Glossary

PBX

private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public. 

PBXs make connections among the internal telephones of a private organization—usually a business—and also connect them to the public switched telephone network (PSTN) via trunk lines.

Trunk

trunk line is a circuit connecting telephone switchboards (or other switching equipment), as distinguished from local loop circuit which extends from telephone exchange switching equipment to individual telephones or information origination/termination equipment.

When dealing with a private branch exchange (PBX), trunk lines are the phone lines coming into the PBX from the telephone provider. This differentiates these incoming lines from extension lines that connect the PBX to (usually) individual phone sets. 

Extensions 

telephone extension is an internal telephone line attached to a Private branch exchange (PBX). The PBX operates much as a community switchboard does for a geographic telephone numbering plan and allows multiple lines inside the office to connect without each phone requiring a separate outside line. In these systems, a dialer usually has to dial a number to tell the PBX to connect with a landline to dial an external number. Within the PBX, the user merely dials the extension number of the person. Each phone line may be extended up to a fixed maximum.  

Secure Call

Secure Call is a voice connection which can't be wiretapped and it runs over Voice Over IP (VoIP) communication protocol.

End to Site security model

Secure call is encrypted from client up to server.

The end to site security model provides a strong security level and can be used among two or more  equipped devices and/or among SNOM 300 landline devices or also for connecting other  PBX, secure or not. If PBX is not secure, we face a crypto-to-clear scenario, where the call is secured between  and but is not secured between  and PBX.
Given that in this security model the server can decrypt secure calls content, it is possible to provides advanced telephony features such as:
  • 3-way calls
  • call transfer
  • conference rooms

 

End to End security model

Secure call is encrypted from client up to the other client. Despite server relays encrypted traffic, it does not knows the encryption keys, so it cannot decrypt the call content.

The end to end security model provides the highest security level but can be used only between two  equipped devices.
This security model does cannot be used to integrate enterprise PBXs.

In this security model the server cannot decrypt secure call content, so advanced features are not available. 

 is the PBX committed to perform Secure Calls both end to end and end to site. It differs from a standard PBX for exposing just the Secure Call service to VoIP  clients and can be connected to a standard PBX via SIP Trunks if configured accordingly.

 is the VoIP client for Secure Calls connections. It has to be used along with .


Conference Rooms

The Conference Room is the kind of call that more persons can partecipate. The conference calls are usually defined as "rooms", whose access can be limited by time settings or pass code. 

This feature is not available for account end-to-end accounts.


Conference Calls

Conference Call, as its name implies, is a call involving at least three users. It differs from Conference Room by the fact that Conference Call is one Secure Call at which third parties got invited. So it's one sort of dynamic conference room. All users invited are thus added to conversation in progress by either caller or callee.

This feature is not available for account end-to-end accounts.

Call transfer

Call transfer is a typical PBX performance which is implemented in  as well. One of the partecipant can hold on his/hers peer and perform a new call to the number to which trasfer the call. If the desired number picks up the call, then the transferrer can close the communication and let his/hers peer talk with the trasferred number.

This feature is not available for account end-to-end accounts.

Jitter 

In VoIP systems audio signal is split into multiple packets, which are sent over network. Due to network equipment behavior, packets flow is never regular and constant. Especially on mobile/radio networks packets are delivered in bursts, leading to irregular and variable latency. Jitter is the variation in latency as measured in the variability over time of the packet latency across a network

Automatic Activation 

The Automatic Activation is the way of create new users automatically without any need they interact with anybody. As from user's side the procedure is that he/she gets first an invite SMS or E-Mail useful to download the application  and then a configuration SMS or E-Mail which provides automatic configuration of the client itself. All the user has to do is to follow the links into both the Texts/E-mails and  would go automatically on line.

Provisioning

 The Provisioning is the configuration needed for delivering both for the  application and its configuration and nowadays it's used by the Automatic Activation only. 

Presence

The Presence is how we call the user's status, also known as the user's reachability. By checking an user's Presence it is possible to know if a he/she is on line and can receive a secure call before trying to.

Audio Messaging

The Audio Messaging is the means used by  for communicating to an user about the failed calls. You have several messages that can be spoken and each of them can be localised in English, French, Italian, Spanish and German.

Secure Message

Secure Message is a text message that can be sent and received only by using  and that shares the same communication infrastructure of Secure Call. The maximum length of each message is 160 character.

Call Roaming

Call Roaming is specific configuration set up by default on both  and . It let any call to continue even if one network change event occur (e.g: wi-fi network got lost in favour of 3g data mobile one). During network change itself the call is muted but as soon as any connection is again available, then voice stream is back streaming. Call roaming is subject to Strictrtp and Rtp timeout options in