1. Supported IP/VoIP Hardware
Exchange Server 2010 unified messaging relies on the
capability of the IP/VoIP gateway to translate time-division
multiplexing (TDM) or telephony circuit-switched based protocols, such
as Integrated Services Digital Network (ISDN) or QSIG, from a PBX to
protocols based on voice over IP (VoIP) or IP, such as Session
Initiation Protocol (SIP), Real-Time Transport Protocol (RTP), or T.38
for real-time facsimile transport.
Although there are many types and manufacturers of
PBXs, IP/VoIP gateways, and IP/PBXs, there are essentially two types of
IP/VoIP gateway component configurations:
IP/VoIP Gateway—
A legacy PBX and an IP/VoIP gateway provisioned as two separate
devices. The Unified Messaging server communicates with the IP/VoIP
gateway.
IP/PBX— A modern IP-based or hybrid PBX such as a Cisco CallManager. The Unified Messaging server communicates directly with the PBX.
Table 1 lists the currently supported IP/VoIP gateways.
Table 1. Supported IP/VoIP Gateways for Exchange 2010 UM
| Manufacturer | Model | Supported Protocols |
|---|
| AudioCodes | MediaPack 114,
MediaPack 118 | Analog with In-Band or SMDI |
| AudioCodes | Mediant 1000/2000 | -T1/ or E1 with CAS—In-Band or SMDI, T1/E1 with Primary Rate Interface (PRI) and Q.SIG or Analog PSTN |
| Dialogic | 1000/2000 | -T1/ or E1 with CAS—In-Band or SMDI, T1/E1 with Primary Rate Interface (PRI) and Q.SIG or Analog PSTN |
| Ferrari AG | OfficeMaster 3.2 | PSTN Analog |
| Net | VX1200 | -T1/ or E1 with CAS—In-Band or SMDI, T1/E1 with Primary Rate Interface (PRI) and Q.SIG or |
| Nortel | CS1000 | Direct SIP |
| Quintum | Tenor-series | Analog PSTN |
To
support Exchange Server 2010 unified messaging, one or both types of
IP/VoIP device configurations are used when connecting a telephony
network infrastructure to a data network infrastructure.
All these solutions must communicate with the unified messenger through SIP over TCP (TLS encrypted) and SRTP.
2. Unified Messaging Protocols
The Exchange 2010 Unified Messaging servers use
several telephony-related protocols to integrate and communicate with
telephony devices. These protocols are listed and discussed in the
following list:
Session Initiation Protocol (SIP)—
This is the signaling protocol that is used to set up and tear down
VoIP calls. These calls include voice, video, instant messaging, and a
variety of other services. The SIP protocol is specified in RFC 3261
produced by the Internet Engineering Task Force (IETF) SIP Working
Group. SIP is only a signaling protocol and does not transmit data.
After the call is set up, the actual communications take place using
the RTP for voice and video or T.38 for faxes.
Note
Exchange 2010 only supports SIP over TCP. SIP, in
general, can be configured to run over User Datagram Protocol (UDP) or
Transmission Control Protocol (TCP). UDP is connectionless and does not
provide reliability guarantees over the network. TCP is
connection-oriented and provides reliability guarantees for its
packets. Exchange 2010 UM supports either SIP over TCP, SIP over TLS,
or Dual where both are supported simultaneously.
Real-Time Transport Protocol (RTP)— This
protocol sends the voice and video data over the TCP/IP network. The
protocol relies on other protocols, such as SIP or H.323, to perform
call setup and teardown. It was developed by the IETF Audio-Video
Transport Working Group and is specified in RFC 3550. There is not a
defined port for the RTP protocol, but it is normally configured to use
ports 16384–32767. The protocol uses a dynamic port range, so it is not
ideally suited to traversing firewalls.
Real-Time Facsimile Transport (T.38)—
This protocol is an International Telecommunication Union (ITU)
standard for transmitting faxes over TCP/IP. The protocol is described
in RFC 3362. Although it can support call setup and teardown, it is
normally used in conjunction with a signaling protocol such as SIP.
It is important to note that the Exchange 2010
Unified Messaging server is also a Windows server, a web server, and a
member of the Active Directory domain. There are a myriad of protocols,
including domain name system (DNS), Hypertext Transfer Protocol (HTTP),
Lightweight Directory Access Protocol (LDAP), remote procedure calls
(RPCs), and Simple Mail Transfer Protocol (SMTP), among others, that
the server uses to communicate with other servers in addition to the
telephony communications.
Unified Messaging Port Assignments
Table 2
shows the IP ports that unified messaging uses for each protocol. The
table also shows whether the ports can be changed and where.
Table 2. Ports Used for Unified Messaging Protocols
| Protocol | TCP Port | UDP Port | Can Ports Be Changed? |
|---|
| SIP-UM Service | 5060 | | Ports are hard-coded. |
| SIP-Worker Process | 5061 and 5062 | | Ports are set by using the Extensible Markup Language (XML) configuration file. |
| RTP | | Port range above 1024 | The range of ports can be changed in the Registry. |
| T.38 | | Dynamic port above 1024 | Ports are defined by the system. |
| UM Web Service | Dynamic port above 1024 | | Ports are defined by the system. |
