The Basics of Telephony

Because Unified Messaging must be integrated into your company's telephony solution, it's important to understand the most crucial terms and definitions to be able to follow.

Note

If your company is already connected to Office Communications Server 2007 or later with your telephone system, you don't need to consider the details in the following sections; Exchange 2010 will use OCS as the gateway.

Types of Telephone Systems

Three general types of business telephone systems can be integrated with Unified Messaging:

• Centrex Phone System Phone companies lease a Centrex phone system (also known as Central Office Telephone Exchange) to businesses. The Centrex phone system uses the phone company's central office (CO) exchange to route internal calls to an extension. A new Centrex version called IP Centrex is available. With IP Centrex, the organization does not rent phone lines from the telephone company's CO. Instead, the CO sends the phone calls through a VoIP gateway, which routes them over a VoIP gateway or through the Internet. At the organization's office, another VoIP gateway translates the call to a traditional circuit-switched call.
  
  
• Key Telephone System This phone system is similar to the Centrex system in that the organization leases several phone lines from the telephone company. However, with the Key Telephone System, each phone line connects to multiple telephones in the organization. When someone calls the company, all phones ring that are associated with that line. Businesses with Key Telephone Systems often arrange for someone to answer incoming calls, and then announce the call to the correct recipient.

  Note

  Some key telephone systems can work with UM if an IP gateway is added. However, some less sophisticated systems may not work even if a supported IP gateway is used. Make sure you contact your vendor before you try to use your key telephone system with Exchange 2010.

• Private Branch Exchange System A Private Branch Exchange (PBX) system is different from the other telephone systems in that it typically has only a single connection to the phone company and all call switching happens at the organization. The connection to the phone company usually occurs through a T1 or E1 line, both of which provide multiple channels to enable multiple calls over the same line, also called trunk lines. The PBX routes internal phone calls and those between external and internal users. In a PBX system, each user has a telephone extension. When an internal user places a call to another internal user, she uses only the extension number, and the PBX routes the call to the appropriate extension.

Types of PBX

PBX systems are the most common telephone system type that medium- and large-size organizations use. Several types of PBX systems are available:

• Analog PBX Analog PBX systems send voice and signaling information, such as the touch tones of dialed phone numbers, as actual analog sound. Analog PBX systems never digitize the sound. To direct the call, the PBX and the phone company's CO listens for the signaling information.
  
  
• Digital PBX Digital PBXs encode analog sound into a digital format. They typically encode the voice using a standard industry audio codec, G.711. After digital PBXs encode the sound, they send the digitized voice on a channel using circuit switching. The process of circuit switching establishes an end-to-end open connection, and leaves the channel open for the call's duration and for the call's users only. Some PBX manufacturers have proprietary signaling methods for call setup, such as Avaya Definity G3si PBX.
  
  
• IP PBX IP PBXs include a Network Interface Card (NIC) to provide voice over regular network. The phone converts voice into digitized packets, which it then transfers over the network. The network sends the voice packets via packet switching, a technique that enables a single network channel to handle multiple calls. The IP PBX also acts as a gateway between the internal packet-switched network and the external circuit-switched networks that phone company's use. In this situation, external phone calls arrive at the IP PBX on the normal public phone lines, and the IP PBX converts the phone call to packets sent on the internal IP-based network. An example of this is Cisco Call Manager.
  
  
• Hybrid PBX Hybrid PBXs provide both digital and IP PBX capabilities. This hybrid approach enables a customer to run a mixture of digital and IP-based phones. Most modern PBXs are in this hybrid category, such as SEN HiPath 4000.

VoIP Gateway Introduction

A VoIP gateway is a third-party hardware device or product that converts traditional phone-system or circuit-switching protocols into data-networking or packet-switched protocols. The VoIP gateway connects a telephone network with a data network.

Unified Messaging servers can connect only to packet-switched data networks. This means that organizations with a traditional PBX must deploy a VoIP gateway to communicate between the PBX and the Unified Messaging server.

Unified Messaging Protocols

There are a number of voice-related, IP-based protocols. A Unified Messaging environment with Exchange Server 2010 uses the following:

• Session Initiation Protocol (SIP) SIP is a real-time signaling protocol that creates, manipulates, and disconnects interactive communication sessions on an IP network. The UM role uses SIP mapped over Transmission Control Protocol (TCP) and supports TLS for secured SIP environments. SIP clients, such as IP/VoIP gateways and IP/PBXs, can use TCP port 5060 or port 5061 (for Secure SIP) to connect to UM server roles. You can find more information about the SIP protocol at http://tools.ietf.org/html/rfc3261.
  
  
• Real-time Transport Protocol (RTP ) RTP is for voice transport between the IP gateway and the Unified Messaging server. RTP provides high-quality, real-time, streaming voice delivery. One of the issues with sending voice messages over an IP network is that voice requires real-time transport with specific quality requirements to ensure that the voice sounds normal. If the protocol uses large packets, listeners must wait for the entire packet to arrive before they can respond. Any delay in packet delivery can produce undesirable periods of midstream silence. Packet loss can cause voice garbling. You can get more information about the RTP protocol at http://tools.ietf.org/html/rfc3550.