fd en extract iee5

Features of Integral Enterprise

Extract of enhancements for SW Release IEE 5

5 - EnglishOctober 2008

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Features Extract IEE 5 10/2008 3


Extract of chapters with enhancements for the software release IEE 5

This document describes the features of Integral Enterprise for business and technical staff.

The web presentation allows direct access to each individual feature. A comprehensive index will help tolocate individual terms.

In addition, the footer on each page contains links to the changes made to the individual versions and tothe table of contents.

The general descriptions are available in six languages. In addition, technical notes with setup aids in MMLin are appended to the descriptions of the features in German and English.

© 2008 Avaya Inc. All rights reserved.

Release Integral Enterprise Edition 5Updated: IEE 5, October 2008

The software Integral Enterprise Edition 5 (=IEE 5) extends scalability, redundancy and encryption forVoice over IP and especially for SIP. Specifically, it offers the following changes compared to IEE 4:

• The functions of Voice over IP [ → 19 ] are improved through enlarged Expansion limits [ → 24 ], improved Security [ → 30 ] and additional Redundancy [ → 27 ].

• Enlargement of the SIP functionality by connecting the SIP Enablement Server [ → 15 ] and additional encryption.

• The connection of EVOip active SIP [ → 16 ] enables voice recordings via the SIP interface.

• In networks over IP-QSIG [ → 10 ] encrypted communication is usable between systems Integral Enterprise.

• The implementation of SNMP supports additional SNMP-Traps for the Network Management Console [ → 7 ].

• The VoIP-Manager is provided in version 2.0, look at VoIP-Manager [ → 29 ].

Release Integral Enterprise Edition 5


Contents


ContentsFeatures of Integral Enterprise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Release Integral Enterprise Edition 5 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

Contents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

Network Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 Network Management Console . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Management Information Base . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8 VoIP Monitoring Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

QSIG/QSIG+ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 Networking Communications Manager and Integral Enterprise . . . . . . . . . . . . . . . . . . . . . 10

Service features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11 Benefit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

SIP = Session Initiation Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12 SIP subscribers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13 SIP lines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14

SES = SIP Enablement Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 EVOip active SIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

SIP standards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

Voice over IP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19 Useable features (Start) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19 Useable features (Continuation) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22 Legend . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 Enhancements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 Expansion limits . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26 Modes and Compatibility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26 Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27 IPMR protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27 Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28 VoIP-Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30 IP users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 Media Streams . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 IP user mobility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Voice recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

List of attachments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Last Page . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

6 Features Extract IEE 5 10/2008

Network Management


Network ManagementFor integration into the Avaya network from IEE 3, the Simple Network Management Protocol (SNMP) issupported and messages are send to the VMM by means of the RealTime Control Protocol (RTCP).

•With SNMP it is possible to transmit data via Integral Enterprise to the Network Management Console (= NMC) and to use the data for network administration purposes.

•With RTCP, the VoIP speech quality on the VoIP gateway is determined and transmitted to the Avaya VoIP Monitoring Manager (VMM).

From IEE 4 the representation of Integral Enterprise and its IP users has been improved.

Network Management ConsoleIEE 3 supports the Simple Network Management Protocol (SNMP). With this it is possible to transmit datavia Integral Enterprise Communication Manager to the Network Management Console and to use the datafor network administration purposes. In addition, it also enables user-friendly monitoring of all networkcomponents.

From IEE 4 onward, the representation allows a distinction to be made between Integral Enterprise andMG100.

• SNMP support • SNMPv3 (= Simple Network Management Protocol)• SNMP agent on Master Server Blade (ACB) • As an object, Integral Enterprise is a part of the Network Management Console (= NMC)• Network Discovery and view in Avaya Integrated Management

In the left-hand pane the entire network can be seen as a tree structure from which individual objects canbe selected, with Communication Manager or Integral Enterprise PBXs. The right-hand pane shows thedetails for the selected object: e.g. status, PBX name, IP address, system type and (for telephones) alsothe subscriber number.


With IEE 4, Integral Enterprise sends SNMP traps for certain events to NMC among others. As a result,alarm messages from Integral Enterprise can trigger certain responses, such as fault notification via SMS.The list of SNMP traps was significantly expanded for IEE 5.

Management Information Base

The Management Information Base (= MIB) describes informations, which can be queried or modified viaSNMP.

AIC Message from Version

2 Failure of uninterruptible power supply (UPS), inactive module IEE 43 Failure of uninterruptible power supply (UPS), active module IEE 4

98 Failure of a call control group IEE 4126 Partial failure of central charing task (ZGDE) IEE 4127 Total failure of of central charing task (ZGDE) IEE 4231 Hard disk fill level (file SQF-FEAUHS.SMF) exceeds 87,5% IEE 4399 Failure external sync. line not usable IEE 4611 Module fan failure (of MG1000) IEE 4613 Fan power supply failure IEE 4651 ACT file full IEE 4675 Hard disk failure (HGS) IEE 4

53 Dialog device defective (attendant console) IEE 569 Locking status of a connecting device has changed IEE 577 Analog line faulty IEE 583 General error from task MSU1 IEE 5

555 Digital line faulty IEE 5615 ESU external signalling: Level change of signal 1 to low IEE 5616 ESU external signalling: Level change of signal 1 to high IEE 5617 ESU external signalling: Level change of signal 2 to low IEE 5618 ESU external signalling: Level change of signal 2 to high IEE 5619 ESU external signalling: Level change of signal 3 to low IEE 5620 ESU external signalling: Level change of signal 3 to high IEE 5621 ESU external signalling: Level change of signal 4 to low IEE 5622 ESU external signalling: Level change of signal 4 to high IEE 5

Network Management


VoIP Monitoring ManagerWith IEE 3, the speech quality is monitored in Integral Enterprise for the VoIP speech data. Here, the Qua-lity of Service (QoS) parameters on the VoIP gateway are determined with the aid of the RealTime ControlProtocol (RTCP) and transmitted to the Avaya VoIP Monitoring Manager.

• VoIP Performance Monitoring • Collection of speech quality data on the VoIP gateway by means of RTCP:

Jitter (in ms), round trip time (in ms), packet loss (in %)• Real-time monitoring and automatic alarm generation• Cyclic transmission of reports to the VoIP Monitoring Manager (VMM)

From V4, the VoIP Monitoring Manager can present the speech quality in both directions in conjunctionwith the T3 IP telephone software from C Packet Step III onward - i.e. from the PBX and from the tele-phone (further example displays can be found later on in the document).

For notes on activation of monitoring with the aid of the real-time control protocol and on the setting up ofthe VoIP Monitoring-Managers please refer to the handling-specification:

• HSP_1776: VMM Support in Integral Enterprise

You will also find examples of RTCP messages and other VMM displays here as well.


QSIG/QSIG+QSIG+ has the all the functions of the standard QSIG as well as those of the Avaya protocol TNet.

The QSIG protocol was standardized so that it could be used to establish mixed networks with telecommu-nication systems made by other manufacturers. In addition, it also permits company-specific expansion.

Many manufacturers use proprietary protocols to build homogeneous networks with their own systems. Alt-hough this does offer the potential for unique selling points and competetive advantages, it also requirestwo interfaces to be maintained for networking:

1. in mixed networks through QSIG,2. in homogeneous networks through a proprietary protocol.

By contrast, Avaya uses the protocol QSIG+ in all networks. This simplifies maintenance and enables cer-tain features to be implemented across the network.

The standard H.323 protocol with QSIG extensions is used for QSIG over IP.

From IEE 5, secure communication can be used between systems Integral Enterprise in networks over IPQSIG tunnel:

• The signaling is encrypted by TLS (= Transport Layer Security).• The media streams are encrypted by AES (= Advanced Encryption Standard),

the AES keys are transmitted by QSIG RAS.• FAX over G.711 is also encrypted, but Fax over T.38 is not encrypted.

For details, also for setting up the encryption, please refer to the handling specification• HSP_1747: Connectivity of Communication Manager and Integral Enterprise using QSIG over IP.

Networking Communications Manager and Integral EnterpriseFrom IEE 3 onward, Communication Manager (CM) and Integral Enterprise (IE) can be networked to eachother over ISDN QSIG and IP QSIG.

Networking over ISDN QSIG is based on the time division multiplexing process in the traditional networkconnection technology. For details, please refer to the handling specification

• HSP_1771: CM-IE/Connected by TDM QSIG.

For QSIG over IP, the standard H.323 protocol with QSIG extensions is used.

QSIG/QSIG+


For details, also for setting up the encryption, please refer to the handling specification • HSP_1747: Connectivity of Communication Manager and Integral Enterprise using QSIG over IP.

Here is an overview of Possible Services and a Feature comparison in different networks VPN,IPN,TNet withQsig,Qsig+ .

To create a PBX network with Communication Manager and Integral Enterprise, QSIG lines and QSIG datamust first be configured in both PBXs.

Communication Manager 3.1 has been extended by the addition of the following QSIG features: • ring tone distinguishes between internal and external calls,• as of Communication Manager 4.0, umlauts for display are decoded according to ISO 8859-1,• the external signalling of the two PBXs has been aligned, improving the previously restricted function

of a central operator position.

Service features

QSIG is implemented in accordance with the ISO standard in AVAYA Communication Manager.

Although QSIG is a standardised protocol, individual QSIG features are implemented differently in the twoPBXs - Integral Enterprise and Communication Manager. This results in minor differences in the behaviourof the PBXs concerning some features such as call diversion and route optimisation or over the jurisdictionof a central operator position.

The following features are supported:• Basic Call (BC)• Generic Functions• CCBS/CCNR• Diversion• Call Offer• Call Transfer• Line Identification• Name Identification• Path replacement• Message waiting• Advice of charge (as of Communication Manager 4.0)

Communication Manager may use the Integral Enterprise as a server for DECT, including network-wideroaming.

3.

BenefitProduct maintenance and repair are made easier.


SIP = Session Initiation ProtocolFrom IEE 3, Integral Enterprise has SIP interfaces for subscribers and lines.

The Session Initiation Protocol (SIP) is a network protocol for establishing a communication session withmultiple subscribers. It follows the architecture of standard Internet applications. The protocol is specifiedin RFC 3261 (previously RFC 2543).

SIP is used for the administration of any session with multiple subscribers. The actual data for communica-tions are exchanged via other protocols which are suitable for the purpose.

As well as Avaya devices, it is also possible to connect various other manufacturers’ devices via the openinterface. The list below contains a number of suitable devices. Corresponding reports about interoperabi-lity can be found in the Technology Center.

• Avaya SIP one-X Deskphones 9620, 9630 and 9640• Avaya SIP IP Phones 4602SW, 4610SW and 4621SW,• Avaya one-X Desktop Edition (SIP-Softphone)• Cisco 79xx SIP• SNOM 3xx• Nokia E60• Counterpath eyeBeam V1.5

For more details please refer to the handling specification • HSP_1754: Integral Enterprise SIP.

From IEE 5, SIP participants may register in an Integral Enterprise via the SIP Enablement Server (SES).

1. Conventional peripherals2. T3 IP phone with tunneling3. SIP phone4. SIP softphone, Windows XP5. SIP Wifi phone (Wireless Fidelity) 6. SIP Wifi PDA (option for the future)

7. W-LAN AccessPoint (IEEE 802.11b)8. SIP line for connecting peripherals, for networking or

for establishing a connection to the carrier.9. UMS solution C3000 with XCAPI (Unified Messaging

System)10. SIP Enablement Services = SES (from IEE 5)

SIP = Session Initiation Protocol


SIP subscribersParticipants at the SIP interface have the following features from version IEEx onward:

Especially, the communication of SIP participants is encrypted from IEE 5 onward, the signaling by TLS (=Transport Layer Security) and the media streams for participants in the IPMR by SRTP (= Secure RealtimeTransport Protocol).

Features: from versionRegistration (RFC 3261) IEE 3Secure login with PIN (MD5) IEE 3Basic call (RFC 3261) IEE 3Distinctive ringing (distinction between internal and external calls) IEE 4Subscriber number display (RFC 3261) IEE 3Suppression of the number display IEE 3Name display (RFC 3261) IEE 3Suppression of the name display IEE 33-way conference (mixer in the terminal device/User Agent) IEE 3Call waiting IEE 3Message waiting indication (MWI, RFC3842) IEE 4Call hold / Retrieve / Music on Hold (MoH) IEE 3Call transfer IEE 3DTMF post-dialing (RFC 2833) IEE 3Block dialing IEE 3Transport with TLS (= Transport Layer Security) IEE 5DNS and Domain name support IEE 5additional features via Feature access codes (with # and *)

• Call diversion (CFU)• Call diversion if busy (CFB)• Call diversion if no response (CFNR)• Call diversion if not logged in (CFNLI)• Abbreviated dialing (without post-dialing)• Follow me• Emergency call• Pick up• Double call• Hunt group• General clearing• Call tracing (MCID)

IEE 3

additional features via Comfort buttons (if registered at SES) • Call Forwarding (CFU) • Call Forwarding Busy (CFB) • Directed Pickup

IEE 5


SIP linesThe interface for SIP lines offers several options for connectios:

1. From IEE 5, the SIP Enablement Server (SES) [ → 15 ] can be linked on this interface. It provides the new features Instant Messaging and Presence for SIP participants and also additional comfort but-tons for Avaya SIP telephones.

2. The interface allows IP connections to various carriers. For example, profiles are available there for British Telecom (BT), Deutsche Telekom or Arcor. For corresponding interoperability reports please refer to the Technology Center.

3. From IEE 3, the SIP interface serves for connecting various applications. It is compatible with the uni-fied messaging system C3000 from v1.7 (UMS on XCAPI with SIP). From IEE 5, Recordings are pos-sible via VoIP with EVOip active SIP [ → 16 ].

The following features are available from version IEEx onward:

With XCAPI, the following additional features are also available:

Features: from versionExtension IEE 3Remote number ID IEE 3Suppression of the number display IEE 3Name display IEE 3Suppression of the name display IEE 3Transport via TLS (= Transport Layer Security) IEE 5

Features: from versionMessage waiting indication (prefix) IEE 3Transmission of the redirecting number (IETF Draft: draft-levy-sip-diversion-08: Diversion indication in SIP)

IEE 3

Call transfer IEE 4

Supported Codecs: from VersionG.711 IEE 3G.729A IEE 3T.38 for Fax over IP in Realtime IEE 4DTMF - Transmission through RFC2833 IEE 3SRTP (= Secure Real Time Protocol) IEE 5 at IPMR



SES = SIP Enablement Server

The SIP Enablement Server extends the SIP scenario of the Integral Enterprise from IEE 5 onward. Itallows the features Instant Messaging and Presence for SIP participants and in addition Comfort But-tons for Avaya SIP telephones.

SIP phones are connected to the Integral Enterprise either directly through the dedicated interface for SIPparticipants or indirectly through the SES. The SES itself is connected to the Integral Enterprise through aSIP line.

SIP phones can register at exactly one SES (without redundancy). A SIP participant connected to the Inte-gral Enterprise via SES is called a SES participant. The SIP line, which connects the SES to the IntegralEnterprise, is called SES line.

A PBX serving SES participants is called a Media Server. A representative of a SES participant in the Inte-gral Enterprise is called a Media Server participant.

The VoIP board supports one SES line and 239 SES participants. The possible number of SES participantsmay be increased by means of further VoIP boards as additional media servers on the same SES. Thenew IPMR board supports one SES line and 999 SES members.

Further details, especially on the setting up, please refer to the handling specification • HSP_1760: SIP SES Integration in Integral Enterprise.

Instant Messaging and Presence

SES participants with an Avaya one-X Desktop Edition (SIP softphone) may use the feature instant mes-saging (= IM). They may send short text messages to each other via phone, in other words they may chatto each other. This direct contact is an additional way of cooperation.

The basis of this cooperation are presence informations of the possible partners, displayed at the SIP pho-nes of Avaya. They indicate whether a desired partner is just ready for talk.

Detail (from IEE 5):

10. SIP Enablement Server = SES 4. SIP Softphone, Windows XP


The Avaya one-X Desktop Edition (SIP softphone) displays icons according to the presence states.

When phoning the status "On the phone" is set automatically. The SES participants can set all states byselecting from a menu bar. If he does not want to publish his presence status, he can simply set "Invisible".

Comfort Buttons

The user interface of the Avaya SIP Telefone one-X Desktop Edition, connected to the SES, can displaycomfort buttons. They serve to activate or deactivate the corresponding features similar to the selection offeatures via Feature Access codes.

From IEE 5 onward the following Comfort Buttons are available: • Call Forwarding (CFU)• Call Forwarding Busy (CFB)• Directed Pickup

Remark: Assuming the corresponding features are activated as system features and also as participantfeatures.

EVOip active SIP

The software EVOip active SIP at the Integral Enterprise from the company ASC allows the recording oftelephone conversations, both of unencrypted as well as encrypted conversations.

Conversations are recorded directly over the network, stored and archived optionally. Control of the recor-ding, as well as search and playback of recorded telephone calls are possible.

The recording is set for certain agents, either for incoming or outgoing calls or for both directions. Theagent cannot find out, when his conversations are recorded, neither by display nor by inserted tone. But hecan, however, mute certain sections of conversation by keystroke.

Available Busy Away On the phoneInvisible



If the partner is an internal participant, his display shows "Cut-in" during the recording (if his phone sup-ports this advice). He can neither start an inquiry nor a conference during the recording.

The following terminal types can be recorded as agents: digital participants, DECT participants, VoIP parti-cipants and stimulus participants. Moreover, the following conditions must apply for agents:

• the agent is connected to the S-Ref• the agent is neither attendant console nor dealer nor analog line or participant nor TNET• the agent is no line (Q T-Ref)• the agent is not connected to a MAC board• the agent is not recorded otherwise (by EVOip or Silent Monitoring)• the agent does not have a Conference and cannot start one• the agent may not have any Cut-in status

Further details, especially on the setting up, please refer to the handling specification • HSP_1782: CSTA Service Single_Step_Conference.

SIP standards

The following standards are supported from versionRFC1889 RTP: Real-Time Transport Protocol IEE 3RFC2327 SDP: Session Description Protocol IEE 3RFC2396 URI generic syntax IEE 3RFC2617 Digest Authentication IEE 3RFC2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals IEE 3RFC3261 SIP: Session Initiation Protocol ("SIP 2.0"), IEE 3RFC3262 Reliability of Provisional Responses in the Session Initiation Protocol

(SIP)IEE 3 partial

RFC3264 An Offer/Answer Model with the Session Description Protocol (SDP) IEE 3RFC3310 UPDATE Method IEE 3RFC3325 Private Extensions to the Session Initiation Protocol (SIP) for Asserted

Identity within Trusted NetworksIEE 3

RFC3332 A Privacy Mechanism for the Session Initiation Protocol (SIP) IEE 4RFC3350 RTP IEE 3RFC3351 RTP / AVP IEE 3RFC3398 Integrated Services Digital Network (ISDN) User Part (ISUP) to Session

Initiation Protocol (SIP) MappingIEE 4

RFC3420 Internet Media Type message/sipfrag IEE 3RFC3515 REFER method IEE 3


RFC3711 SRTP = Secure Real-Time Transport Protocol IEE 5 at IPMR

RFC3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

IEE 3

RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Proto-col (SIP)

IEE 3 partial

RFC3968 The Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the Session Initiation Protocol (SIP)

IEE 3 partial

RFC3969 The Internet Assigned Number Authority (IANA) Uniform Resource Identi-fier (URI) Parameter Registry for the Session Initiation Protocol (SIP)

IEE 3 partial

RFC4028 Session Timers in the Session Initiation Protocol (SIP) ab IEE 4RFC4123 Session Initiation Protocol (SIP)-H.323 Interworking Requirements IEE 3RFC4346 The Transport Layer Security (TLS) Protocol 1.1 IEE 5RFC4568 Session Description Protocol (SDP) Security Descriptions for Media

StreamsIEE 5

Diversion - draft-levy-sip-diversion-08 IEE 3Session Initiation Protocol Call Control Transfer: draft-ietf-sipping-cc-transfer-06 IEE 3 partial Session Initiation Protocol Service Examples: draft-ietf-sipping-service-examples-10 IEE 3 partial

The following standards are supported from version

Voice over IP


Voice over IPThe Integral Enterprise offers even IP-based voice communication (Voice over IP) in full functionality. Thefeature table shows how several features are supported by E07 as well as their availability at various IPdevices and softphones.

With IEE 5 voice over IP offers increased scalability, enhanced functionality and improved security bymeans of a new multi-functional IPMR board (= IP Media Resource 256).

Useable features (Start)See Legend [ → 24 ] for identification of the entries, see Continuation [ → 22 ]

VoIP

Support H.323

T3 IP Tunnel

96xx Tunnel

SIP Softphone

R2.1Lock via software lock: Local (n-levels) na 1 1 -

Central (n-levels) - 1 1 xAlpha input via: Numeric keyboard na Classic x -

Alphabetic keypad na Comfort x xHeadset na x x PC-Head-

setSecond receiver, recording na x - -External handsfree equipment na x - -DSS module na x x -Door device na x - -Second alarm na x - -Audio settings via menu, e.g.: Change receiver volume na x x x

Mute (switching off the microphone) na x x xTone ringing, can also be switched offTone for attention cannot be switched off

na x x (x)

Call list No. of entries na 30 30 30 with time x x x

Call counter (max.) 99 99 99 ?and date x x xautomatic deletion can be configured, x x xStatus display "LIST" for new entry x x -no entry for RUL, RWL, Pick up x x -only 1 local call list per user in the terminal x x -


Cut-in on analog and digital subscribers x x x -Switchover to CC (Call Center) mode: - x2 x2 - Login and logout,

Change password, Display ’logged on’, Display user subscriber no. /Name, Alert tone in the HSG, Order code input for outgoing CC calls, Order code input, Selection of agent groups, Auto answer by user/reviser, Auto release, Automatic status enquiry, Logging in/out, Emergency call button (CC agent can request help from a supervisor at the push of a button, either by listening in or by cutting in), No CC calls in calls list, Manual post-editing time for CC can be switched off, Post-editing time automatic/manual/cancel, Pause function, Status display for order code, Queue display

User prompting Cursor control na x x -

Block dialing in refer-back with partner button (without possibility to correct)

na x x -

Block dialing in refer-back with destination key (without possibility to correct)

na x x -

Code dialing in the system - x x xCSTA functions: - x x - Make Call, Clear Connection, Divert Call, Invoke

Features Alternate Call, Answer Call, Conference Call, Consultation Call, Reconnect Call, Transfer Call, Call Completion (Call Back or Intrusion), I/O Services for ZETB

Date, time - x x - Manual entry of date, time and time server IP

address,Load time/date from the time server

- - - -

VoIP

Support H.323

T3 IP Tunnel

96xx Tunnel

SIP Softphone

R2.1

Voice over IP


Remarks:1. with Feature Access Code2. not with T3 IP Compact or with 9620

See Legend [ → 24 ] for identification of the entriessee Continuation [ → 22 ]

Display Languages (number) na 12 12 -

Display own subscriber number / name,only one user per set at any one time

na Num-ber/ name

Num-ber/ name

-

Status display of active features na x x -Announcement for Comfort/Classic - x x - Receive announcement - x x -Call tracing of calls - x x -Follow me (Dialing procedure) - x x (x)1

Handsfree na x x - Automatic connection of the FRE na x x -Function keys, freely programmable na x x -Call charge metering in the display Unit/Charge-individual/total na x x

Bilingual in euros na x x Delete all entries na x x (x)1

Picking up a call - x x (x)1

Journal consisting of: na x x Call list (no. of entries) na 30 30 x

Call list (number of calls) na 10 10 xRedial (number of attempts) na 10 10 x

Conference for three subscribers - x x xMulti conference up to eight subscribers - x x -Customer data backup on background memory - x x -Monitoring na x x -

VoIP

Support H.323

T3 IP Tunnel

96xx Tunnel

SIP Softphone

R2.1


Useable features (Continuation)See Legend [ → 24 ] for identification of the entries See Starting [ → 19 ]

VoIP

Support H.323

T3 IP Tunnel

96xxTunnel

SIP Softphone

R2.1

Call toggling x x x xMacros: Call diversion, pick-up na x x - Partner call diversion na x x -

Stand-in switching na x x -Message Waiting function (request with key) - x x -Minimail, Mailbox

- xx

--

-

Emergency call function local (no. of destinations) x 10 10 -Partner group closed - x x -Partner group individual x x Partner display, maximal number - 16 16 -

Partner pick up, also in the call status - x x -Partner double call - x x -Partner initialization - x x -Partner call signal to busy secretary - x x -VIP function, send special call signal - x x -Partner call diversion, can be switched on during call

- x x -

Partner diversion for group in the partner menu - x x -Display of partner call diversion - x x -Handsfree answer back with flashing LED - x x -Boss key - x x -

Suppress PIN in the display and redialing memory na x x xIPUM (IP User Mobility) Transfer of authorizations (see note 2)

- x x x

Relocate - x x -Remote Maintenance - x x - Software Download

Saving customer dataTerminal reset

Voice over IP


Refer-back x x x x Refer-back Block dialing for partner and destina-

tion keys- x x -

H.450.4 Call Hold/Retrieve x na na xCallback on busy - x x -Call diversion Call diversion explicitly x x x (x)1

Call Forward Busy x x x (x)1

Call forwarding (x)1

With self-input (internal/external) - x x -Changing the call diversion remotely - x x -Double Call With self-input - x x (x)1

Do not disturb na x x xHunt group line, incl. Withdraw - x x -Silent Monitoring and Coaching - x x -Barring features na x x -Key beep na x x -Telephone book local (entries) na max.300

min.171max.300 min.171

x

It is possible to enter a name when storing while calling.

x x x

Field text completion 12 spaces x x xDisplay of the level of assignment x x -

Central phone book - x x xTime setting (local for 10 appointments) na x x - With preselection of a number x x -Test mode, Display and keyboard na x x -Disconnect Hanging up without receiver na x x xDoor opening function (system solution) - x x -Transfer, Connection transfer x x x - H.450.2 Call Transfer x na na (x)1

Converting incoming external numbers In name, if saved in the TEBU

na x x x

VoIP

Support H.323

T3 IP Tunnel

96xxTunnel

SIP Softphone

R2.1


Remarks:1. with Feature Access Code2. IPUM (IP User Mobility)

Prior to IEE 3, a subscriber number was either treated as an IP or ISDN subscriber. Accordingly, as hybrid operation was not permitted, a distinction was made between

• PUM = Personal User Mobility between ISDN telephones,• IP PUM = IP User Mobility between IP phones.

From IEE 3 onward this distinction is no longer required. Overall PUM allows users to switch and log in at ISDN telephones and IP phones in turn.

See Legend [ → 24 ] for identification of the entries See Starting [ → 19 ]

Legend

Suppression of numbers and name display no entry in redialing memory local x x x -

remote x x x -Dialing with receiver on-hook na x x after pressing destination key na x x xRedial (batch mode) na 10-times 10-times Redial after time na 9x 9x -

Redial regardless of connection status na x x -Switching between calls x x x xTransferring two-party conversation x x x -World time display na - x -Destination dialing na x x xSecond call passive x x x xSecond call, can also be switched off if second call on/off on F key (menu can only be activated in idle)

- x x -

Second call can generally be switched off na x x -

VoIP

Support H.323

T3 IP Tunnel

96xxTunnel

SIP Softphone

R2.1

x Realized/supported featuresNumber Meaning clear from context- Unsupported featuresNo Unsupported featuresn/a Non-relevant features

Voice over IP


EnhancementsSince the introduction of voice communication over IP (VoIP) with E070V02 it was continually improved:

The SW-Release IEE 5 provides increased scalability and advanced functionality by means of a newboard "IP Media Resource 256" (= IPMR) for the Integral Enterprise. It replaces the VoIP board and takesover its function.

• up to 192 channels• up to 750 IP subscribers at one board• Encryption for SIP clients with TLS (= Transport Layer Security)• Encryption for SIP clients to IPMR on SRTP (= Secure Real Time Protocol)• redundant Ethernet interface• distributed media streaming with up to 750 IP subscribers

For the HW requirements for IEE 5 and for the description of its configuration please consult the handlingspecifications:

• HSP_1783: IP Media Resource 256 (IPMR) (from IEE 5, 10/2008)

The SW release E070V04 expands Integral Enterprise in direction towards VoIP integration, but all of thefunctions of the SW release E070V02 can still be used. In addition, improved features are available for T3IP telephones, which are equipped with SW for tunneling (B-packet) and with HW for VLAN support:

• Higher scalability in the number of IP subscribers and IP lines,• Better representation of branch office concepts through decentralised media streams,• Redundancy of the VoIP module,• Interaction between the new T3 IP phones with VLAN support and B-packet software,• Log-in procedure with display of potential reasons for rejection on the display of the T3IP phone,• IP QSIG improvements in terms of quality of service,• Improved servicibility through upgraded service applications: ICU Editor and ISM, plus VoIPSW shell

access.• The features Call diversion for IP users and Voice recording have been improved.

For the HW requirements for the software release E070V04 and for the description of its configuration ple-ase consult the handling specifications:

• HSP_3704: VoIP in Integral Enterprise, MG100 oder I33 (from E070V04, 02/2004)

For the HW requirements for the software release E070V02 and for the description of its configuration ple-ase consult the handling specifications:

• HSP_3703: VoIP in Integral Enterprise, MG100 oder I33 (from E070V02, 07/2003)


Expansion limitsWith IEE 5 the scalability is substantially increased with up to 750 IP subscribers and 192 hybrid channelsby means of the new IPMR board, compared to the previous VoIP board.

Modes and CompatibilityThe IPMR board can be used in two different modes:

• The VoIP modeis used, when a VoIP board with existing IP participants is replaced by a new IPMR board with the upgrad on software IEE 5 or higher. The same (one ot two) slots may be used further, however only 32 or 64 hybrid channels are useable.

• The IPMR modeis used, if more hybrid channels should be available, moreover up to 750 participants are possible. Then the highways of the racks R3 and/or R4 are used (at duplicated IPMR for redundancy). They don’t have to be physically there, therefore this case is also possible in a C1 module!

Scalability in SW release until IEE 4 inclusive

from IEE 5

with board with VoIP with IPMR inVoIP mode

with IPMR inIPMR mode

Hybrid channels per board max. 64 max. 64 max. 192IP users per board max. 240 max. 240 max. 750Boards per module max. 16 max. 16 max. 3IP users per module max. 960 max. 1000 max. 1000Users per module (Sum of IP users and ISDN users and lines)

max. 960 max. 1000 max. 1000

Comparison of the two boards: VoIP IPMRredundant Ethernet interface no yesEncryption for IP phones T3 and 96xx (AES-128) yes yesEncryption for SIP phones (SRTP) no yesEncryption for IP QSIG yes (from IEE 5) yesmax. number of hybrid channels for networking 64 192max. number of IP users per board 240 750max. number of IP users per modul 1000 (from IEE 5) 1000

Voice over IP


Compatibility of the two boards VoIP and IPMR with different systems and software releases:

RedundancyThe reliability of IP telephony is increased by a redundant Ethernet interface with IEE 5 (= L2 Failover).The IPMR board has two Ethernet interfaces as well as two corresponding jacks Ethernet1 and Ethernet2on the front bar.

In the case of failure of the primary interface it is switched on the secondary redundant interface. This pro-cedure ensures that the IP telephony is not disturbed at failure of Switch1. Once Switch1 is operationalagain, it returns back to the primary interface.

In addition, the active IP participants may get redundant counterparts on a second standby IPMR board,who become active, if the first IPMR board fails.

That doesn’t have any impact on the well-known redundancy for the boards VoIP and CF.

IPMR protocolsThe IPMR board supports the following protocols for connecting participants and for networking:

• TN1R6-Tunnel• IP-QSIG Integral Enterprise and IP-QSIG Communication Manager• SIP• SIP-Trunk• H.323

System Integral 33 / MG100 Integral 55 / MG1000Software till IEE 4 incl. from IEE 5 till IEE 4 incl from IEE 5

VoIP yes yes yes yesIPMR in VoIP mode no yes no yesIPMR in IPMR mode no no no yes


Different types of devices can be connected to an Integral Enterprise with IPMR. The IPMR board suppliesthem with IP-based voice services.

• TN1R6T3 IP phones use the protocol TN1R6 for signaling. The signaling information is sent via an IP tunnel. Therefore the phones T3 IP and T3 realize exactly the same features.IP phones of the 9600-series with TN1R6 software provide the same features as T3 IP phones.

• H.323Standard H.323-compliant devices from other manufacturers (for example Innovaphone IP200) can also be connected. In this case the Integral Enterprise supports some key features on basis of H.450.x

• SIPThe IPMR board also supports the SIP protocol (= Session Initiation Protocol). Thus several SIP cli-ents from Avaya and several 3rd-party phones are connected, all, hardware phones as well as soft-phones or WiFi handsets. For further details, please refer to the handling specification

• HSP_1754: Integral Enterprise SIP.

Networking of systems is possible by IP protocols as follows:• IP-QSIG

The protocol H.323 via QSIG is supported for IP networking between systems Integral Enterprise and Communication Manager. Encryption is also possible between systems Integral Enterprise. For further details, please refer to the handling specification

• HSP_1747: Connectivity of CM and IE using QSIG over IP.

• SIP-TrunkSystems are networked to each other by using the SIP protocol at the SIP trunk interface, for example:

• Server based Windows applicatios, connected via SIP CAPI driver,• or SIP service provider.

For further details, please refer to the handling specification • HSP_1754: Integral Enterprise SIP.

Requirements

• VoIP-Readiness:The data network of the customer (LAN, WAN, Corporate Network) must offer the necessary perfor-mance or it has to be implemented. The Quality of Service is important, certain maximum values for delay, packet loss and jitter must be met, specially for Fax over IP with the protocoll T.38.

• Software:The communications software IEE 5 (Integral Enterprise Edition 5) includes enhancements for com-missioning and support of the IPMR functions. The applications CAT, ICU-Editor, ISM etc. from versi-ons 15.xxx onward support the IPMR board, they are provided with IEE 5.

Voice over IP


• Required space: • In the VoIP mode, one slot and possibly the neighboring slot is needed for an expansion with a

maximum of 64 simultaneous hybrid channels.• In the IPMR mode, one single slot is sufficient for an expansion to 192 simultaneous voice chan-

nels (exclusively in Integral Enterprise from IEE 5 onward). In that case the appropriate virtually used racks R3 and R4 are no longer usable.

• Maximum expansion: • 192 hybrid channels and 750 participants at Integral Enterprise,• 64 hybrid channels and 240 participants at Integral 33 or at MG100.

VoIP-Manager

The VoIP Manager is used for configuration and management of IP terminals. The data needed by the ter-minals may be configured efficiently with the help of the VoIP Manager, especially while installing IP termi-nals in large customer networks.

For that purpose the VoIP Manager produces a structure and allocates the IP terminals of the customer’snetwork. This structure has three levels and consists of several PBXs (eg I55 or Integral Enterprise) andgatekeepers configured within the PBXs (IPMR boards or VoIP boards) and various user groups within thegatekeepers.

For example, a user group may contain all the terminals of a building or a floor or all the terminals of a cer-tain type (eg T3-IP, 96xx).


The parameters of the IP terminals are divided into classes and can be configured in the different levels(eg user group, terminal). For example, all the IP terminals of a user group can be configured with thesame parameters.

The VoIP Manager is a client/server application and supports terminals of the series T3-IP and 96xx.

New features from version 2.0 (with IEE 5):• The VoIP Manager supports the IPMR board with up to 750 participants.• The VoIP Manager sets up secure connections to IP devices over TLS (= Transport Layer Security).• The user interface of the VoIP Manager shows for each device, whether TLS is used.• The defaults of the according user group are used for a new entry.• Columns of a user group can be sorted and edited. The settings are preserved and don’t fall back to

default.• An administrator may change the network name of the IP phone.

Security

•As of IEE 3, exclusively secure protocols are used for PBX administration, such as SSL and IPsec.

•User identification and registration on T3 IP telephone with hardware level II and software C-package level 2:

• proprietary based on H.235 v3 (Annex H),• fixed delay on unsuccessful identification,

i.e: further attempts to log in are ignored for 15 seconds after three failed attempts to log in.• password contains up to 6 digits.

•Signalling between the T3 IP phone and the VoIP board is encrypted using AES 128 (= 128-bit Advanced Encryption Standard).

•Media streams between the T3 IP phone and the VoIP board are encrypted:

• encryption of user data in accordance with AES,• centralised encryption of the media streams,• decentralised encryption of the media streams only between IP phones on the same VoIP board.

•Encryption of the SIP interface from IEE 5:

• The signaling messages are encrypted by TLS (= Transport Layer Security)• and the voice packets for IP participants in the IPMR board will be encrypted by SRTP (= Secure

Realtime Transport Protocol). SRTP is an encrypted version of RTP (= Realtime Transport Proto-col) also using AES to encrypt the voice data.

For further details, please refer to the handling specifications • HSP_1774: Integral Enterprise IP Security and • HSP_1754: Integral Enterprise SIP.

Voice over IP


IP usersInitially, a T3IP Tunnel-Phone has to log in at one of the VoIP boards, which is then the default user of thecorresponding gatekeeper. Two mechanisms are supported for this login process:

1. Gatekeeper Discovery Method:The IP phone searches the LAN for a gatekeeper (by means of a Multicast Gatekeeper Discovery message) and logs in at the gatekeeper which replies first.

2. Default Gatekeeper Method:The address of a "Default Gatekeeper" is allocated to each IP phone for login (possibly also the address of an "Alternative Gatekeeper" as a redundant measure in case of failure of a VOIP module). Here, these addresses are

• either distributed via DHCP (not allowed in all customer networks)• or given a fixed assignment by the VoIP-Manager.

The gatekeeper discovery method can only be used if the IP telephones and the associated VoIP board(and no other VoIP board) are in the same LAN or VLAN. However, customer LANs usually have a diffe-rent structure, and in these cases the second method needs to be used.

As the gatekeeper address is assigned to the default user, a T3IP Tunnel Phone is always attached to thesame gatekeeper (VoIP module), regardless of the IP users logged in there. Following registration of thisdefault user, real IP users can log on to the T3IP Tunnel Phone.

In combination with the system SW E070V04 and the terminal SW for the T3IP B-packet, the two mostimportant reasons for rejecting a login are shown on the display of the T3IP:

• "Wrong PIN" when an incorrect PIN is entered and

• "Call number engaged", if the relevant subscriber is currently mid-call or being called.

Media StreamsE070V04 supports the decentral speech data transfer ’Media Streaming’ between two T3IP Tunnel Phoneswhich are registered on the same VoIP board (with the same Codec).

• If two IP subscribers are logged on at different VoIP modules, then they will communicate via the Inte-gral Enterprise PCM switching units (CF2 and B3 modules).

• However, if the two IP call partners are logged on at the same VoIP modules, then the speech pak-kets are exchanged directly in the IP network - as decentral media streams - without participation of the VoIP board and the switching units of Integral Enterprise.

As a result, the load on the VoIP board is reduced and the bandwidth for two connections via the WANroute is saved.


Decentral media streams are only connected for calls between two T3IP Tunnel Phones. During confe-rences or cut-ins and when loading tones, media streams are automatically routed via the VoIP boardagain.

Decentral media streaming can be switched off (in the event of potential problems in a specific customerconfiguration), either for all connections in the VoIP board or individually in the T3IP Tunnel Phone.

Example - branch office scenario:In the branch office scenario, T3IP Tunnel Phones are connected via a WAN route to the VoIP. If the samegatekeeper is allocated to all of the IP phones in a branch office, then all calls between IP users within thebranch office will remain as decentral media streams in the LAN of the branch office. This remains the caseeven if the IP users logged in within the branch office change.

With IEE 5, decentralized media streaming is extended up to 750 IP participants by means of the newIPMR board compared with the previous VoIP board.

IP user mobilityThe customer data of an IP user is assigned to a hardware address, but can be moved automatically if theIP user moves to another hardware address:

1. IPUM (IP User Mobility):An IP user moves from one T3IP tunnel phone to another T3IP tunnel phone without logging off from the first phone. With E070V04, IPUM functions within a module; with IEE 3 it also functions system-wide.

2. Move:The IP user first logs off from his IP phone. After logging off, he moves to another IP phone. There are no restrictions on the "Move" function in E070V04.

Ex.1: An IP phone changing to an alternative gatekeeper after the failure of the default gatekeeper corresponds to a move by the IP user, since all IP users logged on to this board will be automatically logged off after failure of a VoIP board.

Ex.2: An IP phone logs off from its current gatekeeper and logs on to a new gatekeeper. This also cor-responds to a move by the IP user, since the IP user logged on to the IP phone will be logged off before the IP phone is logged off.

This option makes configuring a system much easier. As a result, the process of distributing IP phones iscarried out independently of the customer data generation process! Accordingly, IP phones can be assi-gned to various VoIP boards according to their spatial distribution without acknowledgement by futureusers. The users can then log on to any IP phone.

The redundancy procedure is controlled by the IP phones. If the IP phones detect that they have lost theconnection to their gatekeeper, they log on to the alternative gatekeeper. Afterwards, the customer data ofthe online subscribers is automatically moved from the first module to the next.

Voice recording


Voice recordingVoice recordings are a standard function of Integral Enterprise.

From E070V04 the recording, including that of hybrid calls for IP users, is supported via the CF board: • Calls for one IP user with one ISDN or analog subscriber,• Calls for one IP user with one line and• Calls between two IP users who are logged in to different VoIP boards.

• Voice recording is not possible for ACD agents who are logged on to an IP user. This must then be performed using the subscriber number of the IP user.

• If an IP user migrates, i.e. changes his hardware address, the voice recording must be redefined in the IDM, and restarted. Therefore an IP user with voice recording must not use the User Mobility fea-ture.

From IEE 5 voice recording is possible also on SIP connectivity, see EVOip active SIP [ → 16 ]


List of attachmentsFor detailed descriptions, the following chapters refer to external documents with HSP number and lastrevision date:

HSP external documents date1747 Connectivity of Communication Manager and Integral Enterprise (QSIG over IP) 05/20081754 Integral Enterprise SIP 06/20081760 SIP SES Integration in Integral Enterprise 05/20081771 Communication Manager -Integral Enterprise, connected by TDM-QSIG 03/20061774 Integral Enterprise IP Security 05/20081776 VMM Support in Integral Enterprise 05/20081782 CSTA Service Single_Step_Conference 01/20081783 IP Media Resource 256 (IPMR)(ab IEE 5) 10/200837033704

VoIP in Integral Enterprise, MG100 oder I33(ab E070V02)Neue LMs E070V04 / IPV3 (ab E070V04)

04/200301/2006

- HSP zu ACB/ACB1/S5500, APS IEE 5.0 08/2008

Index


Index

CCall parking 10Centralized Call Distribution Service 10Clock synchronization 10Comfort Buttons 15Common information 10Conference 10

DDouble call 10DSS module 10

EEVOip active SIP 16

IInstant Messaging 15IPMR 15

MMessage Waiting 10Minimail 10

NNetwork Management Console 7NMC 7

PPresence 15

QQoS 7QSIG+ 10Quality of Service 7

RRealTime Control Protocol 7recording 16RTCP 7

SSES 15SIP Enablement Server 15SNMP 7SNMP Simple Network Management Protocol

7SW E070V02 19, 25SW E070V04 19, 25, 26, 31, 33SW IEE 3 7, 10, 12, 19, 30SW IEE 4 7, 12, 26SW IEE 5 7, 10, 13, 15, 16, 19, 25, 26, 27, 28, 29, 30,

31, 33Synchronization 10

TTNet 10

VVMM 7Voice over IP 19, 33Voice quality 7Voice recording 16VoIP 15VoIP Monitoring Manager 7


Last Page

fd en extract iee5

Documents

uninterruptible

mg100 oder

central charing

central operator

feature access

handling specification

handling specification

session initiation