Chapter 17 VoIP
LTE-5121 User’s Guide
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17.8 Technical Reference
This section contains background material relevant to the VoIP screens.
17.8.1 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.
17.8.2 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 address identifies an e-mail account. The format of a SIP identity
is SIP-Number@SIP-Service-Domain.
SIP Number
The SIP number is the part of the SIP URI that comes before the “@” symbol. A SIP number can
use letters like in an e-mail address ([email protected] for example) or numbers like a
telephone number ([email protected] for example).
SIP Service Domain
The SIP service domain of the VoIP service provider is the domain name in a SIP URI. For example,
if the SIP address is
, then “VoIP-provider.com” is the SIP service
domain.
SIP Registration
Each LTE Device is an individual SIP User Agent (UA). To provide voice service, it has a public IP
address for SIP and RTP protocols to communicate with other servers.
A SIP user agent has to register with the SIP registrar and must provide information about the
users it represents, as well as its current IP address (for the routing of incoming SIP requests).
Содержание LTE5121
Страница 4: ...Contents Overview LTE 5121 User s Guide 4...
Страница 12: ...Table of Contents LTE 5121 User s Guide 12...
Страница 13: ...13 PART I User s Guide...
Страница 14: ...14...
Страница 22: ...Chapter 1 Introduction LTE 5121 User s Guide 22...
Страница 28: ...28...
Страница 42: ...Chapter 4 Broadband LTE 5121 User s Guide 42...
Страница 70: ...Chapter 5 Wireless LTE 5121 User s Guide 70...
Страница 98: ...Chapter 7 Static Route LTE 5121 User s Guide 98...
Страница 102: ...Chapter 8 DNS Route LTE 5121 User s Guide 102...
Страница 114: ...Chapter 9 Quality of Service QoS LTE 5121 User s Guide 114...
Страница 138: ...Chapter 14 Parental Control LTE 5121 User s Guide 138...
Страница 148: ...Chapter 15 Certificates LTE 5121 User s Guide 148...
Страница 151: ...Chapter 16 VPN LTE 5121 User s Guide 151 Figure 104 IPSec VPN Add...
Страница 162: ...Chapter 16 VPN LTE 5121 User s Guide 162...
Страница 198: ...Chapter 19 User Account LTE 5121 User s Guide 198...
Страница 214: ...Chapter 26 Backup Restore LTE 5121 User s Guide 214...
Страница 224: ...Chapter 28 Troubleshooting LTE 5121 User s Guide 224...
Страница 234: ...Index LTE 5121 User s Guide 234...