Book Image

Building Enterprise Ready Telephony Systems with sipXecs 4.0

Book Image

Building Enterprise Ready Telephony Systems with sipXecs 4.0

Overview of this book

Open source telephony systems are making big waves in the communications industry. Moving your organization from a lab environment to production system can seem like a daunting and inherently risky proposition. Building Enterprise Ready Telephony Systems with sipXecs delivers proven techniques for deploying reliable and robust communications systems. Building Enterprise Ready Telephony Systems with sipXecs provides a guiding hand in planning, building and migrating a corporate communications system to the open source sipXecs SIP PBX platform. Following this step-by-step guide makes normally complex tasks, such as migrating your existing communication system to VOIP and deploying phones, easy. Imagine how good you'll feel when you have a complete, enterprise ready telephony system at work in your business. Planning a communications system for any size of network can seem an overwhelmingly complicated task. Deploying a robust and reliable communications system may seem even harder. This book will start by helping you understand the nuts and bolts of a Voice over IP Telephony system. The base knowledge gained is then built upon with system design and product selection. Soon you will be able to implement, utilize and maintain a communications system with sipXecs. Many screen-shots and diagrams help to illustrate and make simple what can otherwise be a complex undertaking. It's easy to build an enterprise ready telephony system when you follow this helpful, straightforward guide.
Table of Contents (17 chapters)
Building Enterprise-Ready Telephony Systems with sipXecs 4.0
Credits
About the Author
About the Reviewer
Preface
Glossary

Chapter 1. Introduction to Telephony Concepts and sipXecs

In this chapter we'll introduce some important telephony concepts to establish some necessary background information. Then we'll move on to an overview of sipX Enterprise Communications Server (sipXecs), its features, and its functionality.

Traditional phone system concepts

There are two types of traditional phone systems, PBXs and Key Systems. A Private Branch Exchange (PBX) is typically found in larger organizations. Key telephone systems that allow users to directly select outside lines via keys on the handsets were designed with smaller organizations in mind. Both types of systems typically consist of interfaces to a telecommunications provider, interfaces to telephone handsets, a voicemail system for auto attendant and leaving messages, and call-routing logic.

The traditional PBX is usually thought of as being housed in a cabinet with various interfaces and logic boards inserted as cards into a backplane across which all of the cards can communicate. These backplanes are vendor specific, so you are typically locked in to purchase all cards and phones from a single vendor. Additionally, many first-generation IP-based phone systems may also be thought of as traditional systems. These early IP systems use proprietary signaling over IP or protocols that have fallen out of favor (MGCP/H.323).

The PBX communicates with the outside world from the interface to a telecommunications provider. In a traditional PBX, this interface is typically some sort of analog circuit (loop-start or ground-start) or digital circuit (E1/T1, ISDN, or Primary Rate Interface [ PRI]).

The telephone set interface is how the PBX connects with the various user devices that it is in direct control of. This is traditionally an analog interface to a limited-feature phone (like a typical home telephone) or a digital interface to a more feature-rich phone.

Voicemail systems in the traditional PBX are designed to handle recording and playback of messages to users of the system, notifying the users they have messages via a Message Waiting Indicator ( MWI), and also automated attendant duties. The automated attendant's function is to answer inbound phone calls, play a message, and wait for a caller to enter an option or extension.

The call-routing logic in a phone system determines where calls route to, based on a number that was dialed (be that an extension on the system or an external phone number). Other factors may also come in to play with call routing such as permissions, time of day, what line a call came from, and so on.

Telecommunications provider interface

The interface to a traditional telecommunications provider (a phone company) can take different forms depending on how your calls are being delivered. If your calls are being delivered by a traditional provider over E1, T1, PRI, BRI, or analog line, this interface device is a hardware-based gateway.

E1s, T1s, and PRIs are all digital circuits that can carry multiple conversations. E1 is a physical layer protocol, much like Ethernet, that defines a 2Mbps pipe. This pipe can be used for data—split into 32 64Kbps communications channels—or a mixture. If the pipe is used for communications channels, 30 of the channels can carry telephone conversations and the remaining 2 carry signaling and timing information.

A T1 is similar to an E1, and it is common in North America. T1s are 1.544Mbps pipes that can carry 24 telephone channels. There are no signaling channels on a T1. Also, like an E1, T1s can be channelized and utilized to deliver voice and data.

E1 and T1 circuits have some problems associated with them. They are limited in what information they can carry and the circuits are relatively slow to set up. ISDN signaling is a more modern protocol that was designed to overcome these problems. On E1s, EuroISDN signaling is standard. On T1s, different providers utilize different standards. NI1, NI2, DMS100, and DMS250 are all examples of ISDN signaling protocols, each delivering different levels of functionality.

A PRI (Primary Rate ISDN) is an E1 or T1 with ISDN signaling running on top of it. ISDN signaling provides reliable call setup and tear-down detection, as well as detailed information about each call. In the UK, a PRI is also referred to as ISDN30. Voice channels on a PRI are referred to as B channels and the signaling channels are referred to as D channels. On an E1, a PRI will provide 30 B channels of voice and utilize one of the signaling channels as the D channel. Since T1s have no signaling channels, a PRI on a T1 will utilize one of the channels as a D channel and have 23 B channels for voice.

As a cheaper alternative to PRI,BRI (Basic Rate ISDN) may be offered in some areas. A BRI has 2 64Kbps B channels and a single 16Kbps D channel for signaling. In the UK, a BRI may also be called ISDN2e.

Analog lines from local telephone companies come in a couple of different flavors, both delivered over a pair of copper wires. They will be referred to as Ground Start Trunks (GST) or loop start circuits. Ground start circuits provide disconnect notification by actually grounding the circuit (when a caller hangs up the phone), which is also called answer and disconnect supervision. Loop start analog circuits are the more typical home and key system phone lines. Loop start lines use either a polarity reversal (called battery reversal), or removal of the line voltage (battery drop) for answer and disconnect supervision.

Telephones on a traditional phone system

Telephone sets on a traditional phone system will interface to the system by using one of the one of three methods: analog, digital, or via IP.

Analog phones are usually the same sort of phones you might find in a residence. They can provide signaling to the PBX for special functionality by flashing the hook switch and utilizing different DTMF codes. Dial Tone Multi Frequency is the sounds you hear when you push the dial pad buttons of a phone. Analog phones need to be manually configured as there is no means for passing codes down to phones and programming any special keys that may be on the phones.

Digital phone sets provide higher functionality and programmability for phone systems. They are proprietary to each vendor and type of phone system. Digital sets can be programmed centrally. They provide excellent call quality and usually have many buttons that can be programmed to provide different functionality to the user. The majority of phones shipped with phone systems were digital until 2005/2006 when IP phone sets surpassed them in total numbers shipped.

Many traditional phone systems vendors have seen the advantages of an IP-based system and have adapted their phone systems to support IP-based phones. A traditional phone system that has been adapted to support a mix of phones is referred to as a hybrid system. What we'll refer to as first-generation IP-based phone systems utilize a proprietary protocol for communications, or one of the older voice standards. Examples of proprietary protocols are SCCP (Cisco), UNIStim (Nortel), and MiNet (Mitel). As with digital phones, proprietary protocols require vendor-specific phones. Session Initiated Protocol (SIP), H.323, and MGCP are examples of standards-based protocols. Phones that conform to standards are designed to work on many different phone systems.

Voicemail systems

Voicemail systems are an important part of any business phone system. These systems provide auto attendant functions, and the playing and recording of messages. The voicemail system can be thought of as the voice of the phone system.

When calling into a phone system, the caller will hear the main auto attendant, which provides the caller with a menu of choices. The auto attendant plays a recorded message and waits for the caller to enter DTMF tones selecting a menu option or dialing an extension. Newer advanced auto attendant systems have grown to include voice recognition for menu items or extension selection.

The voicemail system also handles the recording and playback of user greetings and voicemail messages. Many modern voicemail systems allow multiple greetings to be selected by the user for out-of-office or extended-leave situations so that the user doesn't need to keep re-recording his or her notifications.

Unified messaging systems are an extension of voicemail systems that allow users to have a single inbox combining voicemail, email, and faxes. A true unified system will integrate these systems at the server level such that when you open or delete voicemail on a computer, it is marked as read or deleted in the voicemail system. A simple version of unified communications involves SMTP forwarding of voicemail to an email, or requires a setup of client software that handles email integration on the user's computer.

Traditional voicemail systems are usually sold to customers with support for a certain number of ports. The ports control how many simultaneous voice sessions can occur between the phone system and the voicemail system. The system may be contained on a card in the system or on a separate server outside the phone system cabinet.

An important but seemingly simple responsibility of the voicemail system is to signify to users that they have messages waiting. This notification usually takes the form of a Message Waiting Indicator (MWI) light that is lit on handsets.

Call routing logic

The "brains of the operation" in the traditional phone system is the call routing logic. The routing logic is called different things by different vendors, but may be referred to as the call controller or call manager. Its job is to evaluate calls and direct them (referred to as switching) to where they need to go based on many different factors. These factors include, but are not limited to, what number was dialed, who dialed it, and what time of day it is.

Calling functions and features

There are hundreds of call routing functions and phone system features that have been developed over the years. The following are some of the more common call functions and features.

Call hold

With call hold, the user presses a button on his or her phone that places a caller into a mode such that neither party can hear each other. Often, music or an announcement is played while the party is on hold (Music on Hold, or MoH). In small key systems, users on other phones can pick up on a line that has been placed on hold. With PBXs, the call is usually retrieved on the same phone that the call was put on hold with.

Call park orbits

Call park orbits were designed for PBX systems where the concepts of phone lines to users don't exist. Putting a call into a park orbit is accomplished by transferring a call to a holding queue (orbit). That call can be retrieved on any phone by dialing a retrieval (also referred to as a pickup) code and the park orbit number.

Call pickup

Call pickup is the ability of one user to pick up another user's ringing phone. Often, permissions are required to do this function. This feature is typically accomplished by dialing a pickup code and the extension of the ringing phone.

Call transfer

Call transfer is the ability of a user to send a phone call to another extension on the phone system. There are two types of transfer: consultative and blind. In a consultative (also referred to as attended or supervised) transfer, the calling party confers with the party that it will transfer the call to before the call is transferred. In a blind (also referred to as unattended) transfer, the call is simply transferred to the selected extension.

Call forwarding

Call forwarding is a service that allows a user (or the phone system) to have a call redirected to another extension or number. The forwarding decision can be a strict choice to always forward, or it could be based on certain criteria such as whether the called party is busy, who is calling, time of day, and so on. Time of day forwarding is also referred to as "Time-based Follow-me/Find-me".

Speed dial

Speed dials in a traditional PBX are phone numbers that can be dialed in order to dial a more complicated number. For example, a user would dial 752 and the phone system would actually dial 18005555555. Most systems allow user-specified as well as system-wide speed dials.

Direct Station Selection/Busy Lamp Field

Direct Station Selection (DSS) can be thought of as a one-touch speed dial assigned to a key on a user's telephone. The user presses the button and the number assigned to the button is automatically dialed. When combined with information about an extension on the receiving end of the DSS, the feature is referred to as a Busy Lamp Field ( BLF, DSS/BLF or Presence). If the remote party is on the phone, a BLF will usually have a solid light on or near the button. If the remote party's phone is in a "Do Not Disturb" mode, (the phone rejects all calls) the light may blink.

Hunt groups

A hunt group is a collection of extensions that ring in a particular order when the hunt group number is dialed. The hunt group number is often referred to as the pilot number of the hunt group. Linear hunt groups always start ringing the first extension in the list and end ringing the last extension in the list. With a circular hunt group, the phone system remembers the last number that answered ringing and begins ringing on the next number in the list and when the end of the list is reached, it wraps around to the first number in the list again.

Automatic Call Distribution

Automatic Call Distribution (ACD) can be thought of as intelligent hunt groups. They allow phone system users (agents) to sign in and out of calling queues. Calls then ring agents based on different factors such as who is the first person in the ACD list, or which agent has been idle the longest. The ACD systems also allow other niceties such as wrap up time for agents after a call is completed.

Dial plans

The system dial plans provide the routing logic for inbound calls and outbound calls from the system. The dial plans evaluate the dialed numbers by looking for patterns of digits and directing calls to different destinations. It is up to the phone system designer to set up their dial plans based on their phone providers and the phone numbers they know their users will need to dial.

Intercom

The intercom function in a phone system allows a single user to dial another user's extension, makes the receiving user's phone automatically go "off-hook" in speaker phone mode, and allows the two parties to converse.

Paging

Paging is similar to intercom functionality, but it differs in one way. It is designed to allow a single user to broadcast a message to a group of other phones without the ability of the receiving phones to talk back to the caller.

Conferencing

A conference is a call between three or more parties. A conference may be a simple phone-based multi-party conversation, or may be hosted by a full-featured conference server. A simple phone-based conference requires a phone user to call multiple parties and establish the conference call. A conference server allows more parties, achieves finer-grained control by a conference moderator, and allows participants to come and go as they choose. A conference server will host many "rooms" where participants can meet. These conference rooms are often referred to as "Meet-Me" conferences.