IP Network Support for Voice ________________________________________

A key requirement for successful VoIP deployment is the availability of an internal IP-based network that is capable of supporting real-time telephone and facsimile. As was noted above, voice quality is affected by delay, jitter, and unreliable packet delivery - all of which are typical characteristics of the basic IP network service.

Most of today's data network equipment - routers, LAN switches, ATM switches, network interface cards, PBXs, etc. - will need to be able to support voice traffic. Furthermore, VoIP-specific equipment will either have to be integrated into these devices or work compatibly with them. VoIP equipment must also accommodate environments ranging from private, well-planned corporate Intranets to the less predictable Internet. Three different techniques are used (separately or in combination) to improve network quality of service.

Providing a controlled networking environment in which capacity can be pre-planned and adequate performance can be assumed (at least most of the time). This would generally be the case with a private IP network (an Intranet) that is owned and operated by a single organization.
Using management tools to configure the network nodes, monitor performance, and manage capacity and flow on a dynamic basis. Most internetworking devices (routers, switches, etc.) include a variety of mechanisms that can be useful in supporting voice. For example, traffic can be prioritized by location, by protocol, or by application type, thereby allowing real-time traffic to be given precedence over non-critical traffic. Queuing mechanisms can also be manipulated to minimize delays for real-time data flows. More recent developments, such as tag switching and flow switching, can also improve overall performance and reduce delays.
Adding control protocols and mechanisms that help avoid or alleviate the problems inherent in IP networks. Protocols such as RTP (real-time protocol) and RSVP (Resources Reservation Protocol) are also being used to provide greater assurances of controlled QoS within the network. Other mechanisms such as admission controls and traffic shaping may also be used to avoid overloading a network (this would be comparable to getting a network busy signal on the telephone at peak periods such as Christmas).

Carriage of Voice Traffic ____________________________________________

VoIP equipment, which can be categorized into client, access/gateway, carrier class/infrastructure segments, should be configurable to capitalize on these different techniques but must also be sufficiently flexible to add new techniques as they become available. Producers that make use of embedded software should focus on how to best utilize the functions instead of focusing on the problems associated with implementing and testing the objects themselves. Real-time voice traffic can be carried over IP networks in three different ways:

Voice trunks can replace the analog or digital circuits that are serving as voice trunks (such as private links between company-owned PBXs) or PSTN-access trunks (links between a PBX and the carrier). Voice packets are transferred between pre-defined IP addresses, thereby eliminating the need for phone number to IP address conversions. Fallback to the PSTN (or other private voice circuits) is always an option in this scenario.
PC-to-PC voice can be provided for multimedia PCs (i.e., PCs with a microphone and sound system) operating over an IP-based network without connecting to the PSTN. PC applications and IP-enabled telephones can communicate using point-to-point or multipoint sessions (a form of Internet ham radio). This type of system may emulate a CB radio or an Internet chat group and could be combined with shared data systems such as whiteboards (i.e., multimedia solutions).
Telephony (any phone-to- any other phone) communications appears like a normal telephone to the caller but may actually consist of various forms of voice over packet network, all interconnected to the PSTN. Gateway functionality is required when interconnecting to the PSTN or when interfacing the standard telephones to a data network. In the future, IP-enabled telephones will connect directly. For true universality, standards for VoIP (and voice over frame relay or ATM) must be adopted and applied.

 

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