SIP Functions and Features

When SIP was developed, it was designed to support five specific elements of setting up and tearing down communication sessions. These supported facets of the protocol are:

• User location, where the endpoint of a session can be identified and found, so that a session can be established
• User availability, where the participant that’s being called has the opportunity and ability to indicate whether he or she wishes to engage in the communication
• User capabilities, where the media that will be used in the communication is established, and the parameters of that media are agreed upon
• Session setup, where the parameters of the session are negotiated and established
• Session management, where the parameters of the session are modified, data is transferred, services are invoked, and the session is terminated

Although these are only a few of the issues needed to connect parties together so they can communicate, they are important ones that SIP is designed to address. However, beyond these functions, SIP uses other protocols to perform tasks necessary that allow participants to communicate with each other.

User Location

The ability to find the location of a user requires being able to translate a participant’s username to their current IP address of the computer being used. The reason this is so important is because the user may be using different computers, or (if DHCP is used) may have different IP addresses to identify the computer on the network. The program can use SIP to register the user with a server, providing a username and IP address to the server. Because a server now knows the current location of the user, other users can now find that user on the network. Requests are redirected through the proxy server to the user’s current location. By going through the server, other potential participants in a communication can find the user, and establish a session after acquiring their IP address.

User Availability

The user availability function of SIP allows a user to control whether he or she can be contacted. Users can set themselves as being away or busy, or available for certain types of communication. If available, other users can then invite the user to join in a type of communication (e.g., voice or videoconference), depending on the capabilities of the program being used.

User Capabilities

Determining the user’s capabilities involves determining what features are available on the programs being used by each of the parties, and then negotiating which can be used during the session. Because SIP can be used with different programs on different platforms, and can be used to establish a variety of single-media and multimedia communications, the type of communication and its parameters needs to be determined. For example, if you were to call a particular user, your computer might support video conferencing, but the person you’re calling doesn’t have a camera installed. Determining the user capabilities allows the participants to agree on which features, media types, and parameters will be used during a session.

Session Setup

Session setup is where the participants of the communication connect together. The user who is contacted to participate in a conversation will have their program “ring” or produce some other notification, and has the option of accepting or rejecting the communication. If accepted, the parameters of the session are agreed upon and established, and the two endpoints will have a session started, allowing them to communicate.

Session Management

Session management is the final function of SIP, and is used for modifying the session as it is in use. During the session, data will be transferred between the participants, and the types of media used may change. For example, during a voice conversation, the participants may decide to invoke other services available through the program, and change to a video conferencing. During communication, they may also decide to add or drop other participants, place a call on hold, have the call transferred, and finally terminate the session by ending their conversation. These are all aspects of session management, which are performed through SIP.

SIP URIs

Because SIP was based on existing standards that had already been proven on the Internet, it uses established methods for identifying and connecting endpoints together. This is particularly seen in the addressing scheme that it uses to identify different SIP accounts. SIP uses addresses that are similar to e-mail addresses. The hierarchical URI shows the domain where a user’s account is located, and a host name or phone number that serves as the user’s account. For example, SIP: myaccount@madeupsip.com shows that the account myaccount is located at the domain madeupsip.com. Using this method makes it simple to connect someone to a particular phone number or username.

Because the addresses of those using SIP follow a username@domainname format, the usernames created for accounts must be unique within the namespace. Usernames and phone numbers must be unique as they identify which account belongs to a specific person, and used when someone attempts sending a message or placing a call to someone else. Because the usernames are stored on centralized servers, the server can determine whether a particular username is available or not when a person initially sets up an account.

URIs also can contain other information that allows it to connect to a particular user, such as a port number, password, or other parameters. In addition to this, although SIP URIs will generally begin with SIP:, others will begin with SIPS:, which indicates that the information must be sent over a secure transmission. In such cases, the data and messages transmitted are transported using the Transport Layer Security (TLS) protocol

Testing Unified Messaging Functionality

Testing the UM functionality is a bit trickier than testing other features such as message routing because it involves multiple components in the testing process such as voice, mailbox access, and so on. Only when testing this functionality end-to-end can you make sure it is working as expected. Several tools are available for testing your UM functionality.

UM Troubleshooting Tool

The UM Troubleshooting Tool is available with Exchange 2010 SP1 as a separate download to proactively test the voice mail functionality and identify any issues. The Troubleshooting Tool is able to simulate a call from OCS or IP gateway to your UM server and verifies that the UM communication is working as expected. It verifies that a call can be established, verifies that the audio flow from the UM server works, and prepares quality metrics for recorded audio.

You can install the UM Troubleshooting Tool on a workstation or server. The recommendation is to use an administrative workstation. It is available in x86 and x64 versions and requires the following prerequisites to be installed:

·         Windows PowerShell v2

·         .Net Framework 3.5 SP1

·         Unified Communications Managed API (UCMA) v3.5

The UM Troubleshooting Tool provides you with a shell similar to the EMS and allows you to test UM connectivity with the Test-ExchangeUMCallFlow cmdlet, as shown in Figure 1.

Figure 1: Microsoft Exchange UM Troubleshooting Tool

 

If you want to use the UM Troubleshooting Tool to test your UM server, you need to create the following:

·         A UM dial plan with Telephone Extension as the URI type and Unsecured as VoIP security

·         A UM IP gateway that points to the IP address of the workstation you installed the UM Troubleshooting Tool on

·         A UM-enabled mailbox with an extension

After you create the prerequisites, run the Test-ExchangeUMCallFlow -Mode GatewayEmulator -VoIPSecurity Unsecured -NextHopAddress -Diversion cmdlet to verify that the UM server is working correctly.

Note

You can use this tool to run against Exchange 2010 UM SP1 servers only!

Exchange UM Test Phone

The Microsoft Exchange UM Test Phone is a software phone that you can use to connect to your UM server and simulate specific IP gateway settings. It is based on the Exchange Speech Engine and can be used to troubleshoot connectivity.

The UM Test Phone (ExchangeUMTestPhone.exe) is no longer available on the Exchange 2010 DVD, but you can get it from an Exchange 2007 DVD and use it against your Exchange 2010 UM server.

Important

UM uses the Unified Communications Managed API 2.0 Core SDK (UCMA) in Exchange 2010 SP1; the Exchange UM Test Phone cannot connect anymore to run against SP1. You can only use it with Exchange 2010 RTM.

You can install the Exchange UM Test Phone on a workstation or server that includes a microphone and speakers so you can verify that the speech is accurate and correct. Like the Exchange 2007 installation files, it is available in x86 and x64 versions. The UM Test Phone is shown in Figure 2.

Figure 2: Using the UM Test Phone 

Detailed information about how to test your UM server with the UM Test Phone can be found at http://technet.microsoft.com/en-us/library/aa997146(EXCHG.80).aspx.

Call Processing Feature/Function Glossary and Definitions

This appendix is an abridged glossary of voice calling features that are available on leading PBX systems. This glossary is intended to be representative of the most popularly requested customer features, but it does not include all of the currently available features. Each PBX system feature set is unique, and feature capabilities differ across manufacturers’ models. Although the typical PBX station user may commonly use only a small fraction of these features under normal working conditions, each feature has some productivity and/or cost savings value. PBX systems that lack more than 15 percent of these features cannot be said to be competitive in the marketplace.

Station User Features

Add-on Conference

This feature allows a station user to add a third party to an existing two-party conversation.

Automatic Callback

This feature is used when a dialed station is busy. When the feature is activated, the system automatically attempts to call the desired station until the line is free. The calling party is alerted that the called party is available. This saves wasted time dialing when encountering busy signals.

Automatic Intercom

This feature provides a talking path between two voice terminal users. A station user presses a programmed automatic intercom button and lifts the handset, or vice versa. The called user receives a unique intercom alerting signal, and the status lamp associated with the dial or automatic intercom button, if provided, flashes.

Bridged Call Appearance

This feature allows the same line appearance to be programmed to appear on more than one telephone. It is very useful in manager–assistant relationships and call answering position environments. A bridged call appearance can reduce the number of abandoned or lost calls and allows the coverage station to prescreen calls for the called party.

Call-Back Last Internal Caller

This feature allows a station user to automatically consult and call back the last internal caller to the station (unanswered call) by implementing the feature code.

Call Forwarding—All Calls/No Answer/Busy

This feature allows a station user to divert all incoming calls to another programmed station. All Call activation diverts all incoming calls to the station; no answer activation diverts calls after a programmed number of rings; busy activation diverts incoming calls when the station is busy. The features are useful when a station user is away from the desk area, wishes to receive calls at another station, or when there is a desire not to receive calls, but the user wants the call answered. The most common coverage station is a voice messaging port. Call forwarding features decrease abandoned or lost calls and improve call coverage service for the calling party.

Call Forwarding—Follow Me

This feature allows a station user to activate the call forwarding feature from a remote telephone by changing an existing forwarding destination. It provides station users with the capability of changing call forwarding destinations without returning to their desks, and can be used to “follow” station users around the system if they wish to receive calls at different stations.

Call Forwarding—Off Premises

This feature allows a station to forward all calls to an off-premises location outside the system. It allows station users to receive calls at a programmed outside telephone line when they are out of the office.

Call Forwarding—Ringing

This feature allows a station user of a multiple line voice terminal with display to forward an incoming call during the ringing period to another station. The station user can read the display for screening information, such as CLID or calling party name, before activating the feature through a programmed feature button or dial access code. The rerouted call destination is input by the station user after feature activation. The operation is transparent to the calling party.

Call Forwarding—Selective Multiple Line

This feature allows a station user of a multiple line voice terminal to selectively call forward any or all line appearance numbers.

Call Hold

This feature allows a station user to place an existing call in a hold state when there is another incoming call or the station user must leave the desktop area for more than a few seconds. Call hold provides station users with the flexibility of handling multiple concurrent calls without re-establishing the connection after finishing another call.

Call Park

This feature allows a station user to “park” a call at the received station, effectively placing the call in a hold state, retrieve the call at another station, and continue the conversation. A second party can also retrieve the parked call if notified by the first party. The feature provides station users with mobility and eliminates the need to return calls.

Call Pickup

The feature allows a station user to retrieve and answer a call directed to another station (direct), any station in the station user’s assigned call pickup group (group), or another call pickup group (designated group). The station user presses a programmed call pickup button or dials the desired feature access code to implement the feature.

Call Transfer

This feature allows a station user to divert an existing call to another station within the system. It eliminates the need for the original calling party to hang up and redial another telephone line to reach the desired or proper called party.

Call Waiting

This feature notifies a station user engaged in conversation that there is another incoming call to the station line number. The notification is usually a special tone or display signal on the telephone. Call waiting reduces lost calls, improves customer service, and reduces the number of calls forwarded to another station or messaging system.

Consecutive Speed Dialing

For speed dialing all station number digits are registered as the speed dial code. This feature allows a common set of partial station number digits to be registered as the speed dial code, and allows the station user to dial the remaining digits of each number to establish the call.

Consultation (Broker) Hold

This feature allows a station user to place one party on hold and confer with a third party on another line. This feature reduces call backs and improves customer service. Some systems allow the station user to toggle back and forth between two lines.

Customer Station Rearrangement

This feature allows station users to physically relocate their multiple line voice terminals internal to the system. When a station user moves between locations, the voice terminal station (number and COS) is logically transferred. This service is activated as follows: dialing the “moving” feature code followed by a personal code; the terminal is out of service; the terminal is disconnected, moved, and reconnected in the new location; the terminal is reinitialized by dialing its extension number, followed by the personal code; the station rings immediately and when answered, the set is validated. The former and latter locations must support the same category of voice terminal.

Discrete Call Observing

This feature allows a supervisor to monitor a conversation between an assistant and a caller on a preselected line. While the supervisor is listening, the voice terminal microphone is off and the assistant is informed that the feature is activated by a notice on the voice terminal display. During monitoring, the supervisor can take over the call.

Distinctive Ringing

This feature allows the system administrator to define distinct ringing patterns for different call types, such as internal, defined internal line, external, private network, emergency, and private line. Station users can use this feature as a call screening device to decide which calls to answer and which are to be forwarded to a coverage station. The number of distinct patterns differs greatly between different system models.

Dial by Name

This feature allows a station user with a voice terminal equipped with an alphabetical keyboard to call an internal extension or external number by typing in a name using last name, first name, or initials. The directory database can be locally stored in the voice terminal or accessed from a centrally located database in the PBX system or application.

Do Not Disturb

This feature allows a voice terminal user to request that no calls, other than priority calls, terminate at a particular extension number until a specified time. At the specified time, the system automatically deactivates the feature and allows calls to terminate normally at the extension.

Elapsed Call Timer

This feature provides a display of the elapsed time when a multiple line voice terminal is connected to any trunk circuit.

Emergency Access to Attendant

This feature allows emergency calls to be placed to an attendant with special priority status. Calls can be placed automatically when the telephone is in an off-hook state or by dial access. The attendant receives visual and audible feedback when the call is received. The feature is important for situations requiring immediate attendant access.

Executive Busy Override

This feature allows a station user to cut into an internal party’s conversation. This feature decreases call backs, saves time reaching the called party, and decreases calls sent to coverage positions.

Executive Calling

This feature allows a station to be assigned VIP class status. The feature allows a VIP station to send a special ringing signal to a called station when idle and automatically send multiple tone bursts to that station when busy.

Executive Access Override

This feature allows a station user to connect a call to an internal extension that is in call forward or do not disturb mode. The service is authorized by COS level.

External Paging with Meet-Me

This feature allows a station user or attendant to dial a local paging equipment access code and connect both parties automatically after the paged party has answered the page and dials a special access code.

Facility Busy Indication

This feature provides multiple line telephone users with a visual indication of the busy or idle status of internal station numbers, trunk groups, hunt groups, or paging zones. Station users can monitor the activity of frequently called numbers with this feature, eliminating encountered busy signals.

Group Listening

This feature allows a station user of a multiple line voice terminal with an integrated speakerphone to place a call using the handset and activate the terminal’s built-in speaker, to allow others to listen to the conversation while the station user continues talking through the handset.

Hands-free Dialing

This feature allows a station user of a voice terminal with a built-in speaker to dial and monitor a call without lifting the handset.

Hands-free Intercom

This feature allows a station user of a voice terminal with a built-in speaker to answer a voice call without lifting the handset. The incoming voice call is heard over the speaker.

Help/Information Key

This feature provides a station user of a multiple line voice terminal with display immediate access to help menus for terminal programming and feature access procedures. Information is displayed by the system in user-friendly way. If a feature access code is changed, it will be displayed automatically in the feature menu. Service consultation shows all the relevant functions and their associated feature codes. This feature allows self-training on the voice terminal and also reduces the need for paper labels on telephone features.

Hot Line

This feature automatically dials calls to preassigned internal stations, off-premises stations, or feature access codes when the handset is lifted. It eliminates the need to dial a number or access code, thereby simplifying and accelerating the process.

Incoming Call Display

This feature, available on telephones with display fields, provides visual notification to the station user of the calling party’s station number or incoming trunk group name. The calling party’s name may also be displayed with the station number. The feature is a screening device to decide whether to answer or divert the call.

Individual Attendant Access

This feature allows users to access a specific attendant console. Each attendant console can be assigned an individual extension number.

Intercom Dial

This feature allows multiple line voice terminal users to gain quick access to select other voice terminal users within an administered group. Calling voice terminal users lift the handset, press the dial intercom button, and dial the one- or two-digit code assigned to the desired party. The called user receives an alerting tone, and the status lamp associated with the Intercom button, if provided, flashes.

Last Number Redialed

This feature stores the last number dialed by the station user and allows the station user to automatically dial the number by using a programmed feature button or feature access code. It simplifies the calling process, reduces misdialed calls, and saves time.

Line Lockout

This feature removes single line voice terminal extension numbers from service when users fail to hang up after receiving dial tone signals, fol- lowed by intercept tone signals. The intervals for each tone signal are administrable.

Loudspeaker Paging Access

This feature provides station users or attendants dialing access to voice paging systems. This is useful for paging purposes regardless of the station user’s location within the premises environment. It is often used with the call park feature.

Malicious Call Trace

This feature allows a station user to notify a predefined set of station positions that a malicious call is in process. The notified station users can then gather information and data about the call to identify the calling source. The feature is useful when a CLID or ANI is not displayed.

Manual Intercom

This feature allows a station user to call a manual intercom group member by pressing the manual intercom button. All member of a manual intercom group share a common signaling path. When the manual intercom button is pressed, a special tone burst is sent over the voice terminal speakers of all group members. When a group member answers, a speech path is established.

Manual Originating Line Service

This feature connects single line voice terminal users to the attendant automatically when the user lifts the handset. The attendant code is stored in an abbreviated dialing list. When the manual originating line service voice terminal user lifts the handset, the system automatically routes the call to the attendant using the hot line service feature.

Manual Signaling

This feature allows a voice terminal user to signal another voice terminal user. The receiving voice terminal user hears a short burst of tone. The signal is sent each time the button is pressed. If the receiving voice terminal is already being alerted with an incoming call, manual signaling is denied.

Message Waiting

This feature enables multiple line appearance voice terminal users, by pressing a designated button on their own terminals, to light the status lamp associated with the message waiting button at another multiple line appearance voice terminal. Activating the feature causes the lamp to light on the originating and receiving voice terminals. Either terminal user can cause the lamp to go dark by pressing the button.

Multiparty Conferencing

This feature allows multiappearance voice terminal users to set up multiparty conferences (typically between four to eight station users) without attendant assistance.

Music on Hold

This feature provides music to a party that is on hold, waiting in a queue, parked, or on a trunk call that is being transferred. The music lets the waiting party know that the connection is still in effect. The system provides automatic access to the music source.

Off-hook Alarm

This feature provides a special alerting tone to a station user who does not hang up the handset after receiving a busy signal. The tone signal is sent after a programmed interval.

Padlock

This feature allows a station user to temporarily prevent outgoing calls from the voice terminal. The selection of an external line by feature code, programmable key, or supervision key is controlled. Dialing the appropriate feature code followed by the personal code reactivates direct access.

Paging/Code Call Access

This feature allows voice terminal users, attendants, and tie trunk users to page with coded chime signals. Multiple individual paging zones can be provided.

Personal Speed Dialing

This feature allows a station user to program personal speed dial numbers at the station instrument. The speed dial feature is activated by using a programmed feature button or access code and pressing a one-or two-digit access code. The feature simplifies the dialing process, saves time, and decreases misdialed numbers.

Personalized Ringing

This feature allows users of certain voice terminals to uniquely identify their own calls. Each user can choose one of a number of possible ringing patterns.

Priority Calling

This feature provides a special form of call alerting between internal voice terminal users. The called voice terminal user receives a distinctive, administrable alerting signal.

Private Line

This feature provides a dedicated trunk for direct access to or from the public network for multiple line appearance voice terminal users.

Privacy—Attendant Lockout

This feature prevents an attendant from re-entering a multiple-party connection held on the console unless recalled by a voice terminal user.

Privacy—Manual Exclusion

This feature allows multiple line appearance voice terminal users to keep other users with appearances of the same extension number from bridging onto an existing call.

Recall Signaling

This feature allows an analog station user to place a call on hold and consult with another party or activate a feature. After consulting with that third party, the user can conference the third party with the original party by another recall signal or return to the original party by flashing the switchhook twice.

Ringer Cutoff

This feature allows the user of a multiple line appearance voice terminal to turn certain audible ringing signals on and off. The feature does not affect visual alerting.

Ringing Tone Control

This feature allows station users of multiple line voice terminals to select from a menu of ringing tone melodies and to adjust the volume level of ringing.

Save and Redial

This feature allows a station user to save a specific dialed number and then redial the number at a later time. The station user stores and redi-als the number by pressing a save and redial feature key.

Secondary Extension Feature Activation

This feature allows a multiple line voice terminal station user to access a line appearance of another extension, and program a limited set of features, such as call forwarding and call pickup, from that extension.

Send All Calls

This feature allows users to temporarily divert all incoming calls to coverage regardless of the assigned call coverage redirection criteria. The feature also allows covering users to temporarily remove their voice terminals from the coverage path.

Step Call

This feature allows a station user or attendant, after dialing a busy station, to dial an idle station by simply dialing an additional digit. The feature can be implemented only if the dialed digits of the first dialed number and the second number are identical, except for the last digit.

Store/Redial

This feature allows a station user of a multiple line voice terminal to store a particular number for later use. A store/redial key is programmed and assigned to this function.

Supervisor/Assistant Calling

This feature allows a station user with a multiple line voice terminal, who is an assistant to a supervisor, to use a call appearance of the supervisor’s station to screen calls for the supervisor and announce and/or transfer calls to that extension. The assistant can also dial the supervisor during a busy condition and send a message waiting notification to the supervisor.

Supervisor/Assistant Speed Dial

This feature allows a pair of station users to use a programmed feature key to direct speed calls between a supervisor and an assistant, even if forwarding is validated.

Text Messages

This feature allows station users to leave a short text message for other internal users. Messages are stored in the main system database, and are available for selection via a menu on display-based voice. Calling parties can also receive messages from a voice terminal station that are preselected by the called party during no answer or busy conditions. There may be three structures of messages: preprogrammed fixed messages fully defined by system management, part programmable messages defined by system management but to be completed by the station user if the voice terminal has an alphanumeric keyboard, and fully programmable messages written entirely by the station user and offered only by sets provided with an alphanumeric keyboard.

Timed Queue

When a multiple line voice terminal station user originates an outgoing trunk call and encounters a no answer or busy condition, the timed queue feature can be implemented. After pressing a programmed feature button or dialing the feature access code, trunk seizure is repeated and the external station number is dialed after a predetermined interval.

Trunk Flash

This feature enables multifunction voice terminals to access CLASS features that are provided by the far-end CO switching system located directly behind the PBX system. CLASS services are accessed by a sequence of flash and dial signals from the station on an active trunk call. The feature can decrease the number of trunk lines connected to the PBX system by performing trunk-to-trunk call transfers at the far-end CO, which eliminates the use of a second trunk line for the duration of the call and frees the original trunk line for the duration of the call. It can also be used to set up a conference call with a second outside call party, which eliminates the need for a second trunk line for the duration of the call.

Trunk-to-Trunk Connection

This feature allows a station user to conference together two outside trunk calls and abandon the connection without dropping the two trunk-to-trunk connections.

Qsig

Qsig is an inter-PBX signaling system designed for multiple PBX system platform networks. The proprietary nature of IFTN solutions restricted customer configuration flexibility to a single supplier’s product platform. Qsig in its current form originated during the 1990s as a standardization effort by the IPSN Forum, a group of Western European PBX equipment suppliers, with Siemens and Alcatel at the forefront of the movement. IPSN work efforts were handed off to the ECMA and the International Telecommunications Union (ITU) for the formalization of issuing standards and specifications. Qsig is based on the ITU’s Q.93x series of recommendations for basic services and generic features and Q.95x series for supplementary services.

The major benefits for developing Qsig were outlined in the Qsig handbook originally published by the IPNS Forum.

Vendor independence. The nonproprietary nature of Qsig, based on open international standards and supported by all of the leading global PBX suppliers, allows customers to configure an intelligent communications system network when using equipment from more than one supplier.

Guaranteed interoperability. A memorandum of understanding (MoU) signed by the leading global suppliers signifies commitment to Qsig specifications, facilitates interoperability performance tests, and assures customers that they will be able to operate communications networks with a mix of supplier equipment.

Free-form topology. Qsig does not impose the use of a specific network topology, so it can be implemented with any network configuration: meshed, star, main/satellite, etc. Existing networks can, regardless of their topology, be upgraded to Qsig. Newly designed networks can be installed with the most effective and economical topology.

Unlimited number of nodes. There are no nodal limits for a Qsig network. New nodes can be added as needed.

Flexible numbering plan. Qsig does not impose any number plan restrictions for the network, thereby allowing customers to freely adopt customized numbering plans.

Flexible interconnection. Qsig will work over any type transmission network for linking PBX systems, including two- and four-wire analog tie lines, digital leased lines (including ISDN PRI and BRI), radio and satellite links, and VPN services provided by interchange carriers. Associated transmission delays are managed and controlled according to Qsig specifications.

Public ISDN synergy. There is network service compatibility between public and private ISDN transmission facilities. Applications developed for desktop terminals connected directly to a public ISDN network will also be available to desktop terminals provisioned within the Qsig-based customer private network.

Supplementary services for corporate users. Qsig supports private communications features beyond those defined for public ISDN networks, including caller name ID, call intrusion, do not disturb, path replacement, operator services, mobility services, and call completion on no reply.

Feature transparency. Features and functions supported by any network node can be transparently supported across the network to station users configured behind other network nodes. Qsig is structured and organized to adapt to service levels offered by different PBX systems, and it allows each network node to provide only the required level of service. There is an exchange of high-level services between any two nodes, via transit nodes with lower service levels: transit nodes pass communications and control signals between systems.

Innovation. Qsig does not restrict individual PBX manufacturers from developing customized, unique features. A special mechanism within Qsig, generic functional procedures (QSIG GF), provides a standardized method for transporting nonstandard Qsig features. As defined in Qsig GF, the basic rules related to feature transparency allow end-to-end communication through the network, regardless of network structure. Qsig does not prevent the use of innovative, proprietary system features across the customer network and allows for customized new feature development negotiation between PBX suppliers and customers.

Multiapplication domain. Qsig is not restricted to PBX systems and can support applications requiring other peripheral communications equipment, such as VMS, fax servers, data processing equipment, and multipoint conferencing systems.

Evolution. Qsig has an evolutionary path to support communications features, functions, and applications that are developed in the future.

Qsig Architecture

Qsig standards specify a signaling system at the ITU-T ISDN “Q” reference point, which is intended primarily for use on a common channel, although Qsig can be implemented over any suitable inter-PBX connection platform. The “Q” reference point, the logical signaling point between two PBXs, was defined explicitly for the Qsig. The physical connection point to the PBX system is made at the “C” reference (also a new ISDN reference point). There are three sublayer protocols at Layer 3, including the Qsif GF procedures. Qsig GF protocol provides a standardized mechanism to exchange signaling information for the control of supplementary services and additional network features (ANFs) over customer networks.

Qsig basic call (BC) message sequence is an intermediary transit node linking two endpoint PBX systems. Qsig BC is a symmetrical protocol designed for peer-to-peer networks, and it includes transit node capability.

ECMA also has been working on enhancements to its Qsig specifications to support broadband PBX networks. B-Qsig is an extension of Qsig, using many standards as possible available from the ITU-T and ATM forums.

Qsig Supplementary Services and ANFs

The following is a listing of the Qsig supplementary services and ANFs:

• Advice of charge

• Call completion

• Call forwarding and diversion

• Call interception

• Call intrusion

• Call offer

• Call transfer

• Call waiting

• Direct dialing in

• Do not disturb

• Identification services

• CLID presentation

• Connected line identification presentation

• Calling/connected line identification restriction

• Calling name identification presentation

• Calling/connected name identification restriction

• Mobile

• Multiple subscriber number

• Operator services

• Path replacement

• Recall

• Subaddressing

• User-to-user signaling

IFTN Features and Functions

There is no standard level of IFTN feature/function transparency within the PBX industry. Some PBXs support a very high percentage of features and functions across multiple networked systems, up to 90 percent of the total generic software program, and some support less than 50 percent. Almost all PBX system IFTN options support the following basic feature/functions:

• Basic calling with the use of a flexible dialing plan (typically four or five digits)

• Voice terminal display information (calling party/called party name and number, call redirection information)

• Call forwarding services

• Call transfer

• Call conferencing

• Automatic callback

• Bridged call appearance

• Message waiting indication

• Trunk release

• Network-wide attendant services

• Network-wide CDR

An important category of features supported by only a few IFTN solutions is ACD. For example, all 55 of the identifiable NEAX 2400 IPX ACD features are available with Fusion CCS. The NEC ACD Agent Anywhere option is an intelligent network of multiple ACD systems using Fusion CCS links. ACD nodes can communicate with each other and pass and interpret signaling, caller ID, call prompt, and database information across the network. Intelligent interflow routing of callers between nodes improves customer service levels, balances traffic load, and optimizes agent productivity. Fusion CCS also supports centralized management reporting and supervisor workstation data screens. A multiple system ACD network has built-in redundancy to reduce system down time and increase customer satisfaction.

The Agent Anywhere option supports distributed ACD agents behind switch nodes remote from a centralized ACD processor node. ACD agents can be deployed anywhere within a Fusion CCS network, with the only restriction being that the remote switch node be directly linked, (no pass through signaling), to the control switch node. Agent Anywhere can be implemented when using internal or external ACD processing and software options. The Fusion CCS solution supports decentralized agents assigned to the same call split across multiple nodes. Incoming calls to the central control node can be routed to available agents in remote nodes if all agents are busy at the central location. For configu- rations with local incoming trunking at remote node locations, calls can be queued at the central control node location when no remote agents are available.

Many customers’ ACD-based call center systems include a CTI application server. A centralized CTI application server capable of supporting more than one PBX/ACD switch node is less costly and easier to manage and maintain than application servers at each customer location. The NEC Fusion CCS option supports a centralized CTI application server for ACD systems. Another good example of a centralized application server solution used within an IFTN configuration is the Siemens HiPath Allserve 150. A centralized Windows NT application server can support a network of one to four Siemens Hicom 150H systems networked with the Siemens CorNet option implemented over an IP LAN/WAN. The applications, run on a single server for all networked PBXs, include messaging, call center, personal call manager, and call accounting.

Perhaps the most important transparent system operations are management and control from a centralized application server. A central database for all Hicom 150H switch nodes resides in the application server. The centralized server provides one access point to administer and maintain each system. Station move/add/change transactions are implemented as if there were a single system, not multiple switch nodes. A single management or maintenance command can be applied to all switch nodes across the network, instead of inputting individual changes to individual systems. Centralized management system capabilities for all move, add, and change procedures is an important IFTN capability that is not commonly supported by most traditional circuit switched PBX IFTN options but is supported by more of the newer client/server IP-PBX system designs, such as Cisco’s AVVID IP telephony system.

Shared applications resources for call center, messaging, and management operations are an important IFTN cost savings benefit. The first IFTN offerings were limited to shared VMS applications. One VMS supported station users across a network of PBX systems. One of the cost savings components is attributable to the lower price for a single, very large messaging system as opposed to the collective cost of several smaller systems with equal voice mailbox and storage capabilities. Another cost savings component is ongoing management and service. Maintaining one messaging system is less costly and more efficient than maintaining several systems. The same cost savings criteria can be attributed to other shared application resources in an IFTN configuration.

Distinct IP Telephony Features/Functions

Integrated port interfaces. Compared with a legacy digital telephone, an IP telephone can be designed and equipped to provide several unique feature/function capabilities. An IP telephone design attribute not available with traditional digital telephones is the integration of a multiport Ethernet hub/switch to allow multidevice sharing of a single connector port to the Ethernet switched network. Most current IP telephone models are equipped with two Ethernet port connectors: one connector for the Ethernet network and one for a desktop PC client. Mitel Networks has indicated that its next-generation models will have another external connector port to support two Ethernet devices external to the telephone. An integrated Ethernet port interface reduces telecommunications outlets, inside wiring, and Ethernet switch port requirements. Cisco Systems was the first supplier to incorporate an integrated Ethernet switch into its IP telephones in its 7900 series. Mitel Networks followed Cisco’s approach by including integrated Ethernet switch ports in its second generation of IP telephones. Avaya, still marketing its first generation of IP telephones, offers its IP telephones with an integrated Ethernet hub.

The difference between an IP telephone with an integrated switch or hub may not be important to most customers, but providing a high level of voice-grade communications to the desktop is of primary importance. Voice communications QoS at the desktop can be supported using a variety of methods, such as Ethernet LAN 802.1 p/Q, or CoS programming (by switch or hub port). For example, each Cisco 7900 IP telephone internal Ethernet port can be programmed for different classes of service; the default service level of the voice port is a 5 and the data port is a 0. The system administrator can override the default service levels, if required, by an individual desktop station user. IP telephones with an internal Ethernet hub must include customized software to prioritize voice communications.

Besides Ethernet port connectors, current IP telephones may also support peripheral data devices through a USB port or infrared interface to a PDA. A USB port theoretically can be used for a variety of devices, such as printers, scanners, or digital cameras. There are several reasons to link a PDA through an infrared interface, including dialing from the directory or programming. Mitel Networks has introduced an IP telephone model with a docking station interface for a PDA. The PDA likely would function as the instrument’s display field, and provide data download capabilities for call processing and handling applications.

Ethernet power distribution. IP telephones, like PC clients, require power. Traditional PBXs power analog and digital telephones use internal power supplies to distribute power over inside telephony wiring. Converged (IP-enabled circuit switched) IP-PBX cannot distribute power across integrated IP gateway circuit cards to the LAN; neither can the LAN-connected telephony servers used in client/server IP-PBX designs. The first generation IP telephones were powered with an AC/DC transformer connected to a local AC power outlet. Each IP telephone required its own transformer and a dedicated UPS for emergency power support. Although an IEEE subcommittee had been working on its recommended standard for in-line power over an Ethernet LAN, IEEE 802.3af, Cisco could not wait and developed its own proprietary solution. Other proprietary solutions soon followed from other IP-PBX system suppliers, including 3Com and Alcatel. Third-party solutions, from suppliers such as PowerDsine, are available and work with IP telephones from other leading IP-PBX suppliers, such as Avaya and Siemens. In-line power options are currently priced at $50 to $100 per Ethernet port, but prices are expected to decline over time.

An Ethernet switch is equipped with an integrated or external power patch module, and power is distributed directly only to IP telephones, supported by the switch. Power is transmitted over unused Ethernet cabling wire pairs to only those Ethernet ports identifying themselves to the switch as IP telephone devices. IP telephones identify themselves to the LAN switch during an automatic self-discovery installation method or through manual programming by the system administrator. The Ethernet switch queries the IP telephone as to how much power is required or assumes a default power level.

Some of the basic specifications of IEEE 802.3af are:

• DTE power shall use two-pair powering, where each wire in the pair is at the same nominal potential and the power supply potential is between the two pairs selected.

• The power detection and power feed shall operate on the same set of pairs.

• The DTE power maximum voltage shall not exceed the limits of SELV per IEC 950.

• For DC systems, the minimum output voltage of the source equipment power supply shall be at least 40 V DC.

• For DC systems, the source device shall be capable of supplying a minimum current of at least 300 mA per port.

• The solution for DTE powering shall support mid-span insertion of the power source.

• 802.3af systems shall distribute DC power.

Until the IEEE 802.3af standard is finalized, IP telephones will continue to be powered by available in-line power options or local AC power transformers. need for stand-alone telephony gateway equipment linking a traditional PBX system and an IP router. Calls placed from an IP telephone can be routed directly across a LAN and WAN without IP telephony servers.

Compressed voice. Traditional digital telephones are designed with codecs that digitize analog voice signals into digital format using 8-bit word encoding and 8-KHz sampling, resulting in 64-Kbps digital transmission over inside wiring and across the internal PBX switching network. IP telephones can compress voice signals for lower transmission rates and decreased bandwidth requirements. The most common digital encoding schemes currently used for voice transmission over Ethernet and IP WAN networks are G.711 (64 Kbps), G.723.1 (5.3 to 6.3 Kbps), and G.729/A (8 Kbps). G.711 is traditional PCM (no compression), but the two other codec specifications use compression algorithms. The total bandwidth used for voice transmission with IP transmission protocol is greater than the noted transmission rates; about 16 Kbps of additional transmission bandwidth is required because an IP destination address and overhead signaling bits are added to the voice datagram packets. Compressed voice transmission creates an overhead delay factor that may affect the quality of a conversation, but the trade-off is the potential for more efficient use of expensive off-premises network transmission resources. A T1 carrier circuit that typically supports a maximum of 24 voice-grade channels can support an equal or greater number of voice channels, with sufficient available bandwidth for concurrent data communications transmission, if voice is encoded using G.729/A compression. Using an IP telephone for voice compression eliminates the need for stand-alone telephony gateway equipment linking a traditional PBX system and an IP router. Calls placed from an IP telephone can be routed directly across a LAN and WAN without IP telephony servers.

Other IP telephone functions that reduce transmission bandwidth requirements are VAD and silence suppression. VAD detects voice communications signals entering the handset mouthpiece (microphone), and silence suppression signals the onset of “silent” voice transmission. A telephone call usually has a high percentage of silence during a conversation between parties, often as much as 50 percent of total talk time. A circuit switched connection is highly inefficient because much of the time there is no voice activity, but 8-bit words of “silence” are transmitted. With VAD and silence suppression, an IP telephone can reduce bandwidth transmission requirements because packets are not continually transmitted when no one is talking. When there are no voice communications signals picked up by the IP telephone microphone, a special signaling packet is transmitted to the destination IP address indicating the beginning of a silent period, when no new voice packets are being transmitted between the two endpoints. When voice activity resumes, another signaling packet is forwarded to inform the destination IP address that incoming voice packets are now on their way, effectively ending the period of silence. VAD and silence suppression packets are transmitted only when someone is actually talking, resulting in fewer packets and more efficient use of network resources.

Web browser. The most significant feature difference between a legacy digital telephone and an IP telephone is the integration of an embedded Web browser and pixel-based display monitor. The first question most people ask about Web-enabled IP telephones is: “Why do I need a telephone with Internet access if I have a PC?” The manufacturers of Web-enabled IP telephones are quick to point out that their product should not be considered a replacement for a fully functional PC client, but as a supplemental communications device for access to information when data processing is not required. These new IP telephones are best described as network communications portals that combine telephony functions with access to network information servers.

Thin client IP telephones have many of the internal design attributes of a computer: CPU, memory, operating system, applications software, and embedded communications protocol stacks. The RTOS of the thin client IP telephone may be proprietary, as in the Cisco Systems 7940/7960 models or the popular VX Works RTOS used by the Siemens optiPoint 600. Avaya’s 4630 IP telephone was the first Web browser model with a color display and touch screen control. The use of color can greatly enhance the functionality and ergonomics of the desktop instrument, particularly when displaying graphic information or photographs. Touch screen control, instead of cursor control buttons, provides point-and-click mouselike activation of features and menu selection. A telephone with touch screen control is not new; industry veterans may recall the Northern Telecom M3000 digital telephone introduced in 1985.

General desktop applications using an integrated Web browser include:

• Access to directories external to the IP-PBX system directory database

• Messaging (voice, text, fax)

• Web page information screens

• Personal calendar

• Conference planning

• Transportation schedules and reservations

• Financial data (real-time stock quotes, investor information)

The accompany diagram of the Avaya 4630 IP telephone with a color touchscreen display illustrates the various applications supported by an IP telephone with an integrated Web browser interface.

Figure 1: Screenphone applications.

Figure 2: IP screenphone applications.

Figure 3: IP screenphone applications.

Figure 4: IP screenphone applications.

Using a telephone for e-mail or calendar access may seem strange if a personal computer is only inches away on the desktop, but it can be quicker and easier with the telephone. Telephones are always “on,” and information access is immediately available at a touch of a button. Booting up a desktop computer is getting longer and longer, as each release of Windows becomes more and more complex and the number of programs loading grows even larger. Many companies have several antivirus programs that run a series of system and memory checks before the computer is ready for use. The reliability level of a telephone has proved to be at least an order of magnitude greater than desktop computers, and it is less likely that the telephone will freeze due to program interactions or some other operating system glitch.

The Web browser feature can be especially useful in vertical markets where voice station users do not normally have a desktop computer. The healthcare, retail, and hospitality sectors are characterized by a significant number of stations users who have voice-only instruments at their disposal. For example, many nursing stations still have dumb CRT terminals for information access. In the retail sector, most point-of-sale (POS) terminals have no Web server access. In hotels, guest rooms have telephones, and Ethernet ports, but no computers. There is also a sizable number of installed telephones across all industry sectors with no nearby PC client. Many telephones are not located on a desktop shared by a computer: lobby telephones; cubicle telephones; conference room telephones; and wall-mounted telephones in hallways, cafeterias, or locker rooms. An IP telephone with a Web browser can be used as an information kiosk in public locations, such as shopping malls, bus terminals, or airports.

Mobile. There are three subcategories of mobile telephones for use behind a PBX system: cordless, premises wireless, and cellular. PBX cordless telephones can be proprietary or standard 2500-type analog. Proprietary cordless telephones are supported by proprietary PBX port circuit cards and have a unique signaling and control channel that allows for multiple line appearances and full PBX feature access and performance (including display-based information). Usually using spread spectrum technology and operating in the 900-MHz frequency range, a proprietary cordless telephone can often be used as a substitute for desktop models. A growing number of circuit switched PBX systems supports this option, including Avaya, Nortel, Siemens, NEC, and Toshiba. Analog cordless telephones, the same type commonly used for residential applications, appear to the PBX system as 2500-type telephones and offer limited feature/function access but a degree of station user mobility not offered by fully wired desktop models.

Premises wireless handsets are included as part of a premises wireless telephony option working behind the PBX system. The wireless handsets for these systems are proprietary to each system’s controller cards and base station transceivers. Base station coverage is limited in terms of geography and traffic handling. Most base stations support radio transmission ranges of about 50 to 150 meters, and between 2 and 12 simultaneous conversations per coverage cell. The wireless handsets closely resemble consumer cellular telephones, with several notable differences. Several manufacturers market wireless handsets with multiple line appearance buttons, fixed and programmable feature/function keys, and multiline displays that provide station users with information and data comparable to those of desktop digital telephone models. The high cost of a premises wireless handset and the infrastructure required to support coverage and traffic has limited the appeal of wireless telephony options, despite the ability of the station user to stay in touch with the PBX system regardless of location within the customer premises.

The first generation of premises wireless handsets was based on traditional circuit switching TDM/PCM standards. The recent introduction of wireless IP telephony solutions allows customers to use the existing LAN infrastructure to support distributed base stations. IP-PBX systems can interface directly to the wireless LAN infrastructure, but an MG is required for work behind a circuit switched PBX system. A leader in wireless IP is Symbol Technologies, whose Spectrum 24 wireless LAN system supports a wireless IP handset for use behind a PBX system. The Spectrum 24 uses spread spectrum frequency hopping within the 2.4- to 2.5-KHz band for transmission between access point transceivers and handheld communications devices. Data rates up to 2 Mbps per channel are supported. Each access point serves as an Ethernet bridge and can support wireless transmission coverage up to 2,000 feet in open environments and up to 180 to 250 feet in a typical office or retail store environment. Symbol’s NetVision Phone system provides enterprise voice communications capability and allows for integration into an existing PBX system (via a gateway) for premises and off-premises communications. The system includes NetVision Phones, access points, and a telecom gateway (third party). Each access point typically can support between 12 and 16 active clients and up to 10 voice-only conversations. There is a voice prioritization algorithm at the access point and client levels to minimize voice transmission delays. Fast roaming and load balancing support hand-offs between access points. Access point pinging detects and tracks station devices. The NetVision Phone is based on the ITU H.323 standard and converts analog voice signals into compressed digital packets (G.729/A 8-bit sampling rate, 160 bytes per packet) that are sent via the TCP/IP protocol over standard data LAN networks with the CSMA/CA wireless access protocol. TCP/IP addressing is used to tie to an extension number or a name directory. Several dialing mechanisms are supported:

• Direct entry of complete or partial IP addresses

• Direct entry of an “extension” number

• Speed dial operation via speed dial keys

• Recall/redial of a previous number

• Using a name directory internally mapped to an IP address

• Pressing the Send button begins the keypad dialing process

NetVision is a single line telephone, with a second “virtual line appearance” to support two concurrent conversations (one line is active and the other is in the hold mode). Intercom calls are supported between the phones over the LAN infrastructure, including broadcast capability to any number of phones. A multiline display field provides for incoming CLID services, and fixed function keys are used for one-button feature access. Symbol also offers a NetVision Dataphone for use with Spectrum 24. This telephone handset has an integrated Web client for accessing applications and databases and bar code scanning capability. Proprietary versions of NetVision telephones are used by Nortel Networks and Mitel Networks behind their IP-PBX systems. The NetVision IP wireless telephony system interfaces to the IP-PBXs via port interface gateway line cards. The accompany diagram illustrates the integration of the wireless NetVision handsets into an Ericsson MD-110 PBX configuration (Figure 5). The NetVision terminals are typical of IP wireless handsets that are designed for enterprise mobile applications.

Figure 5: Virtual IP telephony extensions.

Premises cellular is the third mobility communications option. The same cellular handset used with network cellular services, such as Sprint PCS, AT&T Wireless, and Cingular, can also interwork with a PBX system for premises mobile communications requirements. The first premises cellular options required an on-site mobility server and cell transceiver that linked to a local carrier’s network. The mobility server provided an interface between the PBX system and the premises cellular infrastructure to support control signaling and feature support to cellular handsets while the station user was on the customer premises. This mobile communications option had several drawbacks, including cost (mobility servers and transceivers are expensive for limited numbers of subscribers) and network compatibility. The premises transceiver could link to only one cellular carrier service, such as TDMA or GSM. All premises subscribers required a cellular handset that worked with the same network carrier service. Although some business customers supplied their employees with a cellular handset and had a low-cost contract with a single service provider, the more likely scenario was that PBX station users had a great variety of cellular handsets supported by different network service carriers. A better solution was needed than an expensive cellular infrastructure linked to a single service provider.

Ericsson, a leader in mobile communications networks, developed a more cost-effective and flexible premises cellular option. The MD-110 Mobility Extension option is based on an integrated interface circuit card housed in the PBX’s port carrier that can support a cellular handset with the use of any type of service standard from any local carrier. An ISDN PRI trunk circuit link is used to network the PBX system to the cellular network. Dialing procedures from the cellular handset will be in line with the terminal’s existing network service procedures, plus fully support the MD110-procedures, including station features (via voice prompts) and network call routing. The Ericsson Mobility Extension option is carrier service provider and transmission/encoding independent.

PC client softphone. The final category of PBX telephones is the PC client softphone. There are several categories of softphones. The first generation of softphones was based on CTI desktop applications using first-party (desktop telephone API link) or third-party (client/server configuration) call control. The CTI-based softphone requires a telephone instrument (analog or digital) for voice transmission to/from the desktop. An IP softphone is a PC client functioning as the voice terminal using an integrated microphone/speaker option to support LAN-based voice transmission, with signaling and control to and from a telephony server over the LAN/WAN infrastructure. For implementation of either softphone, a station user accesses and implements PBX features (dialing, call answering, call coverage, call processing) using a keyboard and/or mouse control for a GUI computer screen. Communications solutions using PC client software tools offer station users many advantages over traditional telephone instruments, with a limited number of feature/line keys and relatively small noncolor display fields. The accompanying diagram is an illustration of the Nortel Networks i2050 soft client phone (Figure 6). Some suppliers also offer customized client keyboards with integrated handsets for use as a softphone. The accompanying photograph is a Siemens optiKeyboard designed for use with its family of softphone client solutions (Figure 7)

Figure 6: IP softphone: Nortel Networks i2050.

Figure 7: Siemens optiKeyboard.

Market demand for CTI-based softphones has been very weak. Many station users prefer to depress traditional telephone buttons to access features rather than interact with a GUI-based computer screen to perform drag, point, and click operations. Telephone instruments also offer a far greater degree of reliability than PC hardware/software and are not affected to the same level as AC-powered desktop computers by local power problems. A major problem associated with first-party control CTI softphones was the requirement of a relatively expensive digital telephone equipped with an API link to the desktop computer. Third-party control client/server CTI configurations could be implemented with a lower-priced analog telephone, but station user functionality is severely affected when the desktop computer fails or is not performing properly. The primary market for desktop CTI has been among call center customers because the current ACD agent position depends heavily on desktop computer equipment and GUI-based interactions, and the cost of the solution is not significant compared with overall contact center expenses.

The emergence of IP-PBX systems may spur demand for PC client softphones because the cost of the solution may be far less than that of a high performance IP telephone. There likely will be great resistance to IP softphones from most station users who have grown comfortable with traditional telephone instruments, but the many potential benefits of the new solution may stimulate market demand.