This should be a new chapter of a much shorter depth than the previous version. We may work a glossary-like version that explains common terms (20-30) and compares them in various solutions and points to the appropriate term native to each. Include references to outside material for a more detailed analysis.

General Terms

VoIP protocols

There are many protocols involved in the various functions of VoIP services. However, the two most popular protocols used for the signaling of calls are H.323 and SIP. Interoperability of clients to services relies primarily on signaling and secondarily on the exchange of media. For a general discussion of VoIP, check Wikipedia

VoIP codecs

The term codec refers to the method of coding/decoding of the media data (voice and video) to meet bandwidth requirements of the available network channel. A caller device has to support the same codecs as those available on the other side, i.e. the called device or server. Otherwise the exchange of media may not take place, even though call signaling negotiation has been successful. There are many codecs available for voice, the most popular of which are the G.xxx series as standardized by the ITU. The most commonly available codecs are: G.711 - Pulse code modulation (PCM) demanding a 64 kbps channel and  G.729 - Conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP) demading a 8 kbit/s channel. For a general discussion of Voice Codecs, see Wikipedia

VoIP services

Attempting a simple classification of VoIP services

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general, value-added and ECS

SIP Proxy server

H.323 Gatekeeper

Agent - client - endpoint

Gateway

Media proxy

PBX

Call routing

, for the purposes of this cookbook, we see:

• basic services: registration of endpoint on servers, call routing, location and call translation
• value-added: voice-mail, missed call notification
• enhanced communication services: interoperability of voice calls with other media, like video, instant messaging, as well as trusted domain infrastructure to support level of trust indication to the end user

Endpoint - Agent - Client - Phones

All three terms refer the device (phone) or software (soft-phone) that the end user utilizes to place and receive calls. Phones are termed "endpoints" in H.323 terminology, "agents" in SIP terminology and all other terms are used interchangeably.

Gatekeeper-Registrar-Location-Proxy Server

Phones in most cases require a server to register to, before receiving calls, so they can be located while mobile. This requirement, as well as the need for reliable caller ids, imposes the need for authenticating phones, before allowing them connection to their local domain server. These servers are termed "gatekeepers" in H.323 terminology, "registrar-location-proxy" servers in SIP terminology.

Gateway

A device for allowing calls between two incompatible devices that can not directly call each other. For example, a sip phone needs a sip2h323 gateway in order to call an H.323 endpoint. Also, a sip phone needs a sip2pstn gateway in order to call a traditional PBX or PSTN phone line. A gateway converts the signaling between the two end devices and terminates media channels on both sides, playing the intermediary for the call.

Media proxy

An intermediary device that terminates media channels, but does not handle signaling at all. Media proxies are used mostly for traversing firewalls, bypassing NAT, or protecting domains from direct access by outsiders, by allowing calls to reach the border-perimeter of a domain, without opening up the whole network.

PBX

A traditional telephone center used in most organizations-enterprises, responsible for connecting (switching) telephone ports between subscribers (local phones). Recently, some PBXs are replaced by IP-PBXs which rely on VoIP at their core, rather than switched circuitry.

Call routing

The functionality of VoIP servers to route incoming calls to their destination. This task is easy for calls placed between two users on the same local server. However, in most cases, an incoming call may indicate a destination (dialed number) in a remote server and in a format that needs interpretation, in which case the routing protocols that the local server implements will need to check and translate the incoming dialed number.

NAT traversal

Phones that are located behind NAT networks are usually at a disadvantage, as communication with their local server is obstructed by the NAT. There are many methods to help traverse the NAT barrier, which vary based on the VoIP signaling protocol. These methods require support on the local server, on the phone itself and sometimes they require the existence of a Media Proxy as well.

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SIP-related Terms

Domain

Identity

TLS

SIP.edu

ITAD/ISN

Telephony Terms

E.164

Definition.
link for National prefixes

ENUM

Golden-tree and unofficial ENUM trees
Differences between infrastructure ENUM