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 the discussions in this article.
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.
1. 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.
2. 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.
3. 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.
Table 1 lists the types of telephony systems and explains when a VoIP gateway is required.
Table 1. VoIP Gateway Requirements for Telephone Systems
TYPES OF TELEPHONE SYSTEM | VOIP GATEWAY REQUIREMENT |
---|
Traditional Centrex | Required |
IP Centrex | May not be required if supported |
Key Telephone System | Required, some phone systems are not supported |
Analog or Digital PBX | Required |
IP or Hybrid PBX | May not be required is supported |
A list of supported PBXs and IP gateways for Exchange 2010 Unified Messaging can be found on Microsoft TechNet in Telephony Advisor for Exchange 2010 at http://technet.microsoft.com/en-us/library/ee364753.aspx.
4. 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.