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4. Overview of VoIP Technology·Gateways
·IP PBXs
·Elements of a VoIP Network
·Latency
| | | A VoIP gateway can be loosely defined as a mechanism that takes circuit-switched voice from a traditional PBX, converts it to IP and transfers it across a LAN or WAN to another gateway where it is reconstituted back into a format that is understood by the receiving phone system.
Gateway functionality can be obtained through stand-alone boxes, modules or chassis cards for proprietary boxes; also expandable routers or software and expansion cards for Windows NT servers.
One of the front-runners in VoIP, Cisco is taking a modular approach with a voice-over-IP card that fits its routers. Cisco says all of its products can easily be equipped for voice and that voice packets can be guaranteed via quality-of-service (QoS) policy implementation on a Cisco-switched network.
Lucent, Nortel Networks and Siemens offer similar strategies for providing VoIP gateway capabilities in some form or another.
While gateways are the most popular VoIP products on the market -- available from at least 30 vendors -- the key point here is that you have voice packets running over IP. However, the packets are not running on the Internet, and you're not gaining any of the features and capabilities you get by converging voice and data networks.
You can find a list of gateway vendors Here
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by KeysCapt  edited by Macy
last modified: 2006-09-06 11:50:29 | | | IP PBXs are great if you have the luxury of designing your system from the ground up. They are complete phone systems, usually with options that include many of the IP telephony applications, such as managing your phone from your desktop PC, multi-line call control and automatic call distribution.
IP PBXs are usually NT servers with telephony software and voice cards. Some disadvantages are the ability to scale the system and a dial tone that's dependent on NT, which doesn't offer the same uptime as a switched phone network.
Until recently IP PBXs have been targeted at offices with 100 users or less, but Alcatel recently announced a system that incorporates gateway and call processing in a single device and can accommodate up to 50,000 users. 3Com, Lucent and Cisco have all announced plans to provide the same type of product.
The beauty of an IP PBX is being able to create a distributed system. For example, allowing you to distribute your phone system throughout an IP network, so geographically separated phones with features such as direct dial, call forwarding, conferencing and voice mail provide the appearance of being connected directly to the local PBX.
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by KeysCapt | | | VoIP services need to be able to connect to traditional circuit-switched voice networks.
The H.323 standard components are: H.323 terminals that are endpoints on a LAN, gateways that interface between the LAN and switched circuit network, a gatekeeper that performs admission control functions and other chores, and the MCU (Multipoint Control Unit) that offers conferences between three or more endpoints.
H.323 terminals are LAN-based end points for voice transmission. Common examples are a computer running Microsoft NetMeeting and an Ethernet-enabled telephone. All H.323 terminals support real-time, 2-way communications with other H.323 entities.
The gateway serves as the interface between the H.323 and non-H.323 network. On one side, it connects to the traditional voice world, and on another side to packet-based devices.
The gatekeeper is not a required device in an H.323 network, but if present, it must perform a set of functions. Gatekeepers provide address translation (routing) for devices in their zone. This could be, for instance, the translation between internal and external numbering systems. Another important function for gatekeepers is providing admission control, specifying what devices can call what numbers.
MCU's allow for conferencing functions between multiple terminals. They contain two parts:
A multipoint controller (MC) that handles the signalling and control messages necessary to setup and manage conferences, and a multipoint processor (MP) that accepts streams from endpoints, replicates them and forwards them to the correct participating endpoints.
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by KeysCapt | | | In contrast to broadcast-type media transmission (RealAudio, for instance), latency in a two-way phone conversation can be very undesirable. 150ms is the specified maximum desired one-way latency to achieve high-quality voice. Voice users will notice round-trip delays that exceed 250ms. More than that, and callers start talking over each other. At anything beyond 500ms phone calls are no longer practical. It would be like asking a question and getting the answer after you've gone to lunch. As a reference, the typical delay when speaking through a geo-stationary satellite is 150-500ms.
Delay in data networks is not as critical. An additional delay of 200ms on an e-mail or web page will generally not be noticed, but voice callers on the same network would be hampered by such a delay.
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