Narrativas cibernéticas y arquitectura computacional

IP devices using Inter-Tel Protocol (ITP) and Session Initiation Protocol (SIP) modes allow users to communicate using the Local Area Network (LAN) and the telephone system. The telephone system supports the following IP and SIP devices:

• Hard IP Endpoints

— Models 8664/8665/8668 (wireless)

— Model 8660

— Axxess IP PhonePlus — Eclipse IP PhonePlus

— IP Single-Line Adapter (IP SLA)

• Hard Multi-Protocol Endpoints (Operate in ITP or SIP mode) — Model 8600 — Model 8620 — Model 8622 — Model 8662 — Model 8690 • Soft IP Endpoints — Axxess IP SoftPhone — Eclipse IP SoftPhone — Model 8602 IP Softphone

• Soft SIP Endpoints: Model 8601 SoftPhone for Pocket PC (operates in SIP mode only) As of V8.1, IP and SIP calls can be routed through the system cabinet or they can be config- ured as members of the same Network Group. If the devices are routed through the system cab- inet, they terminate at the 32-Device Internet Protocol Resource Card (IPRC) (see page 76). If the devices are configured to be members of a Network Group, they can use the peer-to-peer (P2P) audio feature (see page 104).

Page 101

IP and SIP Endpoint Identification

IP and SIP Endpoint Identification

At first glance, IP and SIP hard endpoints look nearly identical to digital endpoints. Additional ports on the back of the endpoint, used for LAN and power connections, identify an endpoint as an IP or SIP endpoint. IP and SIP endpoints operate like digital endpoints installed on the system except for the limitations listed on page 101.

Installation and Configuration

To install and configure IP and SIP devices, you must know the required network settings and be familiar with the associated hardware and software. For installation and configuration infor- mation for all IP and SIP devices, refer to the IP Devices Installation Manual (part number 835.2195).

Automated Boot Code Update

Information about the automatic update of endpoint boot code is included in this guide for the following reasons:

• You may be unaware that the update is occurring

• Interruption of power to the endpoint during the download causes the endpoint to be inoperable and unrecoverable

A Model 8620, 8622, or 8662 endpoint with v2.0.0 or later firmware uses a Trivial File Trans- fer Protocol (TFTP) server to update a crucial internal boot code. In any of the following situa- tions, the endpoint checks its configuration file against a corresponding file on the TFTP server:

• On startup

• On command

• On periodic timeout

If the endpoint detects new parameters during this check, it automatically initiates the boot update procedure to download new boot code. However, the download is delayed if a call is in progress or music on hold is enabled. The endpoint display changes to alert you that the update is in progress.

IP Limitations

Currently, IP devices (Session Initiation Protocol and Inter-Tel Protocol) have the following limitations:

• They do not have a secondary voice path and cannot support off-hook voice announce (OHVA).

• They do not support the enhanced speakerphone mode (feature code 310). • They cannot use a PCDPM or MDPM.

• They do not support the DSS/BLF units.

CAUTION

If, during the boot code download, electrical current to the endpoint is interrupted (e.g., through disconnection or a power outage), the endpoint becomes inoperable and cannot be recovered. In that event, the endpoint must be returned to Inter-Tel for repair.

Page 102 Installation and Configuration

• They do not support Desktop Open Architecture Interface (OAI) applications.

• While using peer-to-peer (P2P) audio, SIP and IP devices do not support the Agent Help, Record-A-Call, and Station Monitor features.

Installation and Configuration

To install and configure IP and SIP devices, you must know the required network settings and be familiar with the associated hardware and software. For installation and configuration infor- mation for all IP and SIP devices, refer to the IP Devices Installation Manual (part number 835.2195).

Axxess and Eclipse IP SoftPhone

The Axxess and Eclipse IP SoftPhones allow you to make and receive telephone calls on a PC. The Axxess IP SoftPhone application uses the Executive Display endpoint graphical user inter- face (GUI) and functions like an Executive Display endpoint installed directly on the system, except for the IP limitations listed above. The Eclipse IP SoftPhone application uses the Pro- fessional Display endpoint GUI and functions like a Professional Display endpoint installed directly on the system, except for the IP limitations listed above. The Axxess or Eclipse IP SoftPhone’s audio is handled by the PC’s microphone and speakers. The Axxess and Eclipse IP SoftPhones terminate on IPRC (firmware v8.0 and earlier) in the system cabinet.

NOTE: The Axxess or Eclipse IP SoftPhone does not support v8.1 and later firmware or P2P audio.

Model 8602 IP Softphone

The Model 8602 is a new IP softphone that is used with system versions 9.1 and later. The Model 8602 is a softphone application that enables Voice over IP (VoIP) telephone calls from laptops or desktop computers. The Model 8602 connects to the Inter-Tel® telephone system through an existing IP network. The Model 8602 operates like a Model 8662 endpoint and sup- ports Inter-Tel Protocol (ITP) mode. Once a connection is established, Model 8602 users can converse with another party (or parties) via a headset connected to their computer. The Model 8602 IP softphone requires software v9.1 or later and IPRC firmware v8.2.x or later.

Model 8690

The Model 8690 is an advanced IP endpoint that has an LCD touch screen that displays a tele- phone-type interface. A stylus pen provides access to elements on the interface, which include a dialpad, feature buttons, menu buttons and navigation buttons. The Model 8690 can operate in Session Initiation Protocol (SIP) mode or Inter-Tel Protocol (ITP) mode (see page 104).This endpoint uses a customized version of Microsoft® Windows CE.NET v4.2, and includes ports and connectors for optional hardware and memory cards.

For more information about this unique IP endpoint, consult the appropriate documentation listed below:

• For supported features and end-user instructions: Model 8690 User Guide–Inter-Tel

Protocol Mode (part number 550.8116) or Model 8690 User Guide–Session Initiation Protocol Mode (part number 550.8025)

• For general administrator information: Model 8690 Administrator Guide (part num-

ber 550.8120)

• For installation and maintenance information: IP Devices Installation Manual (part

Page 103

MGCP Gateway and Endpoints

MGCP Gateway and Endpoints

The system also supports IP-based solutions for providing local Public Switched Telephone Network (PSTN) connectivity using the IPRC and a third-party Media Gateway Control Proto- col (MGCP) device. For information about currently supported MGCP devices, contact your local authorized Inter-Tel dealer.

SIP Gateway

Axxess supports SIP (Session Initiated Protocol) trunks communication with the CO via SIP- enabled gateways. As the SIP protocol becomes more popular, it is important to be able to communicate to SIP gateways in the IP-centric world. Inter-Tel currently supports the follow- ing SIP gateways:

• Quintum® AFT 400 SIP gateway • AudioCodes™ MP-114 SIP gateway

The above-mentioned gateways replace the AudioCodes MP-104 SIP Gateway that has gone end-of-life in October 2006. Existing Axxess systems must upgrade to v10.0 or later to support these gateways.

SIP trunks support the following functionality:

• They are transparent to the system user because SIP trunks work like any other CO trunk in the system.

• They support transferring trunks, putting trunks on hold, and connecting trunks to con- ferences, similar to other CO trunks in the system.

• They support making and receiving calls by any endpoint. • They support peer-to-peer audio by IP endpoints.

• They reside in CO trunk groups just like other trunks so that SIP trunk calls can be routed using Automatic Route Selection (ARS).

• They support 911 calls (like MGCP trunks). SIP trunks require:

• An IPRC with v8.1.x or later firmware.

• A SIP trunk software license (part number 840.0267).

NOTE: SIP trunks do not support the Direct Inward System Access (DISA) or speed dial fea- tures.

CAUTION

If an IP or SIP device user dials 911/999 from a remote location and an MGCP gateway is not present and configured properly, the call will be placed from where the telephone system is located. Because 911/999 services use Caller ID to help locate the caller, emergency ser- vice could be misdirected or delayed. Also, the 911/999 service contacted may be local to the system, but not to the IP or SIP device. All IP and SIP device users should be alerted to this situation and instructed to use a local telephone line for placing emergency calls if an MGCP gateway is not present and/or functioning properly. Also note that IP and SIP devices will not function in the event of a power failure at either the local site or the system location. Inter-Tel will not be held responsible for any problems resulting from an unavailable line that is connected to the MGCP gateway. Inter-Tel also recommends that users regularly test the MGCP gateway for dial tone.

Page 104 Peer-to-Peer Audio for IP and SIP Devices Peer-to-Peer Audio for IP and SIP Devices

The P2P audio feature, available in software v8.1 and later, allows certain IP and SIP devices to transmit and receive audio directly with each other rather than through the system cabinet. The P2P feature reduces delay and packet loss for IP and SIP devices.

Devices that support P2P audio include:

• AudioCodes MP-114 or Quintum AFT 400 SIP gateway • IP endpoints, except the IP SoftPhone

• Multi-Protocol endpoints • SIP endpoints

To use P2P, the IPRC and the IP and/or SIP devices must be upgraded with v8.1 firmware. In addition, the devices must be programmed as members of the same Network Group. Calls between IP and SIP devices that are members of separate Network Groups are routed through the cabinet and do not use P2P.

Version 8.1 and later software supports a Network Group Diagnostics feature that allows an administrator to verify that the Network Groups are programmed properly. See page 49 for instructions on how to use the Network Group Diagnostics feature.

IP devices that use P2P audio do not use cabinet resources when they are connected in a P2P call. Therefore, the following features will not work during a P2P call:

• Agent Help

• Record-A-Call • Station Monitor

Inter-Tel Protocol (ITP) and SIP Operating Modes

The Models 8600, 8620, 8622, 8662, and 8690 Multi-Protocol Endpoints have the ability to operate in ITP (formerly known as Axxess IP) mode or Session Initiation Protocol (SIP) mode. The Models 8664, 8665, and 8668 wireless IP endpoints operate in ITP mode only. The Model 8601 endpoint operates in SIP mode only.

ITP mode supports nearly all of the system features except for the limitations listed on page 101. SIP mode offers basic endpoint features and the flexibility of shared extensions, which means you can use one extension with up to five different devices on the telephone and voice processing systems.

NOTE: Endpoints may have certain hardware and software restrictions that prevent it from fully supporting a feature. Consult the appropriate endpoint user guide to determine which fea- tures are supported for your device.

NOTICE

Passing real-time streaming data, such as audio, through encrypted virtual private networks (VPNs) may significantly impact the network performance, router and firewall functionality, and audio quality.

Page 105

ITP and SIP Supported Features

ITP and SIP Supported Features

Users can access most features by entering a feature code using the endpoint’s dialpad buttons. The following table shows the station features (and default feature codes) that are supported for multi-protocol endpoints operating in ITP mode and SIP mode.

Support for some features varies by device. For a complete list of supported features refer to the appropriate user guide. See Table 36 on page page 321 for a table of user guides and part numbers.