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The term LAN telephony refers to the communication between IP stations in an internal network (LAN). To ensure the quality of the voice transmission in LAN telephony, the IP networks being used and the communication system must meet certain requirements. The voice quality and voice communication reliability always depend on the network technology in use.

To guarantee loss-free transmission and good voice quality, voice signals are digitized using audio codecs and marked using special procedures (Quality of Service) so that voice transmission has priority over data.

Stations LAN Telephony Requirements

Requirements

• LAN with at least 100 Mbps and full duplex

• Every component in the IP network must be connected to a separate port on a switch or to a router; a hub should not be used.

• Not more than 50 msec delay in one direction (One Way Delay); not more than 150 msec total delay

• Max. 3% packet loss; if a fax/modem via G.711 is used, the packet loss must not exceed 0.05%.

• Not more than 20 msec jitter

• Support for Quality of Service (QoS): IEEE 802.p, DiffServ (RFC 2474) or ToS (RFC 791)

• Maximum 40% network load

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7.2.1 Audio Codecs

An audio codec is a program that encodes and decodes voice in digital data packets (IP packets). The data compression rate can vary depending on the audio codec used. The bandwidth requirement for transferring an IP packet is lower if the packet is compressed. The decoding of data packets can, however, have a negative impact on voice quality and the playback continuity.

The recipient and sender must use the same codec to ensure that the data can be correctly decoded back into voice after transport.

Supported Audio Codecs

The following audio codecs are supported:

• G.729A, G.729AB: voice encoding at 8 Kbps - good voice quality.

• G.711 (A-law and µ-law): voice encoding at 56 or 64 Kbps - very high voice quality. G.711 is also used in fixed networks (ISDN).

The audio codecs can be assigned priorities between 1 (high) and 4 (low). The communication automatically tries to use the audio codec with the highest priority available for every connection. Using an audio codec with low voice compression (good voice quality) increases network load. In the case of intensive IP telephony, this can lead to diminished voice quality in a network already overloaded by data transfers.

The communication system can enable voice activity detection (VAD) for certain codecs. This can reduce network load during long voice pauses.

You can specify a frame size (IP packet size) of 10 to 90 msec for every codec.

This specifies the sampling rate at which the audio codec splits the voice signal into IP packets. While a higher value (90 msec, for instance) results in a better relationship between payload and the IP packet overhead, it also increases the transfer delay.

It is possible to disable the resource-hungry G.729 codecs and to only use the G.711 codecs. This optimizes the number of possible simultaneous calls. If this function is enabled, the system must be restarted.

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Stations

LAN Telephony Requirements

7.2.2 Transmission of Tones According to RFC 2833

The transmission of DTMF tones and fax/modem tones according to RFC 2833 can be enabled or disabled.

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7.2.3 Quality of Service

Quality of Service (QoS) encompasses various procedures for guaranteeing the highest possible quality and integrity during the transmission of data packets (IP packets). For good voice quality during voice transmission, QoS is used in the IP network to give IP voice packets priority over IP data packets from other

applications.

The IP packets are assigned a special marker (code point) for prioritization. The marker is set in the IP-packet control information. Categorization in different classes is performed based on priority information. If the components available in the IP network (communication system, SIP stations, and Internet routers, for instance) support QoS, you can assign different bandwidth to these classes and thus transport the IP voice packets first.

AF/EF Code Points

For DiffServ-based prioritization, two different code points are defined so that IP-packet transmission can be split into different classes.

• Expedited Forwarded (EF) Code point: guarantees constant bandwidth. The bandwidth is always the same for IP packets marked with this code point.

Once the set value is reached, all IP packets that exceed this bandwidth are dropped.

• Assured Forwarding (AF) Code point: guarantees minimum bandwidth. IP packets that are marked with this code point have a lower priority than EF and must share the bandwidth not used by EF. Once the set value is reached, all IP packets that exceed this bandwidth are rejected.

Four classes are reserved for AF: AF1x (low priority), AF2x, AF3x and AF4x (high priority), where "x" stands for one of three dropping levels: low (1), medium (2) and high (3). In the case of "low", packets are buffered over an extended period, in the case of "high", packets are promptly rejected if they cannot be forwarded. Unmarked IP packets (ToS field=00) are handled in the same way as the lowest priority.

You can set the code point used for marking the IP packets to be transmitted for the following transmission types.

• Signaling Data: for the transmission of signaling data for connection startup and cleardown in IP telephony

• Voice Payload: for voice transmission in IP telephony. Code point EF is the recommended setting here.

• Fax-/Modem-Payload: for fax/modem data transmission in IP networking, for example

• Network Control: for transmitting SNMP traps, for example

Stations IP Stations

The AF/EF code points can be displayed in the form of hexadecimal values.

Priority classes

The priority classes for transmission types can be set in both of the following forms:

• Layer 3 Prioritization - EF/AF code points:

Application in the WAN, e.g., preferred transmission of IP packets via a router.

The following values can be set in addition to the EF/AF code points:

– Best Effort: Best Effort can be used to mark packets that do not require any prioritization, e.g., for the administration.

– CS7: Class Selector 7 can be used to mark important network services such as SNMP packets, for instance.

• Layer 2 Prioritization - Layer 2 QoS values from 0 To 7:

Application in the VLAN, e.g.,preferred transmission of IP packets between switches.

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7.3 IP Stations

IP stations are connected to the communication system via the LAN. An IP station is generally a LAN or WLAN phone.

The following IP protocols are supported:

• Vendor-specific communication system protocol

The communication system uses CorNet-IP (CorNet Internet Protocol) for LAN telephony within the internal network. CorNet-IP, which was developed on the basis of H.323, supports all telephone features of the communication system.

• SIP (Session Initiation Protocol)

SIP is usually used in Internet telephony but is not restricted to it. It can also be used for telephony in the internal network, for example. However, SIP does not support all telephony features associated with the communication system.

The following types of IP stations exist:

• System Client: A system client is an IP station that can use all the features of the communication system via CorNet-IP. This can be an IP system phone such as an OpenStage 60 HFA, for instance, or a PC with CTI software such as OpenScape Personal Edition.

• SIP client: A SIP client is an IP station that uses the SIP protocol. It can access only limited functionality of the communication system via SIP. A SIP client is a SIP phone such as the OpenStage 15 S, for example.

• Deskshare User: A Deskshare User is an IP user who can log in at another IP system telephone (mobile login) and then use this phone as his or her own phone (including the call number).

• RAS User: A RAS user (Remote Access Service user) is granted Internet access to the IP network via the ISDN connection. This allows the

communication system to be maintained remotely.

For each connected IP station, an "IP User" station license is required.

Stations SIP Stations

Two IP stations are reserved for the Online User and for remote access via ISDN.

These IP stations do not require a station license. If one or several of these three reserved IP stations are not required, these stations can be converted to normal IP stations in Expert mode. However, station licenses are then required for these IP stations.

Configuring IP Stations

The following configurations can be performed for an IP station:

• Configuration of standard parameters with the IP Telephones wizard (see Administrator Documentation, Stations).

• Configuration of all parameters (standard and advanced parameters in Expert mode (see Administrator Documentation, Stations).

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