ZyXEL P-2612HNU-F3 User Guide - Page 244

SIP Identities

Page 244 highlights

Chapter 16 VoIP Table 64 VoIP > FXO (continued) LABEL Force to SIP if PSTN unplugged DESCRIPTION Select this check box to have the ZyXEL Device redirect outgoing calls to the registered SIP account if the ZyXEL Device is not connected to the PSTN network. Apply Cancel When you try to make a PSTN call, but the PSTN port on the ZyXEL Device is unplugged, the ZyXEL Device uses the phone port's registered SIP account to make the call. Click Apply to save your changes. Click Cancel to restore your previously saved settings. 16.10 Technical Reference This section contains background material relevant to the VoIP screens. 16.10.1 VoIP VoIP is the sending of voice signals over Internet Protocol. This allows you to make phone calls and send faxes over the Internet at a fraction of the cost of using the traditional circuit-switched telephone network. You can also use servers to run telephone service applications like PBX services and voice mail. Internet Telephony Service Provider (ITSP) companies provide VoIP service. Circuit-switched telephone networks require 64 kilobits per second (Kbps) in each direction to handle a telephone call. VoIP can use advanced voice coding techniques with compression to reduce the required bandwidth. 16.10.2 SIP The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that handles the setting up, altering and tearing down of voice and multimedia sessions over the Internet. SIP signaling is separate from the media for which it handles sessions. The media that is exchanged during the session can use a different path from that of the signaling. SIP handles telephone calls and can interface with traditional circuitswitched telephone networks. SIP Identities A SIP account uses an identity (sometimes referred to as a SIP address). A complete SIP identity is called a SIP URI (Uniform Resource Identifier). A SIP account's URI identifies the SIP account in a way similar to the way an e-mail 244 P-2612HNU(L)-FxF User's Guide

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Chapter 16 VoIP
P-2612HNU(L)-FxF User’s Guide
244
16.10
Technical Reference
This section contains background material relevant to the
VoIP
screens.
16.10.1
VoIP
VoIP is the sending of voice signals over Internet Protocol. This allows you to
make phone calls and send faxes over the Internet at a fraction of the cost of
using the traditional circuit-switched telephone network. You can also use servers
to run telephone service applications like PBX services and voice mail. Internet
Telephony Service Provider (ITSP) companies provide VoIP service.
Circuit-switched telephone networks require 64 kilobits per second (Kbps) in each
direction to handle a telephone call. VoIP can use advanced voice coding
techniques with compression to reduce the required bandwidth.
16.10.2
SIP
The Session Initiation Protocol (SIP) is an application-layer control (signaling)
protocol that handles the setting up, altering and tearing down of voice and
multimedia sessions over the Internet.
SIP signaling is separate from the media for which it handles sessions. The media
that is exchanged during the session can use a different path from that of the
signaling. SIP handles telephone calls and can interface with traditional circuit-
switched telephone networks.
SIP Identities
A SIP account uses an identity (sometimes referred to as a SIP address). A
complete SIP identity is called a SIP URI (Uniform Resource Identifier). A SIP
account's URI identifies the SIP account in a way similar to the way an e-mail
Force to SIP if
PSTN un-
plugged
Select this check box to have the ZyXEL Device redirect outgoing calls
to the registered SIP account if the ZyXEL Device is not connected to
the PSTN network.
When you try to make a PSTN call, but the PSTN port on the ZyXEL
Device is unplugged, the ZyXEL Device uses the phone port’s registered
SIP account to make the call.
Apply
Click
Apply
to save your changes.
Cancel
Click
Cancel
to restore your previously saved settings.
Table 64
VoIP > FXO (continued)
LABEL
DESCRIPTION