Communication Server 1000E Release 5.5

Communication Server (CS) 1000 Release 5.5 requires the IP Line application.

The IP Line application provides an interface that connects IP Phones to a CS 1000 Call Server. CS 1000 Release 5.5 requires a Signaling Server to operate. You must upgrade your Meridian 1 system, if your system is IP enabled to include a Signaling Server, which in turn becomes a CS 1000M system. CS 1000 Release 5.5 is supported on an analog/digital (TDM) only system without a Signaling Server if the system is not IP enabled. For information about upgrading your system, see Communication Server 1000M and Meridian 1 Small System Planning and Engineering (NN43011-220) or Communication Server 1000M and Meridian 1 Large System Planning and Engineering (NN43021-220).

ATTENTION
The IP Line version of software must match the Call Server version.

Digital Signaling Processor daughterboards Digital Signaling Processor (DSP) daughterboards on the Media Gateway
Controller (MGC) provide DSP resources for connecting IP and Time Divisional Multiplexing (TDM) devices together in a CS 1000E Media Gateway (MG 1000E) system. The following DSP daughterboards are available:
• 32 port daughterboard
• 96 port daughterboard

These daughterboards provide an optional solution to installing the Voice Gateway Media Cards within the CS 1000E Media Gateway (MG 1000E) chassis. However, Voice Gateway Media Cards are still supported within an MG 1000E with a Media Gateway Controller (MGC) and DSP daughterboards. The MGC is only used in a Media Gateway chassis or an Option 11C cabinet. For further information about DSP resources residing on the MGC that are configured with DSP daughterboards, see Communication Server 1000E Installation and Commissioning (NN43041-310).

Voice Gateway Media Cards
If a Media Card 32-port card or a Media Card 32S is running IP Line software, it is known as a Voice Gateway Media Card.
In this document, Media Card 32-port card and Media Card 32S card are referred to as Media Card 32-port cards, unless explicitly stated.

DHCP server
A Dynamic Host Configuration Protocol (DHCP) server can be used to provide the required information so that the IP Phone network connection can connect to the Signaling Server or Line Terminal Proxy Server (LTPS). For more information about DHCP, see Converging the Data Network with VoIP Fundamentals (NN43001-260)and IP Phones Fundamentals (NN43001-368).

Unsupported products
The following remote service products do not support the Media Card 32-port line card, Media Card 32S card, and ITG-Pentium 24-port line card:

• Carrier Remote
• Mini-carrier Remote
• Fiber Remote
• Fiber Remote Multi-IPE

System requirements
CS 1000 Release 5.5 software is the minimum system software for IP Line.

Element Manager and Telephony Manager 3.1
In CS 1000 Release 5.5, Element Manager is the primary interface for Voice Gateway Media Cards and IP Line. Telephony Manager (TM) 3.1 is used only to obtain Operational Measurement (OM) reports. TM 3.1 is the minimum required version.

CS 1000 systems
Element Manager is used as the configuration, administration, and maintenance interface for IP Line on a CS 1000 system.
If you are trying to use TM 3.1 to perform an action available through Element Manager, then TM 3.1 launches Element Manager automatically.

Corporate Directory
TM 3.1 is necessary for creation of the Corporate Directory database.

SNMP and alarms
Element Manager does not provide an SNMP alarm browser. Nortel recommends you use TM 3.1 Alarm Manager when SNMP alarm collection is required.

System configurations
Although IP Line can be used in different system configurations and its use can vary in those configurations, there are two basic system configurations supported in CS 1000 Release 5.5.

CS 1000 systems
CS 1000systems have a Signaling Server in their network configuration. The Signaling Server is a server that provides signaling interfaces to the IP network. The Signaling Server central processor drives the signaling for IP Phones and IP Peer networking.
In IP Line, the LTPS executes on the Signaling Server, and the voice gateway executes on the Voice Gateway Media Cards. All IP Phones register with the Signaling Server. The Voice Gateway Media Cards only provide access to the voice gateway.
The Signaling Server is the node leader and, by default, acts as a Master for the node.

Signaling Servers
The following Signaling Servers are available for CS 1000 Release 5.0:
• ISP1100
• HP-DL320-G4
• IBM-X306m
• Common Processor Pentium Mobile (CP PM)

For further information about Signaling Server hardware platforms, see Signaling Server Installation and Commissioning (NN43001-312). In a CS 1000 system, the Signaling Servers can be used in Leader/Follower and Primary/Alternate/Failsafe combinations. In CS 1000 Release 5.5, the Signaling Servers support the following applications:
• Line Terminal Proxy Server (LTPS)
• Virtual Trunk
• H.323 Gateway
• SIP Gateway
• SIP Redirect Server
• Network Routing Service (NRS)
• Personal Directory
• SIP Proxy Server (SPS)
• Element Manager

Signaling Server redundancy Signaling Server redundancy ensures that telephony services can withstand single hardware and network failures. Several Signaling Servers can load share when the system contains multiple Signaling Servers or Voice Gateway Media Cards. One Signaling Server is a Leader Signaling Server that acts as the primary, or master, Terminal Proxy Server (TPS). The other Signaling Server is a Follower Signaling Server that acts as the backup, or secondary redundant
TPS. There are several methods of redundancy for a Signaling Server.

Methods of Signaling Server redundancy Stage Description With a Signaling Server to share the load
1) A Signaling Server, which shares the load, can be configured in a normal configuration.
2) If the primary Signaling Server fails, the Signaling Server, which shares the load, takes over and all IP Phones register with this Signaling Server.
3) If the Signaling Server, which shares the load, fails, one of the Voice Gateway Media Cards is elected to be the node Master.
4) The IP Phones then register to the Voice Gateway Media Cards.

Without a Signaling Server to share the load
1 If there is no Signaling Server to share the load, and the primary Signaling Server fails, one of the Voice Gateway Media Cards is elected to be the node Master.
2 The IP Phones then register to the Voice Gateway Media Cards.

Software delivery
IP Line supports software delivery through the following formats:
1. Compact Flash
2. Signaling Server CD-ROM
You must download the file of Product Category: VoIP & Multimedia Communications, Product Name: IP-Enabled & Pure IP Networks, Product Name: Signaling Server and IP Peer Networking, Content type: Release.

ATTENTION
To configure IP Line in groups five to seven on Option 81C CP PII or CS 1000M MG, the Fiber Network (FIBN) software package 365 is required.

Fax/Modem pass through
The Fax/Modem pass through feature provides a modem pass through allowed (MPTA) class of service (CLS) for an analog phone TN. MPTA CLS dedicates an analog phone TN to a modem or a Fax machine terminal. A connection that initiates from the dedicated TN, and/or calls that terminate at the dedicated TN through a Digital Signal Processor (DSP), use a G711 NO VAD codec on the Call Server. Modem Pass through is a specific configuration of a G.711 VoIP channel that improves modem performance compared to standard VoIP configuration. Auto switch to Voice Band Data (VBD) is a feature of the DSP; the DSP monitors the data stream to distinguish between voice and data calls. The DSP reconfigures to modem pass through mode when it determines the call is a modem call. The DSP mode on the Mindspeed DSP (MC32S/DB96/DB32) for MPTA
to MPTA fax/modem calls displays ModemPT in the dspMode field of the vgw Show command. The Teology DSP (ITGSA) displays Pass Through. For modem calls between CS 1000 systems connected by analog and digital trunks, you must configure MPTA CLS on the Call Server of each CS 1000 system for analog units connected to modems. MPTA CLS configuration is necessary because the call setup negotiation is not done end to end as it is for virtual trunks. If the analog unit on one Call Server
uses MPTA CLS and the analog unit on the other Call Server uses modem pass through denied (MPTD) CLS, the modem call fails.

When MPTA CLS is configured on a TN, the T.38 protocol is no longer supported for that particular TN. Any call setup with an analog phone TN that has MPTA configured must use G711 Codec exclusively, as this is the only codec available for making calls using this TN. G711 codec is present by default in the DSP configuration. To ensure proper functioning of the MPTA CLS, the Enable Modem/Fax pass through mode check box must be selected in the Gateways section of Element Manager. This check box is selected by default in Element Manager. To enable SG3 fax calls over Teology DSP, the Enable V.21 FAX tone detection check box in Element Manager must be selected. For SMC and MC32S cards, this setting is available in the Voice Gateway and IP phone codec profile section in the Nodes summary page. In the modem pass through mode of operation, the DSP state is displayed as MPT on the MGC and MC32S cards and as PassThru on SMC cards.

MPTA CLS is not supported through Telephony Manager (TM) and Basic Client Configuration (BCC). BCM 50 only supports modem pass through over G.711 in Release M50R3. MPT CLS is supported by the G.711 codec only; MPT CLS includes no
other codecs. The packet interval for G.711 codec is set to 20 msecs in MPT. The maximum speed supported for modem and fax is 33.6 Kbps. This limit is imposed by the analog line card.

MPT allows CS 1000 to support the following:
• modem pass through
• Super G3(SG3) fax at V.34(33.6Kbps)
• V.34 rate (33.6 Kbps) modems
• Fax machines that support V.17, V.27, V.29, and V.34

Note: MPT CLS is not be supported on IP trunks.

When the TN on the CS 1000 is configured with MPTA CLS, it supports V.17, V27 and V29 Fax calls. However, the DSP mode is “FaxBegin,” not“ Modem PT”. The MPTA CLS forces calls originated or terminated on the TN to use the G.711 NO VAD Codec. This codec selection supersedes the existing bandwidth management strategy on the CS 1000. When a CS 1000E is inter-connected with a system from a different vendor, the modem pass through feature works if the third party system
supports the modem pass through mode of operation. The Voice Gateway application displays different Auto-switch states
(ModemPT, Passthru) in the dspMode field of the vgw Show command, based on tones detected by the DSP. These tones are generated by modem and fax machines connected in the TDM domain. The Voice Gateway application does not control Auto-switch states during fax and modem calls, and the dspMode reports tone indications from DSP. The Mindspeed DSP sends tone detection events to the host processor and changes to Modem Passthough and Pass-through auto-switch states (with
or without Redundancy), based on the tones detected, as it is configured in Auto-switch mode.

Modem traffic
CS 1000E supports modem traffic in a campus-distributed network with the following characteristics:
• Media card configuration:
— G.711 codec
— 20 msec packet size
• one-way delay less than 5 msec
• low packet loss
• V.34 rate (33.6 Kbps)

Performance degrades significantly with packet loss (must be less than 0.5%) and when the delay (round trip) is greater than 50 msec and mean jitter is greater than 5msec.

ATTENTION
Nortel has conducted extensive but not exhaustive tests of modem-to-modem calls, data transfers, and file transfers between a CS 1000E and MG 1000E, using Virtual Trunks and PRI tandem trunks. While all tests have been successful, Nortel cannot guarantee that all modem brands will operate properly over all G.711 Voice over IP (VoIP) networks. Before deploying modems, test the modem brand within the network to verify reliable operation. Contact your system supplier or your Nortel representative for more information.

Voice Gateway Media Cards
Voice Gateway Media Card is a term used to encompass the Media Card 32-port card and the Media Card 32S card. These cards plug into an Intelligent Peripheral Equipment (IPE) shelf in the CS 1000M systems and into a Media Gateway 1000E and Media Gateway 1000E Expander in the CS 1000E system.
The Media Card 32-port and the Media Card 32S card occupy one slot.
The Media Card 32S card provides the following features:
• Secure Real-time Transport Protocol (SRTP)
• two Digital Signal Processors (DSP), based on an ARM processor
• channel density of 32 ports
• cost improvement over existing Media Cards

The Media Card 32-port card provides the following features:
• increases the channel density from 24 to 32 ports (for the 32-port version)
• reduces the slot count from a dual IPE slot to a single IPE slot
• supports up to 128 IP Phones in failover scenarios

Card comparison provides a comparison of the Media Card 32-port card
and the MC 32S card.

Capacity
The Virtual TN (VTN) feature allows each Voice Gateway Media Card to support more IP Phones than there are physical bearer channels. There are 24 bearer channels on each ITG-P card and 32 channels on each Media Card. Both cards support a 4:1 concentration of registered IP Phones (IP Phone 2001, IP Phone 2002, IP Phone 2004, IP Phone 2007, IP Audio Conference Phone 2033, IP Softphone 2050, Mobile Voice Client [MVC] 2050, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, IP Phone 1210, IP Phone 1220, IP Phone 1230, WLAN Handset 2210, WLAN Handset 2211, WLAN Handset 2212, WLAN Handset 6120, WLAN Handset 6140 to gateway channels. The ITG-P supports 96 registered IP Phones. The Media Cards supports 128 registered IP Phones (when the card has 32 channels). The IP Phones require the services of the bearer channels only when they are busy on a call that requires a Time Division Multiplexing (TDM) circuit such as an IP
Phone-to-digital telephone/trunk/voice mail/conference. When an IP Phone is idle or there is an IP-to-IP call, no gateway channel is required. When the total number of IP Phones that are registered or are attempting to register reaches the limit (96 on the ITG-P or 128 on the Media Cards), the Voice Gateway Media Card recognizes this, and no more IP Phones are assigned to the card. Each Voice Gateway Media Card is restricted to a total of 1200 call attempts per hour distributed across all the IP Phones
associated with the card.

The components on the faceplate of the Media Card 32-port card are described in the following sections:
Reset button
Use the Reset button on the faceplate to manually reset the Media Card. This enables the card to be reset without cycling power to it. The Reset button is used to reboot the card after a software upgrade or to clear a fault condition.

Enable LED
The faceplate red LED indicates the following:
• the enabled and disabled status of the card
• the self-testing result during power up or card insertion into an operational system

PC Card slot
This slot accepts the Type I or Type II standard PC Flash Cards, including ATA Flash cards (3 MB to 170 MB). The slot is labeled /A:
Nortel supplies PC Card adaptors that enable CompactFlash cards to be used in the slot.

MAC address label
The MAC address label on the card faceplate is labeled ETHERNET ADDRESS. It shows the TLAN and ELAN network interface MAC addresses. The Management/ELAN network interface MAC address for each card is assigned during manufacturing and is unchangeable. The MAC address label on the Media Card is similar to the following example:

ETHERNET ADDRESS
TLAN
00:60:38:BD:C9:9C
ELAN
00:60:38:BD:C9:9D

Ethernet activity LEDs
The faceplate contains six Ethernet activity LEDs: three for the ELAN network interface and three for the TLAN network interface. The LEDs indicate the following links on the ELAN network interface and TLAN network interface (in order from the top):

1. 100 (100BaseT)
2. 10 (10BaseT)
3. A (Activity)

Maintenance hex display
This is a four-digit LED-based hexadecimal display that provides the role of the card. It also provides an indication of fault conditions and the progress of PC Card-based software upgrades or backups.

RS-232 Maintenance Port
The Media Card faceplate provides a female eight-pin mini-DIN serial maintenance port connection. The faceplate on the card is labeled J2.

Media Card 32S
The Media Card 32S card (NTDW65AA) provides the following features:
• Secure Real-time Transport Protocol (SRTP)
• two Digital Signal Processors (DSP), based on an ARM processor
• channel density of 32 ports

Secure Real-time Transport Protocol
The Media Card 32S card uses Secure Real-time Transport Protocol (SRTP) to secure the IP media path between the card and the Nortel Phase II IP Phones, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, IP Phone 1210, IP Phone 1220, IP Phone 1230, another Media Card 32S card, or a DSP daughterboard. When Media Security is configured to On, the Call Server sends a message to the Voice Gateway software on the Media Card 32S card to activate SRTP for the media
connection established for that call.

Media Security is configured by the system administrator. For information about SRTP, see IP Phones Fundamentals
(NN43001-368) , and System Management Reference (NN43001-600). Processors Two separate processors on the MC 32S card are known as the Control and Signaling Processor (CSP) and the Media Stream Processor (MSP). The CSP runs application and signaling code, whereas the MSP processes the media streams.

Enable LED
The faceplate red LED indicates the following:
• the enabled and disabled status of the card
• the self-testing result during power-up or card insertion into an operational system

Ethernet port
The Ethernet port is used for debugging. It connects to the six-ports layer-2 switch through port three on the card and is mirrored to any other ports of the layer 2-switch.

Four character LED display
This is a four-digit LED-based hexadecimal display that provides the role of the card. It also provides an indication of fault conditions and the progress of PC Card-based software upgrades or backups.

Ethernet activity LEDs
The faceplate contains two Ethernet activity LEDs: one for the ELAN network interface and one for the TLAN network interface. The LEDs indicate the following links on the ELAN network interface and TLAN network interface:
• Link Activity
• Speed

RS-232 Maintenance port
The Media Card 32S card faceplate provides a female eight-pin mini-DIN serial maintenance port connection. Functional description of the Voice Gateway Media Cards The Media Cards perform three separate functions, depending on the
system in which the card is located, and the type of card:

1. The card acts as a gateway between the circuit-switched voice network and the IP network.
2. The card acts as a Line Terminal Proxy Server (LTPS) or virtual line card for the IP Phones.
3. The Media Card 32S card provides Secure Real-time Transport Protocol (SRTP) to secure the IP media path between the DSP channels and the card.

Gateway functional description
The Gateway performs the following functions:
• registers with the system using the TN Registration messages
• accepts commands from the system to connect or disconnect the audio channel
• uses Real-time Transport Protocol (RTP) and Real-time Control Protocol (RTCP) to transport audio between the gateway and the IP Phone or another gateway
• encodes and decodes audio from PCM to and from the IP Phone format
• provides echo cancellation for the speaker on IP Phones for echoes that originate in the circuit-switched voice network (not applicable to the IP Softphone 2050 or MVC 2050, as they have no handsfree capability)

Gateway functionality on the CS 1000 systems
A Signaling Server is always present in the CS 1000systems. The LTPS executes on the Signaling Server, and the Voice Gateway executes on the Voice Gateway Media Cards and DSP daughterboards. The Voice Gateway Media Cards only provide the voice gateway access.

Active Master
The LTPS maintains a count of the number of IP Phones registered to the card. Each IP Telephony node has one active Master. The active Master broadcasts to all LTPS and requests a response if it has room for another IP Phone.

IP Phone registration
This section describes the maximum number of IP Phones that can register to an LTPS in a CS 1000 system.
IP Phoneregistration on a CS 1000 system. On a CS 1000 system, the IP Phones register with the LTPS on the Signaling Server. If more than one Signaling Server exists, the IP Phone registrations are distributed equally among the Signaling Servers to aid in load balancing. If the primary Signaling Server fails, a secondary Signaling Server takes over (if it exists), and the IP Phones that are registered with the failed Signaling Server re-register with the LTPS on secondary Signaling Server. If no other Signaling Servers exist or if the Signaling Servers fail, the IP Phones register with the LTPS on the Voice Gateway Media Cards.

ATTENTION
Each Signaling Server supports the registration of up to 5000 IP Phones.

Signaling and messaging
The IP Line application sends Scan and Signaling Distribution (SSD) messages to the Call Server through the system ELAN subnet. When tone service is provided, the service is signaled to the LTPS by using new SSD messages sent through the ELAN subnet.

Signaling protocols
The signaling protocol between the IP Phoneand the IP Telephony node is the Unified Networks IP Stimulus Protocol (UNIStim). The Reliable User Datagram Protocol (RUDP) is the transport protocol.

Reliable User Datagram Protocol
Reliable User Datagram Protocol (RUDP) is used for:
• signaling between the Call Server and the LTPS
• signaling between the IP Telephony node and the IP Phones

Description
Signaling messages between the LTPS and IP Phones use RUDP. Each RUDP connection is distinguished by its IP address and port number. RUDP is another layer on top of User Datagram Protocol (UDP). RUDP is proprietary to Nortel.
The features of RUDP are as follows:
• provides reliable communication system over a network
• packets are resent if an acknowledgement message (ACK) is not received following a time-out
• messages arrive in the correct sequence
• duplicate messages are ignored
• loss of contact detection

When a data sequence is packetized and sent from source A to receiver B, RUDP adds a number to each packet header to indicate its order in the sequence.
• If the packet is successfully transmitted to B, B sends back an ACK to A, acknowledging that the packet has been received.
• If A receives no message within a configured time, it retransmits the packet.
• If B receives a packet without having first received its predecessor, it discards the packet and all subsequent packets, and a NAK (no acknowledge) message, which includes the number of the missed packet, is sent to A. A retransmits the first missed packet and continues.

UNIStim
The Unified Network IP Stimulus protocol (UNIStim) is the single point of contact between the various server components and the IP Phone. UNIStim is the stimulus-based protocol used for communication between an IP Phoneand an LTPS on the Voice Gateway Media Card or Signaling Server.

Secure Real-time Transport Protocol
The Media Card 32S card uses Secure Real-time Transport Protocol (SRTP) to secure the IP media path between the card and the Nortel Phase II IP Phones, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, or IP Phone 1150E, IP Phone 1210, IP Phone 1220, IP Phone 1230, or another Media Card 32S card, or a DSP daughterboard. When Media Security is configured to On, the Call Server sends a message to the Voice Gateway software on the Media Card 32S card to activate SRTP for the media connection established for that call. Media Security is configured by the system administrator. For information about SRTP, see IP Phones Fundamentals (NN43001-368), and System Management Reference (NN43001-600).

ELAN TCP transport
Although TCP is used for the signaling protocol between the Call Server and the Voice Gateway Media Card, RUDP remains for the Keep Alive mechanism for the link. This means that RUDP messages are exchanged to maintain the link status between the Call Server and the Signaling Server or the Voice Gateway Media Card. The TCP protocol enables messages to be bundled. Unlike the RUDP transport that creates a separate message for every signaling message (such as display updates or key messages), the TCP transport bundles a number of messages and sends them as one packet. Handshaking is added to the Call Server and IP Line software so that the TCP functionality is automatically enabled. A software version check is performed by the IP Line application each time before it attempts to establish a TCP link with the CS 1000 CPU. TCP transports messages, whereas RUDP establishes and maintains the link. The IP Line software version must match the Call Server software version; otherwise, IP Line terminates the link and logs an error message.

Virtual superloops, Virtual TNs, and physical TNs
Virtual TNs (VTN) enable configuration of service data for an IP Phone, such as key layout and class of service, without requiring the IP Phone to be dedicated (hard-wired) to a given TN on the Voice Gateway Media Card. The concentration of IP Phones is made possible by dynamically allocating a port (also referred to as a physical TN) of the Voice Gateway Media Card for a circuit-switched-to-IP Phone call. All system speech path management is done with physical TN instead of virtual TN. Calls are
made between an IP Phoneand circuit-switched telephone or trunks using the full CS 1000 feature set. Digital Signal Processor (DSP) channels are allocated dynamically for this type of call to perform the encoding or decoding required to connect the IP Phoneto the circuit-switched network. The IP Phones (virtual TN) are defined on virtual superloops. To create an IP Phoneusing VTNs, create a virtual superloop in LD 97 or in Element Manager. To create the virtual superloop in Element Manager,
click System > Core Equipment > Superloops in the Element Manager navigator. A virtual superloop is a hybrid of real and phantom superloops. Like phantom superloops, no hardware (for example, XPEC or line card) is used to define and enable units on a virtual superloop. As with real superloops, virtual superloops use the time slot map to handle IP Phone(virtual TN)-to-IP Phone calls. You can configure up to 1024 VTNs on a single virtual superloop for Large Systems, CS 1000M Cabinetand CS 1000M Chassissystems, and CS 1000E systems. Each Media Card 32-port card provides 32 physical TN. The physical TN
are the gateway channels (DSP ports), which provide 128 channels. The channels (ports) on the Voice Gateway Media Cards are pooled resources. Configure the physical TNs (IPTN) in LD 14. They appear as VGW data blocks.

Licenses
There are three types of licenses:
• Temporary IP User Licence for IP Phones configured for Branch Office or Network wide redundancy
• Basic IP User License for the IP Phone 2001, IP Audio Conference Phone 2033, IP Phone 1110, and IP Phone 1210
• IP User License for the IP Phone 2002, IP Phone 2004, IP Phone 2007, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, IP Phone 1220, IP Phone 1230, IP Softphone 2050, Mobile Voice Client (MVC) 2050, WLAN Handset 2210, WLAN Handset 2211, WLAN Handset 2212, WLAN Handset 6120, and WLAN Handset 6140 If insufficient Temporary IP User Licenses are available, Basic IP User License and IP User License can be used.

If insufficient Basic IP User Licenses are available for the IP Phone 2001, IP Audio Conference Phone 2033, , IP Phone 1110, and IP Phone 1210 then the IP User License can also be used. If there are no Basic IP User Licenses available for the IP Phone 2001, IP Audio Conference Phone 2033, , IP Phone 1110, and IP Phone 1210 and IP User Licenses are used, an error message is generated: "SCH1976: Basic IP User License counter has reached its maximum value. IP User License was used to configure basic IP Phone type 2001. Action: (Recommended) Purchase additional Basic IP User Licenses for IP Phones type 2001, instead of using higher-priced IP User Licenses."
Each time an IP Phone is configured, the system TN ISM counter is decremented. Customers must purchase one license for each IP Phoneinstalled on CS 1000 system. A new license uses the existing keycode to enable the IP Phone in the system software. The default is zero. To expand the license limits for the IP Phones, order and install a new CS 1000 keycode.

ATTENTION
Individual licenses are not supported on Functional Pricing. With Functional Pricing, licenses are provisioned in blocks of eight.

License limits
The total number of TN configured with Temporary IP User Licenses must not exceed 100. The total number of TN configured with Basic IP User Licenses must not exceed 32 767. The total number of TN configured with IP User Licenses must not exceed 32 767. The total number of IP phones configured within the system must not exceed the allowable system capacity limit controlled by customer keycodes.

Zones
To optimize IP Line traffic bandwidth use between different locations, the IP Line network is divided into zones, representing different topographical areas of the network. All IP Phones and IP Line ports are assigned a zone number, which indicates the zone to which they belong. When a call is made, the codecs that are used vary, depending on which zones the caller and receiver are in.

By default, when a zone is created in LD 117 or in Element Manager:
• codecs are selected to optimize voice quality (BQ - Best Quality) for connections between units in the same zone.
• codecs are selected to optimize voice quality (BQ - Best Quality) for connections between units in different zones.

Access zones in Element Manager by clicking IP Network > Zones in the Element Manager navigator. Configure each zone to:
• optimize either voice quality (BQ) or bandwidth usage (BB - Best Bandwidth) for calls between users in that zone
• optimize either voice quality or bandwidth usage within a zone and all traffic going out of a zone

Administration
The Voice Gateway Media Card is administered using multiple management interfaces, including the following:
• Web browser interface provided by Element Manager—Element Manager is used to administer Voice Gateway Media Cards in the systems that use a Signaling Server.
• Command Line Interface (CLI)—The CLI prompt, which displays depends on the type of Voice Gateway Media Card in the system.

Media Card 32-port. oam> or LBD> prompt displays for the Media Card 32S card.
• Administration and maintenance overlays of Call Servers.
• IP Line application GUI provided by TM 3.1. TM 3.1 is used to obtain OM reports only.

TM 3.1
IP Line uses TM 3.1 to obtain OM reports only.
Element Manager
Element Manager is a resident Web-based user interface used to configure and maintain CS 1000 components. Element Manager Web interface enables IP Line to be configured and managed from a Web browser. The Element Manager Web server resides on the Signaling Server or within Enterprise Common Manager (ECM) framework.

Description
Element Manager is a simple and user-friendly Web-based interface that supports a broad range of system management tasks, including:
• configuration and maintenance of IP Peer and IP Telephony features
• configuration and maintenance of traditional routes and trunks
• configuration and maintenance of numbering plans
• configuration of Call Server data blocks (such as configuration data, customer data, Common Equipment data, and D-channels)
• maintenance commands, system status inquiries, backup and restore functions
• software download, patch download, patch activation
• configuration of SNMP parameters (such as SNMP community strings, and SNMP trap destinations)

Element Manager has many features to help administrators manage systems with greater efficiency. Examples are as follows:
• Web pages provide a single point-of-access to parameters that are traditionally available through multiple overlays.
• Parameters are presented in logical groups to increase ease-of-use and speed-of-access.
• Administrators see information that relates directly to the task at hand by using the hide or show information option.
• Full-text descriptions of parameters and acronyms help administrators reduce configuration errors.
• Configuration screens offer preselected defaults, drop-down lists, check boxes, and range values to simplify response selection.

You can access Element Manager directly through a Web browser or Telephony Manager 3.1. The TM navigator includes integrated links to each network system and their respective instances of Element Manager.

Command Line Interface
The Command Line Interface (CLI) provides a text-based interface to perform specific Signaling Server and Voice Gateway Media Card installation, configuration, administration, and maintenance functions. Access Establish a CLI session by connecting a Teletype (TTY) or PC to the card serial port or Telnet through the ELAN or TLAN network interface IP
address.