OXE Gateway

Entity between SIP world and legacy world, the gateway is used to establish a call from a SIP equipment to an ISDN link, to a legacy set, etc… and vice versa

• Do not confuse the SIP gateway with the OmniPCX Enterprise media gateway boards:
• The SIP gateway is a logical entity that resides within the call server (CS) and is responsible for the SIP signaling for the conversation setup,
• The media gateway boards (GD, GA, INTIP) are the physical devices where the media session will be established when calling to a classic PBX set.

• There is one and only one internal SIP gateway. But there can be many different external SIP gateways (we will come back to this in a later section).

• The SIP gateway is associated to a SIP trunk group. Although there can be many SIP Trunk Groups, there is only one SIP trunk group which is associated to the local SIP gateway. We call this special trunk group the local SIP trunk group.

OXE Dictionnary

Contains the SIP users created on the OXE, it is the database that holds the mapping between SIP URLs and PBX directory numbers (MCDUs). Each registered SIP terminal is automatically added to the dictionnary. Classic PBX terminals are added only if a SIP URL is defined for them in the user management.

• Most of the time you shouldn’t do anything with the Dictionnary. Everything will be handled automatically. You need to access the SIP Dictionnary configuration only for configuration of aliases

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.

Telephony Gateways | IP PBX System Design

IP telephones, including PC client softphones, communicate directly with the call telephony server over a customer LAN/WAN infrastructure. Proprietary port circuit cards housed in proprietary port carriers are not required for signaling between the IP desktop and the common control controller, unlike converged IP-PBX designs. Non-IP stations and trunk circuits require telephony gateway interfaces to support server control signaling and voice communication transmissions. Telephony gateways for analog telephones and other 2500-type compatible communications devices, such as facsimile terminals, may be provided through a variety of design methods:

• Integrated call telephony server gateway interfaces

• Desktop gateway modules: proprietary, third party

• Gateway servers/interfaces: proprietary, third party

Several proprietary, closed call telephony servers have integrated gateway interfaces for PSTN digital T1/E1 trunk circuits. The gateway interfaces usually support ISDN BRI or PRI services over the T1/E1 trunk circuits. The Mitel MN 3100, 3Com NBX, and Siemens HiPath 5300 systems have integrated PSTN digital trunk gateway interfaces. For example, the 3Com NBX’s integrated analog line card connects up to four conventional (loop start) PSTN telephone lines, and the T1/PRI trunk card connects to a standard T1 circuit. The HiPath 5300 BRI gateway interface card supports four BRI ports (8 × 64-Kbps channels); the PRI gateway interface card includes a T1 carrier interface.

Mitel Networks uses a different approach to support non-IP peripherals on its MN3300 ICP. The 3300 ICP includes an analog services unit (ASU), and a network services unit (NSU), but also supports traditional Mitel SuperSet digital telephones by a link to a peripheral equipment (PE) cabi- net. An ASU supports four analog trunks and 16 stations (including MOH, paging, and PFT); an NSU supports four T1 digital trunk interfaces. Up to four ASU and four NSU carriers are supported per controller carrier. What is unique about the system is that the call server also provides control signaling to an SX-2000 Light PE cabinet. Supporting the traditional PE cabinet protects a customer’s substantial investment in the installed base of proprietary Mitel SuperSet voice terminals.

Mitel intends the MN 3300 system to be a migration vehicle for its large installed base of SX-2000 system customers and allows customers to link existing PE cabinets to the new call telephony server through one of two options: direct optical fiber cable connection or T1 trunk interface. Customers who want a centrally located call telephony server and PE cabinet can use the optical fiber link. The DTI can support remote PE cabinets. The 3300 ICP was the first client/server IP-PBX design to support common equipment originally designed for a circuit switched PBX system. All communications traffic between digital telephones is handled internally by the PE cabinet’s integrated circuit switched TDM backplane. Calls between PE endpoints and other endpoints (IP telephones and ASU and NSU ports) are handled across the integrated controller gateway channels.

Desktop gateway modules may be proprietary or industry-standard H.323 equipment. The most common desktop gateways support 2500-type communications devices, such as analog DTMF telephones and facsimile terminals. The desktop communications device links directly to the gateway module and converts analog signals to IP format for control and communications signaling. For example, 3Com NBX analog devices are available as single-port stand-alone units and four-port chassis-based cards. The single-port ATA unit also includes an additional Ethernet port that allows an analog device and an Ethernet device to share the same Ethernet LAN cabling. The multiple-port NBX analog terminal card features four analog (FXS) ports. The units connect to a wide variety of industry-standard analog devices and fax machines and provide support for door phones, paging systems, and other applications that may require analog connectivity.

The gateways may be proprietary to an IP-PBX system, like the Siemens HiPath AP 1100 (available in one- and four-port interface models), or third-party products available from a large list of suppliers. For example, Ericsson markets a downsized version of its Webswitch IP-PBX for use as an H.323 gateway module. 3Com, a major enterprise data communications equipment supplier, is another IP-PBX supplier marketing desktop gateway modules, including those that support H.323 and SIP standards.

Another type of desktop gateway module is an add-on adapter that converts a proprietary digital PBX telephone into an IP-compatible voice terminal. A few client/server IP-PBX manufacturers, including Siemens and Nortel Networks, offer this as an option to upgrade installed digital telephones originally designed for use behind their circuit switched PBXs. The same adapters can support IP desktops behind the manufacturer’s converged IP-PBX system solutions.

Gateway servers and interface modules/boards that are not fully integrated into the call telephony server or used as desktop devices are proprietary to a manufacturer’s IP-PBX or conform to industry standards, such as H.323 or MGCP, and used as OEM solutions. One example of a proprietary solution is the Mitel Networks MN 3300 ICP gateway carrier that supports traditional analog trunks (loop start) and digital trunks (DASSII, DPNSS, QSig, Euro ISDN, and BRI) for connection to the PSTN and for connecting multiple sites or systems together. This allows multiple 3300 ICPs to be clustered or networked between multiple sites over IP or traditional TDM infrastructures to support up to 40,000 users. The MN 3300’s call telephony server carrier supports the trunk gateway interface carrier.

The Cisco Systems IP Telephony system, when originally designed as the Selsius System, used desktop modules for support of non-IP communications devices and trunk circuits. The redesigned product supports analog station, analog trunk, and digital trunk interfaces with proprietary circuit boards that are housed in Cisco Catalyst 6000 Ethernet switch carriers. Three different modules are used for analog connections: 24-port analog station FXS (H.323 or MGCP), analog trunk circuit FXO (H.323 or MGCP), and analog E&M tie trunk (H.323 only). The FXS module supports fax relay, which enables compressed fax transmission over the IP WAN. An alternative to the FXS module is the standalone Cisco VG248 analog gateway module that supports 48 fully featured analog phone lines as extensions to the Cisco CallManager system. It is housed in a compact 19-inch rack-mount chassis, and its high-density gateway can be used for analog phones, fax machines, modems, and speaker phones. Digital PSTN trunk interfaces are supported by a limited-capacity stand-alone T1 adapter module or a Catalyst 6000 T1 and services module that provides eight T1 ports (192 channels) or DS0 voice trunks. The module supports voice trunk protocols such as ISDN Primary Rate Interface (PRI) and in H2 CY ‘00, channel-associated signaling (CAS). The module’s DSP resources can also be programmed for call conference bridge services and voice codec transcoding applications, instead of digital trunk gateway interfaces.

The Nortel Succession CSE 1000 MG module supports a variety of non-IP interfaces, such as analog station, analog trunk, and digital trunk. Each MG module has three IPE card slots and can support an expansion module for four additional slots. The first Succession CSE 1000 release is limited to a maximum of four MGs (28 card slots, maximum).