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Sometimes buzz words can be difficult, so here are few defined for you.
Session Initiation Protocol (SIP) is a signalling protocol, designed to initiate and terminate communications between multimedia communication sessions, mainly voice and video calls over the Internet or data networks.
A trunk is a way to get external calls in and out of the PBX. All extensions and accounts are internal, at some point someone will want to talk to other telephone users outside the PBX. This is done by using a trunk from the PBX to the telephone network.
Transport Layer Security (TLS) is a protocol that is used for establishing a secure connection between a client and a server. SIP uses the secure transport layer (TLS). TLS is based on SSL 3.0. The encryption of the voice packets uses a different standard - SRTP. SRTP is based on AES (at least 128 bit). This provides better assurance of privacy even if someone is able to capture packets on your network.
The Real Time Transport Protocol (RTP) is an Internet protocol standard that specifies a way for programs to manage the real-time transmission of multimedia data over network services. In other words the voice part of Voice over IP.
DTMF means - dual tone multi frequency. This is also known as Touch Tone. It is a method, used by the telephone system to communicate the pressed keys when dialling. DTMF allows navigation of voice menus and other advanced calling services.
UHLL means - Universal Hospitality Layer Link. It is a middleware management system that sits between many devices or applications and hotel systems. pbxnsip communicates to the UHLL allowing it to be integrated with many hotel platforms around the world.
The PSTN is the Public Switch Telephone Network, a term for the telephone network as we have used for many years.
Voice over Internet Protocol or VoIP / IPA is a general term for data transmission delivery of voice communications over IP networks such as the Internet or other packet-switched networks. VoIP is also known as IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. VoIP systems or IP PBX's can interface with the traditional PSTN PABX telephone systems to allow for transparent phone communications worldwide.
Network Address translation (NAT) is the translation of an Internet Protocol address (IP address) between networks. One network is designated the inside network and the other is the outside. Typically, a company maps its local inside network address to a global outside (public) IP addresses. NAT unmaps the global IP addresses on incoming packets back into local IP addresses. This helps ensure security. It can also cause severe headaches without a NAT filter or session border controller.
An agent group can also be called an ACD. It is a type of account that allows calls to be placed in a queue. The queue has agents assigned to the queue and those agents can log in or out of the queue. Customers who are in the queue will be played music as well up to 10 different recordings.
UDP (User Datagram Protocol) is a communications protocol that offers a limited amount of service when messages are exchanged between computers in a network that uses the Internet Protocol (IP). UDP is an alternative to the Transmission Control Protocol (TCP) and, together with IP, is sometimes referred to as UDP/IP.
Session Border Controller (SBC) is a device or application that governs the manner in which calls, also called sessions, are initiated, conducted and terminated in a VoIP (Voice over Internet Protocol) network.
Caller Line Identification or caller ID or CLI. Is use a telephone feature to display the caller's number on the screen of the recipient's handset or softphone, provided that the calling number is not blocked. This feature is available on digital and some analogue handsets.
Direct Dialling In, also known as DID, Direct Inward Dialling. DID allows the extension line on a PABX or IP PBX to be called directly from the public network without operator intervention.
T.38 is a technical standard that describes how two endpoints should report packet losses and resend the missing packets so that the communications between devices cam complete a transmission.
Codec is designed to convert an signals - analogue to digital so that a data network can process the information.
- GSM - 13 Kbps, 20ms
- iLBC - 15Kbps - 20ms & 13.3 Kbps - 30ms
- G.711 - 64 Kbps, (alaw/ulaw)
- G.722 - 48/56/64 Kbps
- G.723.1 - 5.3/6.3 Kbps, 30ms frame size
- G.726 - 16/24/32/40 Kbps
- G.728 - 16 Kbps
- G.729 - 8 Kbps, 10ms
- Speex - 2.15 to 44.2 Kbps
- LPC10 - 2.5 Kbps
- DoD CELP - 4.8 Kbps
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