BusinessPhone Communication Platform 5.1
Facilities

## Networking via Voice over IP lines

Definition
References to other documents
Use
Operation
General
Functions
Capacity
ASB 150 02 R12
ASB 150 02 R13
Limitation
Installation
Hardware Installation
Programming
Start of operation
Test the Installation
Maintenance
Equipment/Environment
IPU board
Networking via IP lines
Pre-requisites to the customer‘s LAN environment
Glossary

## Definition

With the function “Networking via Voice over IP lines”, ASB 150 02 provides an integrated trunk gateway functionality between the circuit switched telephony world and the packet switched LAN world.

The trunk gateway functionality based on the IP Unit (IPU) can be used to build up a homogenous proprietary ASB 150 02 network by using the existing customer LAN/WAN environment instead of a complex configuration with leased lines.

The function “Networking via Voice over IP lines” is primarily based on the functionality of “Networking via digital lines”. From a networking functionality it does not matter whether the connection between the nodes is built up via leased lines or IP lines over the corporate network (Intranet).

Therefore this document does not focus on the whole networking functionality, it only describes the additional features on Voice over IP (VoIP) lines.

## References to other documents

Board description of the IPU
see System description, (1551-ASB 150 02 Uen)

Installation and Prerequisites
see Installation instruction for ASB 150 02,
(1531- ASB 150 02 Uen)

Licensing of voice channels
see Maintenance Instructions, (1541-ASB 150 02 Uen)

Capacity upgrading of the IPU
see Maintenance Instructions, (1541-ASB 150 02 Uen)

First test after Installation and Programming
see Start of Operation, (1537-ASB 150 02 Uen)

Firmware upgrading of the IPU
see Maintenance Instructions, (1541-ASB 150 02 Uen)

Technical Data - Voice over IP lines / IP Extensions
see System overview, (1555-ASB 150 02 Uen)

## Use

By using the existing data infrastructure the trunk IP-Gateway makes it possible to save costs especially for network configurations as the amount of leased lines is minimized.

As it is configured with an IPU board, the ASB 150 02 can act as end, transit, or gateway node in a mixed configuration with circuit switched lines

## Operation

### General

#### Functionality

##### Voice functionality

Apart from the networking functionality based on QSIG and proprietary UUS signalling, the following additional functionality will be provided for voice calls.

- 4, 8 or 16 simultaneous bi-directional voice channels according to the used license.
- Address translation between telephony (PNR number) and network number (IP number= IP address + TCP port).
- Definition of Voice Transmission Quality (VTQ) for VoIP calls on individual extension levels by using three different Voice-codecs: G.711, G723.1 and G.729 Annex A (with Annex B)
- To provide Quality of Service for Voice over IP calls “Type of Service “ (prioritization of voice packets) is supported in the IP header.
- DTMF tone handling
- Manual handover to circuit switched lines in case of insufficient voice quality.
- Echo cancellation
- Comfort noise generation
- Silence suppression

##### IP functionality

From the IP world point of view, the following functionality is supported.

- IP address administration
  The IPU supports a static IP address, BootP or DHCP.
- Application selection through ports
  Up to 15 different ports are prepared to select future applications. In ASB 150 02 R12 only one application (Networking with ASB 150 02 nodes) is supported.
- Voice transmission is based on UDP/IP
- Signalling is on TCP/IP

#### VoIP Networking packages

The function networking via Voice over IP lines is licensed from two perspectives - from Networking and from the IPU point of view.

##### Networking packages

The networking functions are licensed by the FECU on the CPU-D4. The following packages are available:

- Standard digital Networking Package
- Full digital Networking Package

with the variation of 2, 4, 32 and unlimited trunk individuals (BTU-B, BTU-D and IPU). For detailed information see document “Networking via digital lines (365/15534-ASB 150 02 Uen).

##### IPU packages

These packages focus on the amount of available bi-directional voice channels through the IP network .The number of available voice channels on the IPU board will be defined by a license file which is located on the Compact Flash card. The Compact Flash card itself will be plugged on the IPU.

For detailed description about the IPU licensing, refer to document Maintenance instructions,
1541-ASB 150 02 Uen.

For a detailed description of the license ordering process see Order information ’BusinessPhone Communication platform ’,113 62-11/ASB150 02

### Functions

#### Voice channel handling

The voice channels on the IPU board are handled as individual trunks (implementation is based on ISDN BA links) and are based on the same parameters as used for QSIG trunks.

##### Outgoing calls

A VoIP call can only be set-up through the PNR as a private network call. After a TCP/IP point-to-point connection has been established to the destination node, the “IP-networking” specific handling is started, by using a tunnelled QSIG protocol for call control. Further on, this call is handled in the same manner as a QSIG network call.

##### Incoming calls

An incoming VoIP call is handled as a CN call.

##### Gateway/transit calls

A call set-up between an IP-interface and other CN (Q‑/T-) interface will be handled as a CN transit call. A call set-up between an IP-interface and non-CN (Q-/T-) interface will be handled as a gateway call.

#### Address translation between PNR number and IP number

A VoIP call can only be set-up through the PNR by dialling the private access number.
Apart from the CN parameters the private access number refers to the IP reference table containing the IP number (IP address + TCP port) of the destination node. (See configuration example on the next page)

A direct call set-up by dialling the route or the directory number of the IP trunk individual will automatically be denied.

##### Configuration example

##### PNR Table

Private network Access Code | Destination PBX | TON priv calling | 1 Route choice | 2 Route choice
|  | digit arrangement |  |  |  | digit arrangement |
route choice | TON priv/pub called | remove | predigits | IP ref. | route choice | TON priv/pub called | remove | predigits | IP ref.
PBX A:
4 | proprietary | RL0 | 10 | UNK, priv |  |  | 1 | 00 | UNK,pub | 1 | 811884 |
5 | proprietary | RL1 | 10 | UNK, priv |  |  | 2 | 00 | UNK,pub | 1 | 0512420505 |
6 | proprietary | RL1 | 10 | UNK, priv |  |  | 2 | 00 | UNK,pub | 1 | 0512420506 |
PBX B:
2 | proprietary | RL0 | 8 | UNK, priv |  |  | 1 | 00 | UNK,pub | 1 | 811002 |
3 | proprietary | RL0 | 8 | UNK, priv |  |  | 1 | 00 | UNK,pub | 1 | 811003 |
5 | proprietary | RL1 | 8 | UNK, priv |  |  | 2 | 00 | UNK,pub | 1 | 0512420505 |
6 | proprietary | RL1 | 8 | UNK, priv |  |  | 2 | 00 | UNK,pub | 1 | 0512420505 |
PBX C:
2 | proprietary | RL1 | 12 | UNK, priv |  |  | 1 | 00 | UNK,pub | 1 | 01811002 |
3 | proprietary | RL1 | 12 | UNK, priv |  |  | 1 | 00 | UNK,pub | 1 | 01811003 |
4 | proprietary | RL1 | 12 | UNK, priv |  |  | 2 | 00 | UNK,pub | 1 | 01811884 |

##### IP Reference table

Node number | Signalling protocol | IP number (IP-Address:Port number)
PBX A
1 | 101 | 192.133.144.155:7000
2 | 101 | 194.188.199.122:7000
PBX B
1 | 101 | 192.155.166.177:7000
2 | 101 | 194.188.199.122:7000
PBX C
1 | 101 | 192.155.166.177:7000
2 | 101 | 192.133.144.155:7000

#### Voice Transmission Quality

##### Definition

In order to keep the required bandwidth for an VoIP call as low as possible, voice codes on the IPU are used to compress the used telephony bandwidth of 64kbit/s to a lower value. However, each compression causes a reduction in the voice transmission quality.

ASB 150 02 supports the following voice codecs.

- G.711 provides uncompressed voice transmission, and nearly the same voice quality as a circuit- switched line, however it occupies more bandwidth, which in case of heavy LAN traffic could result in some speech delays.
- G.729 Annex A (with Annex B) provides compressed voice transmission (8 kbit/s). The advantage of this codec is that the compresion rate down to 8 kbit/s, while the speech quality is on a very high level ( comparable G.711)
- G.723.1 provides compressed voice transmission (6,4 kbit/s) of course with some reductions in the voice quality, however it occupies less bandwidth, .

##### Operation

ASB 150 02 is prepared for 4 different transmission quality levels, which can be defined on extension level for outgoing Trunk IP-Gateway calls. Each level comprises a set of 4 voice codecs, whereby the access of the codecs depends on a different order of priority.
If an extension sets up a VoIP call, the codec with the highest priority related to the dedicated codec set will be requested in the own exchange. If the codec is available, it will be selected and a message is sent to the destination exchange informing about the selected and the alternative codecs. The destination exchange verifies the proposed codecs with the internal resources and confirms the first corresponding and available codec. Depending on the selected codec from the destination exchange, the originating exchange can continue call set up or reallocate a new codec (in the case another codec than the one with the first priority has been choosen from the destination exchange). If the reallocate fails, the call will be released and the PNR automatically switches over to the alternative routes.

The list below shows the different quality levels and the dedicated predefined order of priority of the voice codecs. Per command ( 2165-2168 ) it is possible to redefine the prioritisation of each set.

- Quality level 1
  High bandwidth with best transmission quality

- 1. G.711 (A-law)
- 2. G.711 (m-law)
- 3. G.723.1
- 4. G.729 Annex A with Annex B
- 5. G.729 Annex A

- Quality level 2
  Medium bandwidth with good transmission quality

- 1. G.729 Annex A with Annex B
- 2. G.729 Annex A
- 3. G.723.1
- 4. G.711 (A-law)
- 5. G.711 (m-law)

- Quality level 3 (default)
  Medium bandwidth with medium transmission quality
  1. G.723.1

- 2. G.729 Annex A with Annex B
- 3. G.729 Annex A
- 4. G.711 (A-law)
- 5. G.711 (m-law)

- Quality level 4
  Low bandwidth with low transmission quality

- 1. G.723.1
- 2. G.729 Annex A with Annex B
- 3. G.729 Annex A
- 4. G.711 (A-law)
- 5. G.711 (m-law)

In case of transit/gateway calls always Quality level 3 is used.

#### Quality of Service

To provide Quality of Service for Voice over IP calls, the IPU supports “Type of Service” (prioritisation of voice packets) in the IP header. It has to be verified whether this service is supported and programmed on the routers in the LAN environment. If the service is not supported by the routers, the voice packets will be sent transparently without any prioritization, which might result in some speech delay related to the current LAN traffic.

#### DTMF tone handling

The DTMF digit sending between two ASB 150 02 systems connected via VoIP lines is done with tone signals, detected by a DTMF receiver on the IPU.
The tones are translated to a DTMF message and then forwarded to the network. In the destination exchange the DTMF message is translated back to tones.
The DTMF digit handling from and to the IP network is only applicable for suffix dialling in ’speech-state’.

#### Echo cancellation

The DSM modules on the IPU, have echo cancellers included which suppress the echos generated by trunks or extensions.

#### Silence suppresion

In order to keep the required bandwith during a conversation as low as possible, the DSM modules on the IPU are always supervising the noise level on each voice channel. If the noise falls below a certain limit, no further data packets will be sent to the destination site.

#### Comfort noise generation

If the silence suppresion is active, the opposite conversation partner will hear silence, due to the fact that no data transmission is ongoing. In order to give conversation partner the impression that the connection is still active, a low noise will be generated by the DSM modules on converstation partner‘s IPU.

#### Manual handover to circuit-switched lines in case of insufficient voice quality.

##### Definition

This feature allows the user to manually alter the VoIP speech path to a connected B-party. This may be relevant in cases when the Voice transmission quality over the IP Network is insufficient. The ongoing VoIP call to the B-party will be replaced by a new call using the next route choice in the PNR-table. During the handover, the connected B-party is reserved for the next incoming external call.

##### Operation

If the Voice transmission quality is insufficient, the user can switch over to an alternative connection by using the following procedure.

Originator (A-Party)

- Initiate Inquiry
- Dial *61*
  As soon as the new call is established, the previous call will automatically be released.

In order to switch over to an alternative connection, the following prerequisites have to be fulfilled, otherwise the function is denied and the user receives the congestion tone.

- an alternative PNR route must be available
- the originator must be allowed to use the function
- the current call is an IP call Terminator (B-party)

Terminator (B-party)

After the handover has been initiated by the A-party, the called B-party (same as the one previously connected) is reserved for 20 seconds for the next incoming external call and is informed about the alternative routing by the display Message “SPEECH PATH REPLACEMENT”.
During reservation the B-party can cancel the reservation by pressing the CLEAR -key.

##### Limitation

A comparison of the A-party CLI which is received via the original- and the alternative route is not possible in all cases (e.g. no or incomplete CLI is received via the alternative route). Therefore it cannot be used for a dedicated reservation of the B-party. It might happen, that another incoming external call is presented during the reservation.

The reservation and the information about the ’Handover of the IP call’ is only possible when the connected party resides in a ASB150 02 and when USS 1,2,3 is available on the original path.

Operator telephones are not allowed to initiate a handover.

The reservation of group member extensions is not provided in case of ACD calls or PBX- group calls with parallel distribution.

## Capacity

### ASB 150 02 R12

- 1 IPU per System
- 8 bi-directional voice channels ( indpendent of the used voice codec)

### ASB 150 02 R13

- Multiple IPUs are supported, whereby the total capacity is limited to the maximum number of supported trunk lines as well as to the available free board positions.
- 16 bi-directional voice channels in conjunction with the DSM2 modul ROA 219 5114/2 ( indpendent of the used voice codec)
- 16 bi-directional voice channels in conjunction with the DSM1 modul ROA 219 5114/1 ( only valid for the voice codec G.711. If G.729 Annex A (with Annex B) or G.723.1 are used only 8 voice channels will be supported)

Note

Due to thermal considerations the IPU can not positioned in every free slot. By default, it has to be mounted in the highest board position of the cabinet. If any other position has to be chosen or in case of multiple IPUs then the next right hand board position of the IPU must always be left free.
This restriction does not apply, if the cabinet BDVBS 101 06 (BP128i) is used

## Limitation

The VoIP lines are used to support “speech”, other “audio” calls e.g. modem or fax calls will be allowed but the transmission cannot be secured.

VoIP lines are only supported between ASB 150 02 nodes.

An IP link can neither be a synchronsation source
(master) for the system nor be synchronized against the network.

## Installation

### Hardware Installation

Please refer to the document Installation instruction for ASB 150 02, (1531- ASB 150 02 Uen).

### Programming

The following commands are only accessible via BMS:

#### Initiate Voice IP-Trunks

Enable the desired number of IP-extensions on every IPU which has a configured extension gateway.

Select the desired IPU and click on the IPU Licensing button in the ’Card information’ dialog, to open the IPU Licensing dialog.

The dialog shows the maximum number of licensed voice channels. These voice channels are used by the Trunk Gateway as well as by the IP-Extensions.

Note

Any modification in this dialog, requires a restart of the IPU

#### Trunk Individuals

Define the desired number trunk individuals, which might be less or equal to the number of licensed voice channels.

Values: 0,4,8( default ),12 and 16

#### IP-Extension Individuals

Not relevant for the Trunk Gateway

#### IP-Extension threshold

Not relevant for the Trunk Gateway

#### Voice over IP related settings

Note

Any modification of the commands 2151-2169 requires a IPU restart.

2151 |  | Preferred start-up protocol

2152 |  | IP address (own)

2153 |  | Gateway IP address

2154 |  | Subnet mask

2157 |  | First RTP-port number

2158 |  | Amount of RTP-sessions

2159 |  | MAC address (read only)

2160 |  | Network connection (read only)

2161 |  | Application

2162 |  | Port number

2163 |  | Signalling protocol

2164 |  | Network IP address

2165 |  | Voice Transmission Quality Table 1

2166 |  | Voice Transmission Quality Table 2

2167 |  | Voice Transmission Quality Table 3

2168 |  | Voice Transmission Quality Table 4

2169 |  | Optimal media packet size table

0187 |  | Expensive Route warning tone

0188 |  | Voice transmission quality

0190 |  | Switch to non-IP network allowed ?

#### Networking related commands

#### IP Link settings

1912 |  | Termination in ASB 150

1914 |  | Subscribed charging method / Requested charging method

1915 |  | Calling number allowed to network?

1918 |  | Connected line presentation on

Following commands do not need to be modified, they can remain on the default value

1911 |  | Network signalling tables

1922 |  | Through connect after SETUP to the net

#### General settings

2084 |  | Type of network call handling

2085 |  | PBX protocol identifier

#### Programming of the “Own location code”

5431 |  | Create own location code

5531 |  | Delete own location code

5631 |  | Alter own location code

#### Programming of the Private Network Router (PNR)

5432 |  | Create private network access code

5532 |  | Delete private network access code

5632 |  | Alter private network access code

9101 |  | Destination PBX

9102 |  | Call-back allowed

9103 |  | TON / private calling

9104 |  | Call Class

9105 |  | TON public calling

##### 1 st route choice

9110 |  | 1st route choice

9111 |  | TON: priv / pub called

9112 |  | Remove digits by using 1st route choice

9113 |  | Predigits for the 1st route choice

9114 |  | Travelling Class Mark

9115 |  | Complete call party number length

9116 |  | Subscriber called party number length

9117 |  | IP Reference number

##### 2nd route choice

9120 |  | 2nd route choice

9121 |  | TON: priv / pub called

9122 |  | Remove digits by using 2nd route choice

9123 |  | Predigits for the 2nd route choice

9124 |  | Travelling Class Mark

9125 |  | Complete call party number length

9126 |  | Subscriber called party number length

9127 |  | IP Reference number

9128 |  | Expensive route warning tone

##### 3rd route choice

9130 |  | 3rd route choice

9131 |  | TON: priv / pub called

9132 |  | Remove digits by using 3rd route choice

9133 |  | Predigits for the 3rd route choice

9134 |  | Travelling Class Mark

9135 |  | Complete call party number length

9136 |  | Subscriber called party number length

9137 |  | IP Reference number

9138 |  | Expensive route warning tone

##### 4th route choice

9140 |  | 4th route choice

9141 |  | TON: priv / pub called

9142 |  | Remove digits by using 4th route choice

9143 |  | Predigits for the 4th route choice

9144 |  | Travelling Class Mark

9145 |  | Complete call party number length

9146 |  | Subscriber called party number length

9147 |  | IP Reference number

9148 |  | Expensive route warning tone

#### PNR alternative Routing

1037 |  | PNR route choices at incoming proprietary calls

#### Basic call service – Travelling Class Mark

0171 |  | Travelling Class Mark (TCM)

1035 |  | Travelling Class Mark (TCM)

#### Basic call service – Transit counter

2087 |  | Transit counter

#### Calling/Connected Line/Name identity

0170 |  | Restriction of private identity presentation

2220 |  | Public directory number

2236 |  | Own country code

2237 |  | Own area code

1936 |  | International prefix to be included

1937 |  | National prefix to be included

The following commands need to be set if the public exchange transmits leading zeros in the calling/connected number information. Leading unremoved zeros will automatically cause problems, if the gateway exchange has to modify the received number before it is sent to the network. The commands are also necessary if the connected network node transmits more predigits than necessary e.g. the trunk access code.

1960 |  | Predigits to be removed for calling / connected number if TON=Unknown

1961 |  | Predigits to be removed for calling / connected number if TON=International

1962 |  | Predigits to be removed for calling / connected number if TON=National

1963 |  | Predigits to be removed for calling / connected number if TON=Network-specific

1964 |  | Predigits to be removed for calling / connected number if TON=Local directory

#### Supplementary service UUS

2072 |  | MD110 (ASB 501) Release version of CN

Following commands do not need to be modified, they can remain on the default value

1940 |  | UUS services subscription

1942 |  | UUI length service 1

1943 |  | UUI in ALERTING?

1949 |  | UUI in Generic Functional Protocol?

#### Proprietary service – Call back

0177 |  | Initiate call back to CN?

9001 |  | Call setup version at network Call back

9002 |  | Active time for a “CB on busy” mission in the init. PBX (T1)

9003 |  | Active time for a “CB on free” mission in the init. PBX (T1)

9004 |  | Active time for a “CB on busy” mission in the superv. PBX (T2)

9005 |  | Active time for a “Call back on free” mission in the supervised PBX (T1)

9006 |  | Maximal ringing time at the initiating extension for Call back calls

#### Proprietary service – Call diversion

0176 |  | Call diversion to a CN number allowed

#### Proprietary service – Camp on

0108 |  | Camp on allowed?

#### Proprietary service – Intrusion

0182 |  | Initiate of network intrusion allowed?

#### Proprietary service – Message Waiting Indication

2023 |  | External voice mail number

2071 |  | ISID of external voice mailbox

#### Proprietary service – Rerouting

1110 |  | 1st choice CN reroute position

1111 |  | 2nd choice CN reroute position

1112 |  | 3rd choice CN reroute position

2077 |  | 1st central answer position

2078 |  | 2nd central answer position

2079 |  | 3rd central answer position

### Start of operation

Read out the MAC-address of the IPU-board and provide this information to the LAN administrator for the IP address administration.
Set up the command 2151 Preferred start-up protocol according to the available services in the customer‘s LAN (DHCP,BOOTP or static IP-Address).
In case of choosing DHCP or BOOTP, it is required to invoke a board or system restart to request an IP address from the DHCP or BOOTP Server during start up.

### Test the Installation

Please refer to the document Start of Operation, (1537-ASB 150 02 Uen)

## Maintenance

For upgrading the firmware of the IPU as well as the licensed voice channels please refer to document Maintenance Instructions, (1541-ASB 150 02 Uen), 1541-ASB 150 02 Uen.

## Equipment/Environment

### IPU board

In oder to use the function “Networking via Voice over IP lines” the system has to be equipped with an IPU board.

#### Licensed voice channels

The number of available voice channels on the IPU board will be defined by a license file which is located on the Compact Flash card. The Compact Flash card itself will be plugged on the IPU.

For detailed description about the IPU licensing, refer to document Maintenance Instructions, (1541-ASB 150 02 Uen), 1541-ASB 150 02 Uen.

For a detailed description of the license ordering process see Orderinformation BusinessPhone Communication platform 113 62-11/ASB150 02

#### Supported voice channels

Apart from the Compact Flash card, the IPU has to be equipped with a DSM submodule in order to support the voice channels, and the amount of supported voice channels has to be equal to the amount of licensed voice channels.

One DSM submodule supports 8 (R12) / 16 (R13) bi-directional voice channels.

Article number | Function
ROA 219 5114/1 | DSM1 16 bi-directional voice channels in conjunction with the G.711 voice codec. If the G.729 Annex A (with Annex B) or G.723.1 voice codecs are used only 8 bi-directional channels are supported.
ROA 219 5114/2 | DSM2 16 bi-directional voice channels, independent of used voice codec.

### Networking via IP lines

In order to have access to the services of a given Networking package, a proper FECU must always be plugged-in on the CPU-D4.

For more information regarding the Networking Packages, see Facility Description – General (155 34-ASB 150 02 Uen), chapter “Functionality Of
ASB 150 02 With/Without Connected FECU”.

### Pre-requisites to the customer‘s LAN environment

In order to build up an ASB 150 02 network with VoIP lines, the following prerequisites have to be fulfilled.

- LAN environment
  ASB 150 02 provides an interface to an Ethernet 10BaseT/100BaseTX autosense LAN environment according to IEEE-802.3 for half- and full duplex operation.
- TCP/IP (IP Version 4)
- Between the different network nodes which are to be connected via VoIP lines, a dedicated LAN/WAN connection (corporate Intranet) must exist. The system is not intended to be connected to the Internet.
- Bandwidth
  In order to avoid speech delay variation or conversation interruption (due to packet loss) the traffic situation on the LAN segments used for the VoIP lines should be analysed. Depending on the selected voice transmission quality, a certain bandwidth will be occupied per voice call (see following table).

#### Codec / Bandwith and packet size table

The following table includes parameters to optimise LAN configuration for VoIP operation ( Ethernet packet size,.....)

Codec | Bandwidth | Packet duration | Packet size | Effective bandwidth per channel (one direction) | Comment
[kbit/s] | [ms] | [byte] | [kbits/s]
G.711 | 64,0 | 60 | 480 | 72,8 |
| 40 | 320 | 77,2 |
30 | 240 | 81,6 | default
20 | 160 | 90,4 |
G.723.1 | 5,3 | 120 | 80 | 9,7 |
| 90 | 60 | 11,2 |
60 | 40 | 14,1 |
30 | 20 | 22,9 | default
G.729A(B) | 8,0 | 120 | 120 | 12,4 |
| 90 | 90 | 13,9 |
60 | 60 | 16,8 |
30 | 30 | 25,6 | default
10 | 10 | 60,8 |

#### The following formulas have been used:

- Packet size [byte] = Codec [kbit/s] * 125 * Packet duration [ms] / 1000
- Eff. bandwidth [kbit/s] = (Sum of all Headers [byte] * 8 / Packet duration [ms]) + Codec [kbit/s]
  For the table above: Overhead for each sent RTP package: 26 byte full Ethernet header + 20 byte IP header + 8 byte UDP header + 12 byte (minimum!) RTP header = 66 byte

Note

Codec G.711 offers the best speech quality. The bigger the packet size the longer the speech delay. Therefor the IPU limits the paket size to 480 byte.

## Glossary

Term | Explanation
10BaseT | LAN-standard for 10Mbit/s networks using twisted pair cabling (2pairs).
100BaseTX | LAN-standard for 100Mbit/s networks utilizing 2 pairs of data grade category 5 unshielded twisted pair (UTP) and shielded twisted pair (STP) cabling. 1 pair is used for the data transfer, the other pair is used for collision detection
CAT 5 | Category of a UTP LAN cable which is used for data transfer up to 100MHz (including 100BaseTX and ATM 155Mbps)
CN | Corporate Network
CNP | Corporate Number Plan
Codec | Compression/decompression. An overall term for the technology used in digital video and digital audio.
DHCP | Dynamic Host Configuration Protocol. A TCP/IP protocol allowing the principal parameters of network devices (including IP addresses) to be configured by central DHCP servers. DHCP's purpose is to enable individual computers on an IP network to extract their configurations from a server (the 'DHCP server') or servers, in particular, servers that have no exact information about the individual computers until they request the information. The overall purpose of this protocol is to reduce the work necessary to administer a large IP network. The most significant piece of information distributed in this manner is the IP address.
DSM | Digital Signal Processing Module
Ethernet | A high-speed local area network (LAN), consisting of a cable technology and one or more sets of communication protocols. Developed by the Xerox Corporation, Ethernet is based on the IEEE 802.3 standard and uses the carrier sense multiple access with collision detection CSMA/CD) method to control access to the transmission medium.
Gateway | An entrance into and out of a communications network. Technically, a gateway is an electronic repeater device that intercepts and steers electrical signals from one network technology to another.
IP address | A unique, 32-bit number for a specific TCP/IP host on the Internet. Usually printed in decimal form (for example, 128.122.40.227)
IP Telephony | Technology that lets you make voice phone calls via gateways and standard telephones over the Internet or other packet networks using your PC.
LAN | Local Area Network. A LAN is a group of computers and other devices which are connected by a communications link, allowing the devices to interact on the shared network. It is confined to a small geographical area that does not use long-distance carriers, such as telephone connections. The area is usually limited by the cable length restrictions of the transportation media being used. LANs allow users to communicate and to share resources, such as hard disk storage and printers. A LAN is user-owned and does not run over leased lines, although it can have gateways to the PSTN or other private networks.
MAC address | A MAC address (also called an Ethernet address or an IEEE MAC address) is a number (typically written as twelve hexadecimal digits, 0 through 9 and A through F, or as six hexadecimal numbers separated by periods or colons, i.e. 0080002012ef, 0:80:0:2:20:ef) which uniquely identifies a computer that has an Ethernet interface. Unlike the IP number, it includes no indication of where your computer is located.
QoS | Quality of Service. Guarantees network bandwidth and availability for applications.
Packet Transmission | Data files are split into packets and sent through the packet-switched network. Each packet may take a different route. The receiving side collects the packets, puts them into the correct order and reassembles the data file again.
PNR | Private Network Route
subnet | A portion of a network that may be physically independent of another network portion, but both portions of the network share the same network address, and the portion is distinguished by a subnet number. The sub net number is a portion of the Internet address that designates a subnet. It is used for intranet routing but is ignored for Internet-routing purposes.
subnet mask | The bits of an IP address used for a subnetwork.
TCP | Transmission Control Protocol A connection-oriented protocol in Layer 4, that ensures that the data will arrive in the correct order. A connection is set between two communication devices to send and receive data. If this data is out of order, TCP reorders the information. If the expected information has not been received, it is requested again.
UDP | User Datagram Protocol. UDP acts parallel to TCP in Layer 4 and provides a connectionless datagram delivery service between IP host applications. Connectionless means that the protocol in the end device indicate whether datapackets were lost on the way. UDP avoids the overhead of reliable data transfer mechanism by not protecting against datagram loss or duplication. For some applications, such precision is not necessary. Voice is an example. In a voice transmission, if one or two bits are wrong, the conversation will not become unintelligible. Even if an entire datagram is lost, the people at the ends can repeat themselves, and in that way recover the lost information. For these reasons, as well as the greater simplicity of UDP versus TCP, VoIP applications primarily use UDP.
UDP-pair | A UDP pair consist of two UDP one way channels whereby the first channel is used to transmit UDP packets with RTP-Header and voice and the second one is used for the RTCP protocol.
UTP | Unshielded twisted pair cable
Voice over Internet Protocol (VoIP) | Founded in 1996 by Cisco, Dialogic, Microsoft, US Robotics, VocalTec, and several other leading firms, VoIP is working to develop and promote standards for IP telephony. The VoIP efforts consist primarily of building on and complementing existing standards, like H.323.
WAN | Wide area network. A network which covers a larger geographical area than a LAN and where telecommunications links are implemented by the appropriate PTT(s).

Revision: PC4 © Ericsson Enterprise 2003