User`s guide
Table Of Contents
- VMG8924-B10A and VMG8924- B30A Series
- User’s Guide
- Technical Reference
- Network Map and Status Screens
- Broadband
- Wireless
- Home Networking
- 7.1 Overview
- 7.2 The LAN Setup Screen
- 7.3 The Static DHCP Screen
- 7.4 The UPnP Screen
- 7.5 Installing UPnP in Windows Example
- 7.6 Using UPnP in Windows XP Example
- 7.7 The Additional Subnet Screen
- 7.8 The STB Vendor ID Screen
- 7.9 The 5th Ethernet Port Screen
- 7.10 The LAN VLAN Screen
- 7.11 The Wake on LAN Screen
- 7.12 Technical Reference
- Routing
- Quality of Service (QoS)
- Network Address Translation (NAT)
- Dynamic DNS Setup
- Interface Group
- USB Service
- Power Management
- Firewall
- MAC Filter
- Parental Control
- Scheduler Rule
- Certificates
- VPN
- Voice
- Log
- Traffic Status
- VoIP Status
- ARP Table
- Routing Table
- IGMP/MLD Status
- xDSL Statistics
- 3G Statistics
- User Account
- Remote Management
- TR-069 Client
- TR-064
- SNMP
- Time Settings
- E-mail Notification
- Logs Setting
- Firmware Upgrade
- Configuration
- Diagnostic
- Troubleshooting
- Customer Support
- Setting up Your Computer’s IP Address
- IP Addresses and Subnetting
- Pop-up Windows, JavaScripts and Java Permissions
- Wireless LANs
- IPv6
- Services
- Legal Information
- Index
Chapter 21 Voice
VMG8924-B10A and VMG8924-B30A Series User’s Guide
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Click VoIP > Call History > Call History Incoming Calls. The following screen displays.
Figure 150 VoIP > Call History > Call History Incoming Calls
Each field is described in the following table.
21.10 Technical Reference
This section contains background material relevant to the VoIP screens.
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.
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
Table 118 VoIP > Call History > Call History Incoming
LABEL DESCRIPTION
Refresh Click this button to renew the received call list.
Clear All Click this button to remove all entries from the received call list.
# This is a read-only index number.
time This is the date and time when the call was made.
phone port This is the phone port on which you received the call.
Missed means the call was unanswered.
phone number This is the SIP number that called you.
duration
This displays how long the call lasted.