[3]

• Third party call control (3pcc): Feature that allows to have Click-to-Call or Click-to-Dial feature.
  Additional Resource: http://en.wikipedia.org/wiki/3pcc

[A]

• Analog Telephony Adapter (ATA): Device that connects to standard analog phones to providing conversion between analog voice signal and digital voice data. Many service providers ask users to install ATA at their home location to connect to their VoIP service, such as Vonage.
  Additional Resource: http://en.wikipedia.org/wiki/Analog_telephony_adapter

[B]

• Back-to-Back User Agent (B2BUA): Acts as a UA server (UAS) on one end, on the other end as a UA client (UAC) and generates requests. http://en.wikipedia.org/wiki/Back-to-back_user_agent


• Brekeke: "Croak" or "Ribbit - the sound a frog makes in Hungarian.


• Brekeke SIP Server: The Brekeke SIP Server has the functionality known as proxy server and registrar server. Brekeke SIP Server can also act as a Back-to-Back User Agent (B2BUA). Brekeke SIP Server is a stateful proxy.

[C]

• Call Hunting: Telephony feature that distribute a call to multiple forwarding destinations.
  Additional Resource: Types of Call Hunting feature - Top-down (Linear hunting), Round-robin (Circular hunting), etc.


• Call Whisper:
  Also known as Call Screening. Receipient of the call can screen the call (caller-ID, dialied number, etc.) and decide whether if answer the call or not.
  Additional Resource: http://en.wikipedia.org/wiki/Call_Whisper


• Click-to-Call (CTC): Also known as Call-to-Dial. Click-to-Call allow user to click on a phone number or link to make a call.
  Additional Resource: http://en.wikipedia.org/wiki/Click-to-call


• Conference Call: http://en.wikipedia.org/wiki/Conference_call
  Additional Resource: http://www.sip-dojo.com/what-is-pbx/conference-call

[G]

• Global IP Address: The global IP address is the IP address at which others on the internet see your connection as originating from. The global IP address is unique and assigned only to a single device.
  


• Google Voice: Internettelephony service provided by Google. This service can forward a call to multiple locations and can even perform speech-to-text message translations.
  Additional Resource: http://www.google.com/googlevoice/about.html

[H]

• Heartbeat: A cluster management program for high availability clustering.http://en.wikipedia.org/wiki/Heartbeat_%28program%29
  


• High Availability: http://en.wikipedia.org/wiki/High_availability
  

[I]

• Interoperability: http://en.wikipedia.org/wiki/Interoperability
  Additional Resource: http://wiki.brekeke.com/wiki/Interoperability


• ITSP (Internet Telephony Service Provider): http://en.wikipedia.org/wiki/ITSP


• IVR (Interactive Voice Response): http://en.wikipedia.org/wiki/IVR
  Additional Resource: http://www.sip-dojo.com/what-is-pbx/sip-ivr

[M]

• Media Gateway: http://en.wikipedia.org/wiki/Media_gateway


• Mirroring: Mirroring is a technique to allow a system to automatically maintain multiple copies of data so that in the event of a failure a system can continue to process or quickly recover data.

[N]

• NAT (Network Address Translation): http://en.wikipedia.org/wiki/Network_address_translation

[P]

• PBX (Private Branch Exchange): http://en.wikipedia.org/wiki/Private_branch_exchange


• POTS (Plain Old Telephone Service): http://en.wikipedia.org/wiki/Plain_old_telephone_service


• Private Network: http://en.wikipedia.org/wiki/Private_network


• PSTN (Public Switched Telephone Network): http://en.wikipedia.org/wiki/Public_switched_telephone_network

[R]

• Registrar server: Server that handles location registration messages.


• RFC (Request for comments): http://en.wikipedia.org/wiki/Request_for_Comments


• RTP (Real-time Transport Protocol): http://en.wikipedia.org/wiki/Real-time_Transport_Protocol

[S]

• SDP (Session Description Protocol): http://en.wikipedia.org/wiki/Session_Description_Protocol
  Additional Resource: RFC 4566


• SIP (Session Initiation Protocol): http://en.wikipedia.org/wiki/Session_Initiation_Protocol
SIP server: SIP server offers three types of server functionality as defined in RFC 3261 EProxy, Redirect and Registrar servers.

• SIP UA (SIP User Agent): Terminal device within SIP network (e.g., SIP hardphone, SIP softphone, etc.)


• SIP URL: Address that is used for SIP communication (e.g., 1002@sip.brekeke.com) 
  Additional Resource: RFC2396


• Skype: Software application that allows people to make telephone calls over the Internet. One of Skype’s benefits is its high voice quality, despite VoIP’s prior reputation for having poor voice quality. Currently, Skype is working on a project called,"Skype for SIP" that allows Skype users to connect with SIP services/devices.


• SRTP (Secure Real-Time Transport Protocol): http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol


• Stateful proxy: Compared to stateless proxy, stateful proxy can offer more complex features, such as TCP support and Call Forking. Stateful proxy remembers the requests that have been forwarded, as opposed to stateless proxy, which will not remember requests once they have been forwarded. As a result, stateless proxy can scale larger and faster.


• STUN (Session Traversal Utilities for NAT): http://en.wikipedia.org/wiki/STUN

[T]

• TCP (Transmission Control Protocol): http://en.wikipedia.org/wiki/Transmission_Control_Protocol


• Termination (or Call Termination): http://en.wikipedia.org/wiki/Call_termination


• TLS (Transport Layer Security): http://en.wikipedia.org/wiki/Transport_Layer_Security

[U]

• UDP (User Datagram Protocol): http://en.wikipedia.org/wiki/User_Datagram_Protocol


• UPnP (Universal Plug and Play): http://en.wikipedia.org/wiki/Universal_Plug_and_Play