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Chapter 21 Voice
VMG8324-B10A / VMG8324-B30A Series User’s Guide
251
Each field is described in the following table.
21.8
The Call History Outgoing Calls Screen
Use this screen to see detailed information for each outgoing call you made.
Click
VoIP > Call History > Call History Outgoing
. The following screen displays.
Figure 150
VoIP > Call History > Call History Outgoing
Each field is described in the following table.
21.9
The Call History Incoming Calls Screen
Use this screen to see detailed information for each incoming call from someone calling you.
Table 117
VoIP > Call History > Call History Summary
LABEL
DESCRIPTION
Refresh
Click this button to renew the call history list.
Clear All
Click this button to remove all entries from the call history list.
#
This is a read-only index number.
Date
This is the date when the calls were made.
Total Calls
This displays the total number of calls from or to your SIP numbers that day.
Outgoing Calls
This displays how many calls originated from you that day.
Incoming Calls
This displays how many calls you received that day.
Missing Calls
This displays how many incoming calls were not answered that day.
Total Duration
This displays how long all calls lasted that day.
Table 118
VoIP > Call History > Call History Outgoing
LABEL
DESCRIPTION
Refresh
Click this button to renew the dialed call list.
Clear All
Click this button to remove all entries from the dialed 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 made the call.
phone number
This is the SIP number you called.
duration
This displays how long the call lasted.
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Click
VoIP > Call History > Call History Incoming Calls
. The following screen displays.
Figure 151
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 119
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.
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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 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).
After successful registration, the SIP server knows that the users (identified by their dedicated SIP
URIs) are represented by the UA, and knows the IP address to which the SIP requests and
responses should be sent.
Registration is initiated by the User Agent Client (UAC) running in the VoIP gateway (the Device).
The gateway must be configured with information letting it know where to send the REGISTER
message, as well as the relevant user and authorization data.
A SIP registration has a limited lifespan. The User Agent Client must renew its registration within
this lifespan. If it does not do so, the registration data will be deleted from the SIP registrar's
database and the connection broken.
The Device attempts to register all enabled subscriber ports when it is switched on. When you
enable a subscriber port that was previously disabled, the Device attempts to register the port
immediately.
Authorization Requirements
SIP registrations (and subsequent SIP requests) require a username and password for
authorization. These credentials are validated via a challenge / response system using the HTTP
digest mechanism (as detailed in RFC 3261, "SIP: Session Initiation Protocol").
SIP Servers
SIP is a client-server protocol. A SIP client is an application program or device that sends SIP
requests. A SIP server responds to the SIP requests.
When you use SIP to make a VoIP call, it originates at a client and terminates at a server. A SIP
client could be a computer or a SIP phone. One device can act as both a SIP client and a SIP server.
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SIP User Agent
A SIP user agent can make and receive VoIP telephone calls. This means that SIP can be used for
peer-to-peer communications even though it is a client-server protocol. In the following figure,
either
A
or
B
can act as a SIP user agent client to initiate a call.
A
and
B
can also both act as a SIP
user agent to receive the call.
Figure 152
SIP User Agent
SIP Proxy Server
A SIP proxy server receives requests from clients and forwards them to another server.
In the following example, you want to use client device
A
to call someone who is using client device
C.
1
The client device (
A
in the figure) sends a call invitation to the SIP proxy server (
B
).
2
The SIP proxy server forwards the call invitation to
C
.
Figure 153
SIP Proxy Server
SIP Redirect Server
A SIP redirect server accepts SIP requests, translates the destination address to an IP address and
sends the translated IP address back to the device that sent the request. Then the client device that
originally sent the request can send requests to the IP address that it received back from the
redirect server. Redirect servers do not initiate SIP requests.
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In the following example, you want to use client device
A
to call someone who is using client device
C
.
1
Client device
A
sends a call invitation for
C
to the SIP redirect server (
B
).
2
The SIP redirect server sends the invitation back to
A
with
C
’s IP address (or domain name).
3
Client device
A
then sends the call invitation to client device
C
.
Figure 154
SIP Redirect Server
SIP Register Server
A SIP register server maintains a database of SIP identity-to-IP address (or domain name)
mapping. The register server checks your user name and password when you register.
RTP
When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is used to handle voice
data transfer. See RFC 1889 for details on RTP.
Pulse Code Modulation
Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time intervals and
converts them into bits.

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