Gateways are going away.
Think about it. VoIP gateways bidirectionally transform RTP packet-streams into bitwise digital TDM or analog voice. If you're completing calls IP-end-to-end, you don't need one.
Even if the PSTN refuses to go away, you won't need gateways at your premise much longer. Gateway functionality is already moving up into the network (e.g., in the hands of "hosted PBX" outfits like PingTone); and the process will continue (and offerings become more generic) as CPE IP PBXs become more ubiquitous, and as bottleneck issues (local-number portability, E-911 compliance, data-link powerfail protection, etc.) get worked out. Carriers will host gateways first in behalf of the contact center market, and then for all of us. And then even this centralized service will gradually wither, as ENUM (or whatever) emerges as a uniform standard for public global IP call-direction. Like the $10,000 channel-bank, the classic premise VoIP Gateway is indubitably headed for the tar pits.
Stop laughing! Okay ... you're right. It's going to be a while. And in the meantime, VoIP gateways - extinction-bound though they may be - are foaming up into a rich and varied "Jurassic climax ecology." They're everywhere. And they're more important than ever before.
In-skin gateways, in form-factors mirroring digital station boards and interoffice trunk cards, are now available for pretty much every contemporary digital PBX. They're used for hooking IP stations into the TDM switch, and doing IP interoffice networking via the WAN.
Single-line gateways, most compatible with PBX proprietary 2B+D digital station protocols, work on the station side to turn legacy digital ports and phones into "anywhere" remote extensions.
Stand-alone multiport gateways provide IP trunking (and selective IP station support) for legacy TDM switches. In next-gen distributed IP PBX architectures, the gateway is a real Swiss-Army knife. It provides access to conventional telecom speeds and feeds at head and branch offices - essential for local and WAN-mediated off-net call completion (read: 'basic phoning and interoffice toll bypass'), and crucial for backup, if IP WAN connections are severed. Station-side, gateways are being used to "hide" new IP-at-core switch engines from legacy digital station hardware and trunk/station integrated applications (or at least those apps, like specialized call recording, that are worth preserving as opposed to replacing with apps that come with the new PBX).
A plethora of access devices of every description, all with gateway functionality. IADs, set-top boxes, routers with gateway cards, etc. Even IP telephones are beginning to sport gateway ports for hooking up fax machines and other legacy equipment - successors to the analog option ports popular on high-quality digital stations.
HOW TO BUY A VOIP GATEWAY
"First, take out your credit card ..." In fact, buying an enterprise gateway has become pretty easy. A few years back, the woods were full of first-generation products with significant flaws - bad codec implementations, iffy protocol support, weird echo-cancellation, peculiar comfort-noise implementations, etc. For a while, quite a few products carried DTMF as audio payload, distorting it to the point where distant voicemail and IVR systems got confused.
Today, those problems are largely history. Time, experience, and availability of standard chipsets and DSP algorithms have stabilized the technology and gotten rid of egregious weirdness (it's hard to find a gateway today, for example, that doesn't near-end-detect, transcode, and far-end-regenerate DTMF). That's lucky, because as IP PBXs become more mature, more and more gateways will be sold as purpose-matched PBX components - a single-vendor proposition.
Of course, ironically, if your PBX isn't leading-edge and you want a gateway for interoffice toll bypass or similar apps, you may need to plumb third-party offerings. But this isn't the crapshoot it used to be. Vendors like Mediatrix, Quintum and VegaStream offer a plethora of well-engineered choices at varying scales. At lower line-counts, these tend to be price/performance winners. Obviously, gateway products are also available from the likes of Cisco, Nortel Networks, and other voice/data giants.
In all situations, sound quality is pivotal. And this depends a) on many variables and gateway features (codec characteristics, echo management, etc.), and b) on how these variables and features engage with your specific application (i.e., with your PBX hardware, your LAN, your WAN, your bandwidth requirements, etc.), and of course, with the "really big variable" of whether or not you need calls to traverse the Internet (performance and security issues). Additional value may well be placed on a gateway's ability to monitor connection quality and fall back to PSTN connectivity if IP throughput degrades or dies.
These days, interoffice PBX networking via gateway, where not proprietary, is usually enabled by Q.SIG. And PBX compliance with Q.SIG, though variable, is pretty good - meaning that third-party, Q.SIG-ready gateways offer what may be a more affordable solution than in-skin PBX IP trunk cards. In cases where the PBX cabinet is already maxed-out, perhaps much less expensive (not to mention disruptive).
Here are some select examples of gateways in each broad category.
PBX IP Gateway Offerings
Cisco's (408-526-4000, www.cisco.com) Call Manager is a front-runner in the IP-PBX category, but its 1750, 2600 and 3600 routers are also common choices for third-party gateway, when enabled with FXO and FXS or T1 interfaces. In many cases, a preexisting data router makes the case for adding just such voice components to establish voice WANs.
The VG200, purely for voice, takes the same station and trunk interfaces as voice-enabled 2600 and 3600 routers and gateways, but scales down to a typical four analog ports. It brings analog phones and faxes into Call Manager's system using MGCP or H.323 version 2, and provides local gateway out from Cisco IP phones as well.
Inter-tel (480-961-9000, www.inter-tel.com) is another PBX maker, like Mitel and Siemens, betting on SIP and Windows Messenger for presence-informed collaboration/chat/voice and directory services, and the ubiquity of Windows XP for endpoints.
Inter-tel's SIP server, running on a Win 2000 machine alone or with other Inter-Tel applications, integrates tightly with an Inter-Tel Axxess system through the PBX's in-skin IP resource card (IPRC). David Glissmeyer, of ProTel Networks, an Inter-Tel reseller, says, "I can buy 25 user licenses, whatever SIP phones I want (Cisco's work) and any end point I want. The thing I like about Inter-tel is that it's not an all-or-nothing system."
The IPRC can be "chipped," says Glissmeyer, with firmware to run 16 IP or analog devices, or for IP networking between nodes. "We're doing a lot of installs with that now," he says, "setting up VPNs. The Intertel IP phone works beautifully running over the Internet with a G.729 codec. You just basically plug them in. The IP phone includes a four-port switch for daisy-chaining. To use them at a remote location, you ideally, just need two static IDs from your ISP. An additional $5 per static IP address is well worth the money," says Glissmeyer.
Inter-tel also accepts voice signaling through MGCP gateways; it's own four-port MGCP gateway, for remote sites, accommodates local analog ports for 911 local dial tone. This MGCP gateway holds four local lines, communicates back to the head-office switch, and can act as a remote switch. "At 12 lines, it's more economical to put a little server in there and network them. But it's flexible."
Inter-tel's compatibility with Cisco routers and IP phones gives integrators another option when dealing with customers who have Cisco routers already installed. Glissmeyer: "Say I have a customer with offices in NYC and a three-person office Salt Lake City, and some imbedded 1700-series routers. Say they have a compelling argument for toll bypass. I'll drop a couple of FXS-FXO VoIP cards in each of my routers so they can communicate, and also bring local dial tone in from Salt Lake City. That way, I'm doing everything the little Inter-tel MGCP box is, except I'm only getting POTS functionality.
"The beauty of the Inter-tel box is, I'm getting all the rich features." Glissmeyer says that for a three-person office, a quarter of a T-1 frame, managed by the carrier for an additional $100 per month, would provide the needed bandwidth.
"Now with the new SIP server, if I had two locations, and one was running a SIP-compliant Cisco system, say, I can gateway those into my Inter-tel switch through the SIP server and deliver to them a pretty rich feature set off Inter-Tel's call processing software. A suite of applications on the Unified Communicator software, running with Intertel's SIP server, will give you presence and phone control that approximates the rich feature set of Axxess. This will let Cisco or other SIP phones come close."
Mitel Networks (613-592-2122, www.mitel.com), in its excellent November 2002 "Migration to IP Telephony" white paper, delineates four different migration scenarios. Where budgets can support new systems, it makes the case for the convergent PBX that supports both TDM and Ethernet-based switching, and can accept legacy digital phones, IP phones and analog phones.
Where PBXes cannot be replaced or outfitted with gateway add-ons, it suggests networking them to an IP PBX; the Mitel Networks ICP 3300 and smaller-scale ICPs are sold into this niche, supporting traditional Mitel SX-2000 circuit-switched cabinets, as well as serving as small stand-alones. The ICP consists of a Controller for voice, signaling, and media resources between Ethernet IP phones. For traditional telephony, switching goes over a conventional TDM bus. Network Services Units and Analog Services Units, in separate rackmountable boxes, connect to a range of trunks and analog or digital station sets. The platform can handle up to 1,000 stations, of which 700 can be IP. Clustered, it lists a 40,000-station capacity.
Mitel's SOHO SX-200 PBX can be IP-enabled through the SX-200 IP node, which supports Mitel's wired and wireless IP phones. One SX-200 can support up to two nodes, for a maximum 120 IP devices.
There are two ways to network the legacy and IP PBX; through basic tie-trunking, which will support a common dialing plan but not feature/functions, and an intelligent (Q.SIG) networking option, which can also support such features as Caller ID, transfer, forwarding, conferencing, automatic callback, LCR, and centralized voicemail.
The migration from TDM to IP PBX in this scenario is a gradual unplugging of extensions from one and plugging into the other, as IP phones are purchased and LAN/WAN upgrades made. While both operate, they can provide backup for each other.
NEC USA Corporate Networks Group (800-TEAM-NEC, www.cng.nec.com) makes the NEAX 2400 IPX in a standard cabinet arrangement with card-cage gateways for station and trunk support. More recent iterations of the architecture include the IPX-DM, a "distributed model" pure IP enterprise PBX with matched rackmountable gateways, and smaller-scale versions including the 2000 IPS and IPS-DM. They also make the IPS-DMR - a distributed-model "remote" PBX which is essentially a remote-site gateway with PBX brains built in.
NEC also offers the Dterm IP Gateway for IP-enabling older NEAX and Electra systems. This is a rack-mountable (or desktop) adjunct solution that can be trunk-side optioned to work over an IP LAN, or make a direct connection to a FRAD or CSU/DSU. It supports Remote Voice Protocol (RVP) and Remote Voice Protocol over Internet Protocol (RVPoIP) and is available in 8-port, 12-port and 24-port configurations.
Siemens' (408-492-2000, www.siemensenterprise.com) current enterprise VoIP platform pieces comprise an in-skin gateway, the HG1500, which slides into 3000, 4000, and 5000 series HiPath PBXes. This can be configured for phone-facing or trunk-facing H.323 voice-packet conversion, and comes in forms to accept analog phones or Siemens digital phones. It can also - by virtue of the H.323 annex M, which Joan Van Dermate, Siemens' VP Product management, describes as "a standardized way of adding proprietary features atop H.323" - be used to network two HiPaths over IP and preserve almost full PBX functionality to phonesets. Such networked PBXs can also, optionally, be administered as one centralized switch, sharing Expression UM server, attendant, and call center.
Older, HiCom PBXes and those of non-Siemens vendors can be IP-networked using Siemens' own standalone gateways, the RG2500 and 3000 series. These gateways can be deployed, through different software loads, as stand-alone PBXes with integrated gateway, or as a survivable remote in part of a HiPath 5000 network. When joining a multi-vendor gateway, phone features are limited to those supported by QSIG, assuming the PBX is QSIG compliant. A HiPath can also be gatewayed out through a third-party, H.323 compliant gateway, since Siemens's VoIP signaling is not proprietary.
The HiPath PBX is now being upgraded to accept SIP interworking, as part of a workgroup collaboration product strategy being previewed under the name OpenScape. Described earlier in these pages, OpenScape will include voice among IM, web conferencing, and presence-aware communication portals and rely on Microsoft's RTC server. For now, explains VanDerMate, OpenScape can use a three-legged circuit-switched, H.323 and SIP gateway to bring HiPath (or another system's) voice endpoints into multimedia workgroup sessions.
Third-party VoIP gateways
Citel Technology (877-248-3587, www.citel.com) is a third-party entrant among IP migration offerings. Their CITELink Handset Gateway slides into a 3Com NBX IP PBX and mediates between that switch and Norstar M7000 series digital handsets or analog phones. It puts the NBX's unified messaging and TAPI integration on everyone's desktop, without new wires or handsets.
Most recently, Citel and Mitel Networks announced co-development plans to allow Nortel extensions to hook up to another IP switch, Mitel's 3300 Integrated Communications Platform (ICP). In addition to over 500 PBX features, the 3300 ICP, recently profiled here for its SIP enhancements and YourAssistant browser-based client, brings with it a range of embedded applications including voicemail, auto-attendant, ACD, and 802.11b wireless gateway. Q.SIG and DPNSS allow for tight integration with non-Mitel PBXs.
Mediatrix (819-829-8749, www.mediatrix.com) makes a line of analog gateways in two, four, and 24-port versions, with SIP or H.323 or MGCP signaling. They also sell an inexpensive SIP redirect server, for mapping phone numbers to IP endpoints, in SIP VoIP networks.
While they sell through a network of channel partners and service providers, they do publish a list price for their four-port gateway, the Mediatrix 1204, of $750. All are remotely configurable.
Quintum Technology (732-460-9000, www.quintum.com) operates in the enterprise and carrier spheres with its VoIP Multipath Switch and SelecttNet technology. "Multi" path, because it automatically switches the call to the PSTN if IP quality is poor.
The line, using H.323 VoIP, starts with the two-port A200, a router/gateway for small, remote offices; comes with a NAT router and firewall. Ports to PSTN assure voice reach even in the event of total power failure, and also accommodate hop-off for local dial tone. Even at this SOHO scale, it can support two-stage dialing with IVR. The A/800 series supports up to eight simultaneous VoIP calls, with analog interfaces and IVR or RADIUS authentication.
Digital interfaces for a T-1's or E-1's worth of voice or less come with the D/800 to D/3000, and integrate gatekeeper with gateway. At this level, you get a web-based GUI for configuration, CDRs, support for public and private dial plans, and pass-through support for 800 and 911 and 411 calling.
The Tenor MultiPath architecture provides intelligent call routing and complete support for branch office networking with Nortel's Succession 1000, including connectivity between the Succession 1000 network and the PSTN, connectivity between analog phones/fax machines and the Succession 1000, and PSTN hop-off.
Telstrat (972-543-3500, www.telstrat.com) makes a line of terrific gateway products, OEM'd (and serviced by) Nortel. Their Remote Office 9150 extends 32 Meridian digital phones - with full feature support - to a remote office via IP (or ISDN BRI). The secondary route can be used as fallback if IP network quality degrades. They also make single-line versions, the 9110 and 9115.
Vegastream (561-995-2300, www.vegastream.com) is the manufacturer of at least one major announceable VoIP card for PBX makers: Nortel's Succesion MCS5100. Other contracts are in the works, according to marketing manager Michael Robinson.
Vegastream has done a lot of integration work to provide PBX functions across the IP network, including message waiting lamps. Dial plans can be maintained using a combination of PBX and gateway configuration. They've tested their SIP gateway with PingTel, IP Dialog and Snom phones. Their real-time OS, from WindRiver, is not typically as vulnerable to hackers as PC-type operating systems.
Vegastream's line starts with the FXO version Vega50, with eight CO-facing station ports, SIP or H.323 compliant, and stand-alone or with gatekeeper. Web browser configurable, it accepts dialtone, talk battery, and ring current from the connected equipment. The Advanced Dial Planner provides powerful pattern-matching based call routing capabilities. $2,400 from sysconfig.com. The Vega 100 is the T-1, E-1 PRI version, priced at $6,000 at sysconfig.com.
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