to be established.
IEEE 802.1Q An IEEE standard for providing a virtual LAN capability within a campus network,
used in conjunction with IEEE LAN protocols such as Ethernet and token ring.

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IEEE 802.2 A data link standard outlining how basic data connectivity over cable should be
set up. Used with the IEEE 802.3, 802.4 and 802.5 standards.
IEEE 802.3 The IEEE’s specification for Ethernet, including both physical cabling and
layer-two protocol.
IEEE 802.3ad Specifies link aggregation
IEEE 802.4 Specifies the Tken Bus protocol
IEEE 802.5 Specifies the Token Ring protocol
IEEE 802.11b Direct sequence standard for WLAN in the 2,4 GHz frequency range. Maximum
throughput is 11Mbit/s
IETF Internet Engineering Task Force Task force consisting of over 80 working groups
responsible for developing Internet standards.
IGMP Internet Group Management Protocol Is used in case of multicast streams
IGP Internal Gateway Protocol
IGRP Interior Gateway Routing Protocol
IIS Internet Information Server
IKE Internet Key Exchange Part of the IPSec protocol suite. IKE is the current IPSec
standard for SA rules negotiation, key management and key exchange.
IMAP4 Internet Message Access Protocol 4 With IMAP, you view your e-mail at the
server as though it was on your client computer. An e-mail message deleted locally is still on the
server. E-mail can be kept on and searched at the server. Nethertheless, applications using IMAP
sometimes include a synchronize function to download new mails onto a PC and to upload new
mails towards a mail-server.
Internet The Internet is a global information system constructed by interconnecting
thousands of networks which are logically linked by a global system of unique addresses based on
the Internet protocol (IP). It supports communications using the TCP/IP suite in order to provide
public or private high level services.
Intranet The term intranet refers typically to a corporate network which uses the same
technology that is behind the Internet. Intranets can run over private WAN networks or public
networks such as the Internet.
IP Internet Protocol The layer-three protocol used in TCP/IP set of protocols which
support the Internet and many private networks. IP provides a connectionless datagram delivery
service for transport-layer protocols such as TCP and UDP. IP provides also features for addressing,
type-of-service specification, fragmentation and reassembly, and security. IP is defined in RFC 791.
IPv4 Currently used IP version.
IP v6 the proposed next generation standard for IP addresses, incorporating IPSec
security features and other additions. Ipv6 addresses are 128 bits wide.
IP addressing Each computer (known as a host) has at least one address that uniquely identifies
it from all other computers on the Internet.

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IPv4 addresses are coded on 4 bytes.


IP v4 addressing supports fives different network classes : - Class A : range
0.0.0.0 to 127.255.255.255 : for large networks - Class B :range 128.0.0.0 to 191.255.255.255 :
for intermediate size networks - Class C : range 192.0.0.0 to 223.255.255.255 : for small networks
- Class D : range 224.0.0.0 to 239.255.255.255 : reserved for multicast groups - Class E : range
240.0.0.0 to 247.255.255.255 : reserved for future use
Private adresses which are not routed by the Internet : - range 10. 0.0.0 to
10.255.255.254 - range 172.16.0.0 to 172.31.255.254 - range 192.168.0.0 to 192.168.255.254
Unicast address the packet is addressed to only one host
Multicast address special type of address (hostid of all 1’s). In this case all hosts connected to
the network will accept the packet.
IP Precedence IP Precedence allows three of the ToS bits in the IP header to be set with the
values 0 through 7. This ranking determines the priority of the packet flow as it leaves one network
for another, with 7 being the highest priority.
IPCP IP Control Protocol Used within PPP, to negotiate for IP, the IP compression, IP
address, etc …
IPSec Internet Protocol Security A set of extensions to IP adding security services. The
suite consists of protocols for an authentication header (AH), encapsulating security payload (ESP)
and a key management and exchange protocol (IKE)
IP Spoofing An attack whereby a system attempts to illicitly impersonate another system by
using its IP network address.
IPX Internetwork Packet Exchange NetWare network layer (Layer 3) protocol used for
transferring data from servers to workstations. IPX is similar to IP.
IS Information Systems
ISDN Integrated Services Digital Network A communication protocol offered by
telephone companies that permits telephone networks to carry data, voice, and other traffic.
ISAKMP Internet Security Association and Key Management Protocol A framework
negotiation protocol on top of which IKE is designed.
ISO International Standards Organization
ISP Internet Service Provider An ISP is a company which provides connectivity to the
Internet and network services, most commonly for users who are accessing the Internet via the
telephone network. Typical services are : access to the Web, e-mail, webspace for homepages,
newsgroups
IT Information Technology
ITSP Internet Telephony Service Provider
ITU International Telecommunication Union An international body of member
countries whose task is to define recommendations and standards relating to the international

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telecommunications industry. The fundamental standards for ATM have been defined and published
by the ITU (previously CCITT).

10. J

Jitter A short term timing deviation It’is one of the three major concerns when carrying
Voice over IP. It corresponds to the variation in the delay between packets.

11. K

12. L

L2F Layer 2 Forwarding Is a protocol that allows corporations to extend their own
corporate network through private « tunnels » over the public Internet. L2F is a proposed standard
sponsored by CISCO systems.

L2TP Layer Two Tunneling Protocol Is a standard that combines the best features of two
existing tunneling protocols : CISCO’s L2F and PPTP.

LAC L2TP Access Concentrator

LAN Local Area Network A group of computers and other devices dispersed over a
relatively limited area and connected by a communication link that enables any device to interact
with any other on the network. The LAN corresponds to the network inside the enterprise.

LANE LAN Emulation

LAP Link Access Protocol

LAP-B Link Access Procedure Balanced

LCP Link Control Protocol Used by PPP for the negociation of the communication
parameters : authentication method, maximum receive unit,…

LDAP Lightweight Directory Access Protocol Protocol for access to directory services
managed by a directory server. Informations about hosts, users (e.g. authentication, access
information), traffic handling policy (QoS) can be managed by a directory server.

LL Leased Line

LLC Logical Link Control

LLC/SNAP : Logical Link Control / Sub-Network Access Protocol

LNS L2TP Network Server

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13. M

MAC Medium Access Control


MAC address The layer-two address of a LAN device
MAN Metropolitan Area Network
MAPI Messaging Application Programming Interface
MC Multipoint controller
MCU Multipoint Control Unit This aims at supporting conferences between three or
more H323 endpoints. It may handle the media streams between end-points in a multi-cast approach.
MGCP Media Gateway Control Protocol
MIB Management Information Base
MLPPP Multi Link Point to Point Protocol Allows to use multiple independent channels
(links) to create a virtual single bundle. Based on RFC 1990.
MIME Multipurpose Internet Mail Extensions
MOS Mean Opinion Score
MP-MLQ MultiPulse-Maximum Likelihood Quantization
MPLS Multi Protocol Label Switching Protocol being defined by the IETF to allow IP
packets to be switched in an efficient manner using different types of link layer protocols (e.g. ATM). It
is an optimization of the classical IP routing. MPLS attaches « labels » to IP packets which enables
routers and switches to forward traffic based on information in the labels, rather than inspecting the
different fields deep within each and every packet.
MPOA Multi Protocol Over ATM
MPPP Multi-link PPP It provides bandwith aggregation from multiple links, including
analog and ISDN, to get a higher communication throughput.
MTU Maximum Transfer Unit The maximum packet size, in bytes, that a particular
interface can transmit. Example 1.5Kbytes bytes on Ethernet, 4 Kbytes on FDDI.

14. N

NAS Network Access Server NAS are devices composing a POP. They can be connected
to different kind of networks and interfaces such as PRI, BRI, ATM, Ethernet … A point-to-point
connection is established between the Internet user and the NAS which will route the packets to
the correct interface.
NAT Network Address Translation Is the translation of an IP address used within one
network to a different IP address known within another network. Typically, a company maps its
local inside network addresses to one or more global outside IP addresses and unmaps the global
IP addresses on incoming packets back into local IP addresses. It allows to share a single address
between multiple equipments, and to connect them all to the Internet at the same time.

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NCP Network Control Protocol Used within PPP to negotiate the network protocol
options.
NetBEUI Network Bios Extended User Interface In charge of transport functions (level 4
ofOSI model). Used in particular in case of IBM PC networks.
NetBIOS Network Basic Input/Output System
NFS Network File System
NMC Network Management Centre
NNTP Network News Transfer Protocol Is the predominant protocol used by computers
(servers and clients) for managing the notes posted on Usenet newsgroups.
NTP Network Time Protocol NTP is a UDP-based protocol used for synchronizing a
set of network clocks using a set of distributed clients and servers. Implementation is based on RFC
1305. Simple NTP is documented in RFC 2030.

15. O

OSI Open Systems Interconnection


OSPF Open Shortest Path First Routing protocol based upon the Link State Algorithm
Each router actively test the status of its link to each of its neighbors, send this information to its
neighbors , which then propagate it.

16. P

Packet A packet is the basic unit of transmission under IP. Data streams are broken
into packets by the transmitting machine, passed through the network and then reassembled
at the receiving end.
Packet filtering The ability of a bridge, router or gateway to limit propagation of packets between
two or more interconnected networks.
Packet switching A communications method in which variable-length packets are individually
routed between hosts.
PAP Password Authentication Protocol Is a procedure used by PPP servers to validate a
connection request. Passwords are sent in clear text without security and the originator can make
repeated attempts to gain access. For these reasons a server that supports CHAP will offer to use that
protocol before using PAP.
PAT Port Address Translation
Payload Refers to the portion of a packet following the header.
PCM Pulse Code Modulation
PDU Protocol Data Unit

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PLC Packet Loss Compensation

PoP Point of Presence The node at which an ISP connects a subscriber to the Internet.
To give individual access at the lowest rate possible, dial-in facilities (a POP) are installed by the
Internet Service Provider (ISP) in every telephone area. A POP consists of one or more NAS.
POP3 Post Office Protocol version 3 Is the most recent version of a standard protocol
for receiving e-mail. POP3 is a client-server protocol in which e-mail is received and held for you
by your Internet server. This protocol includes commands to login, logout, fetch messages and
deletes messages. The point of the POP is to transmit the E-mails from the E-mail server towards the
user s PC to be read later.

Port Number Fields of the TCP and UDP header which identifies the source and destination
application program . Is coded on 2 bytes.
POTS Plain Old Telephone Set (System) The basic telephone service supplying standard
single-line telephones, telephone lines, and access to the public switched telephone network.

PPP Point to Point Protocol An Internet protocol which is used to connect serial terminal
devices, usually over dial-up lines. PPP (Point-to-Point Protocol) is a protocol for communication
between two computers using a serial interface. PPP is a full-duplex datalink protocol that can be
used on various physical media, including twisted pair or fiber optic lines or satellite transmission.
It uses a variation of High Speed Data Link Control (HDLC) for packet encapsulation. The PPP
protocol handles : error detection, support of multiple protocols (IP, IPX, …), dynamic IP address,
authentication of the user.

PPPoA PPP over ATM

PPPoE PPP over Ethernet Primarily deployed in DSL environments. Allows authentication,
control of the connection in case of a connection to an ISP through Ethernet. PPPoE leverages existing
Ethernet infrastructures to allow users to initiate multiple PPP sessions within the same LAN.

PPTP Point to Point Tunneling Protocol Is a layer 2 tunneling protocol based on PPP that
allows corporations to extend their own corporate network through private « tunnels » over the public
Internet. PPTP is a proposed standard sponsored by Microsoft and other companies.

PQ Priority Queuing

PRI Primary Rate Interface ISDN interface composed of 30 B channels and one D
channel. A throughput of up to 2Mb/s is possible.

Proxy A software agent that acts on behalf of a user. Typical proxies accept a connection
from a user, make a decision as to whether or not the user or client IP address is permitted to use
the proxy, perhaps does additional authentication, and then completes a connection on behalf
of the user to a remote destination.

PSQM Perceptual Speech Quality Measurement

PSTN Public Switched Telephone Network

PVC Permanent Virtual Circuit

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17. Q

QoS Quality of Service Quality of service means the ability of networks to guarantee
and maintain certain performance levels for each application, according to the specified needs of
each user. It will consist in a special type of treatment applied to a flow of traffic or for a certain user
e.g. Regarding VoIP, Quality of service will help to reduce transit delay, jitter and ensure bandwith
needs.

18. R

RADIUS Remote Authentication Dial-in User Service Is a client/server protocol based


on UDP and software that enables remote access servers to communicate with a central server to
authenticate dial-in users and authorize their access to the requested system or service. RADIUS
provides a central location for storing informations like : authentication attributes, configuration data
for establishing a WAN connection for an incoming call, dialout information, static routes and
filters, accounting information, security information. Developped to better manage large serial line
and modem pools. The client/server model supports security via PAP, CHAP, UNIX login, and other
authentication schemes, such as challenge/response systems.
RARP Reverse Address Resolution Protocol Used to translate an ethernet physical
address into an IP address.
RAS Registration Admission and Status This is defined in H225.0. It performs
registration, admission, bandwith changes, status and disengage procedures between H323
endpoints of a zone and the gatekeeper responsible for that zone.
RAS Remote Access Service/Server
RED Random Early Discard Method which relies on rules based on probability to
instruct a router to begin dropping packets when established queuing thresholds are crossed.
RFC Request For Comment
RIB Routing Information Base
RIP Routing Information Protocol Based upon the Distance Vector protocol. Each
router sends all or some portion of its routing table but only to its neighbors. Each router updates its
routing table based on the vector of these distances (hop counts) that it receives from its neighbors.
Router A layer-three device responsible for making decisions regarding which of several
paths network traffic will follow. To do this, it uses a routing protocol to gain information about
the network, and algorithms to choose the best route based on several criteria (known as routing
metrics). Routers interconnect subnets.
RSA The original and best-known asymmetric encryption scheme where one key (the
public key) and one algorithm is used to encrypt data and another key (the private key) and another
algorithm are used for decryption.
RSVP Resource reservation Protocol RSVP is a QoS signalling protocol for the Internet.
It reserves a portion of the output link in each router along the path of a flow for a particular

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application. It delivers QoS requests to all nodes along the path (s) of the flows and establish
/maintain state of the requested service. RSVP also includes provisions for constraining packet
delay and guaranteeing bandwith availability , but on a managed corporate IP network, only the
prioritization feature needs to be used. RFCs : 2210, 2209, 2208, 2207, 2206, 2205
RTP/RTCP Real-time Transport Protocol / Real-time Transport Control Protocol This provides
the media stream packetization and synchronization for all data networks.

19. S

SA Security Association In the IPSec protocol suite, a dedicated secure virtual


connection between two nodes.
SDH Synchronous Digital Hierarchy
Serial delay Serial delay results when a delay sensitive packet (voice in our case) is stuck in a
buffer behind a packet that has already begun to be sent. All links in a WAN operate in a serial
manner.
Thus the delay sensitive traffic must wait until the packet has passed. The delay
variation introduced by this serial delay depends on the maximum length of the packet (long packets
results in less link overhead but maximizes serial delay)
Thus the delay sensitive traffic must wait until the packet has passed. The delay
variation introduced by this serial delay depends on the maximum length of the packet (long packets
results in less link overhead but maximizes serial delay)
There are 2 ways to deal with serial delay : –the data size can be limited
(Maximum Transmission Unit size) – the speed of the link can be raised
SIP Session Initiation Protocol IETF standard for VoIP systems.
SLA Service Level Agreement
SMTP SimpleMail Transfer Protocol SMTP is used by an E-mail server to transmit an
E-mail to the destination E-mail server. In most cases, the user will use a PC to retrieve his/her
mail from the server using usually POP3 or IMAP.
SNA Systems Network Architecture
SNMP Simple Network Management Protocol Is a protocol used for network
management. The Network Management Center contains an SNMP-manager. The objects to be
managed contain an SNMP-agent and a Management Information Base (MIB). The manager can
read from and write into this database. The agent can also send ’traps’ towards the manager to
report alarms. Summarized, this protocol can be used to handle Fault, Performance, Security and
Configuration Management.
SSL Secure Socket Layer Is a program layer created by Netscape for managing the
security of message transmissions in a network. It is used by secure protocols such as HTTPS.
STA Spanning Tree Algorithm It’s task is to construct a non-looping topology by
deciding not to use certain of the links in the network.

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Subnet A portion of a network in which all stations share a common subnet address.
Subnet Mask Used to subdivide a network into subnets. Defines the number of bits borrowed
for the subnet address. The mask is 32 bits long. Example : 255.255.255.0
Internet routers use only the network id of the destination address to route traffic
to a subnetted environment. Routers within the subnetted environment use the extended-network-id
(network id + subnet id) to route traffic between the individual subnets.
SVC Switched Virtual Circuit

20. T

T120 Data conference protocol


T30 Procedures for document facsimile transmission in the general switched
telephone network
T37 Procedures for batch or Store and Forward Group 3 facsimile communication
over IP networks. Built on TCP and SMTP.
T38 Procedures for real-time Group 3 facsimile communication over IP networks.
Built on UDP in Fax relay mode.
TCP Transmission Control Protocol This is a reliable connection oriented protocol that
allows the error free delivery of a byte stream. Large packets are segmented into smaller ones and
are resequenced at the final destination if necessary. Flow control makes sure that the receiving
side is not overloaded.
TCP /IP The various protocols which support the Internet and many private networks.
Telnet Is the way you can access someone else’s computer, assuming they have given
you permission. With telnet, you log on as a regular user with whatever privileges you may have
been granted to the specific applications and data on that computer.
TFTP Trivial File Transfer Protocol Is a protocol used to transfer files and has been
implemented on top of UDP. In general, it is used to download binary and data files.
ToS Type of Service An 8-bit field within the IP header which can be used by the device
originating the packet, or by an intermediate networking device, to signal a request for a specific
QoS level. The ToS field is redefined in DiffServ.
TTL Time To Live. The TTL indicates the maximum amount of time a diagram is
allowed to remain in the network (usually, the maximum number of hops through routers).
Tunneling Tunneling is a technique whereby information is encapsulated in a protocol that
allows the information to pass through a larger stream of information without fear of interference.
Effectively, « tunneling » creates a secure means to transport information among other information.
It is becoming a popular method of creating virtual private network interconnection over common
media such as the Internet.

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Trafic shaping This applies at access devices to prevent a huge burst of traffic from congesting
the backbone network. Shaping involves accepting a burst from an input device, buffering the traffic
and then « smoothing » out the flow so that the burst is distributed over a long period of time, a
time period based on configuration parameters.

21. U

UDP User Datagram Protocol The UDP protocol is an unreliable connectionless


protocol which is widely used for client/server request/reply applications where the prompt delivery
is more important than the accurate delivery.
URL Universal Resource Locator To retrieve an HTML page from a remote host, a user
will enter a link in his browser. This link is called a URL.

22. V

V.34/V.34bis V.34 is a standard, approved by the ITU, for transmitting data to modems.
V.34bis provides up to 33.6kbps of fallback to 31.2 kbps or V.34 transfer rates (28.8Kbps or fallback
to 24 Kbps and 19.2 Kbps and backwards compatibility with V.32 and V.32bis).
V.90 Is a standard approved by the ITU for transmitting data downstream to modems
at 56Kbps. 56Kbps transmission technologies exploit the fact that most telephone company offices
are interconnected with digital lines.
VAD Voice Activity Detection In Voice over IP (VOiP), voice activation detection (VAD)is
a software application that allows a data network carrying voice traffic over the Internet to detect
the absence of audio and conserve bandwidth by preventing the transmission of "silent packets"
over the network.
VLAN Virtual Lan In a VLAN, individual devices are assigned membership in a group
that has connectivity only to each other and whose traffic does not mix with other traffic as it crosses
backbone networks, distant switches and shared hubs. Many of these groups or VLAN’s may
coexist on the same network infrastructure. VLAN also allows to reduce the broadcast domain to
the ports belonging to the VLAN. 802.1Q provides virtual LAN capability whereas 802.1p provides
prioritizing time-critical flows. The 802.1p header includes three bits for prioritization, allowing eight
priorities to be established.
VoIP Voice over IP Is the voice carrying on IP network
VPIM Voice Protocol over Internet Messaging
VPN Virtual Private Network A Virtual Private Network enables to send data between
two hosts across a shared or public internetwork (ex the Internet) in the same way as in case
of a point-to-point private link. It corresponds to extend an Intranet (Home/remote workers or
LAN-to-LAN connection)
It gives the appearance and benefits of a private network, including continuous
availability and reliability.

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VPN offers the following properties : –1encapsulation of the private data –


authentication of the VPN connection to be established – data encryption to ensure the confidentiality
of the data over non-secure networks
A VPN uses a protocol like PPTP, L2TPor IPSec.
VRPP Virtual Router Redundancy Protocol

23. W

WAIS Wide Area Information System Is an application layer protocol which can be
used to look for information in a large number of documents.
WAN Wide Area Network This corresponds to a network outside the business, and one
that is accessible (such as the Internet, and an Intranet Extension)
WFQ Weighted Fair Queuing WFQ applies to the bandwith an application receives on
an output link :
– Traffic is assigned a priority – Priority influences which traffic is transmitted first
on a congested pipe
It has no impact on a non-congested pipe.
WINS Windows Internet Naming Service
WLAN Wireless LAN
WRED Weighted Random Early Discard Is a variation of RED. A pure RED router just
randomly selects packets to drop when some buffer threshold is reached independently from the
priority of the packet. WRED tries to identify the low-priority traffic and randomly discard those
packets when congestion occurs.
WWW World Wide Web A worldwide network of interconnected computer servers which
allow users to access information rapidly and easily via the Internet.

24. X

X25 ITU standard which was the first international standard for packet switching.
Covers only the bottom three layers of the OSI Model.
XML eXtensible Markup Language

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25. Y

26. Z

Zone A collection of H323 terminals, Gateways and MCUs managed by a single


Gatekeeper.

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