Zyxel VMG8324-B10A Router Manual PDF (Setup & Configuration Guide)

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Chapter 21 Voice

VMG8324-B10A / VMG8324-B30A Series User’s Guide

241

21.4

The SIP Service Provider Screen

Use this screen to view the SIP service provider information on the Device. Click

VoIP > SIP >

SIP Service Provider

to open the following

screen.

Figure 145

VoIP > SIP > SIP Service Provider

Warm Line

Select this to have the Device dial the specified warm line number after you pick

up the telephone and do not press any keys on the keypad for a period of time.

Hot Line

Select this to have the Device dial the specified hot line number immediately

when you pick up the telephone.

Hot Line / Warm

Line number

Enter the number of the hot line or warm line that you want the Device to dial.

Warm Line Timer

Enter a number of seconds that the Device waits before dialing the warm line

number if you pick up the telephone and do not press any keys on the keypad.

Enable Missed

Call Email

Notification

Select this option to have the Device e-mail you a notification when there is a

missed call.

Mail Server

Select a mail server for the e-mail address specified below. If you select

None

here, e-mail notifications will not be sent via e-mail.

You must have configured a mail server already in the

Email Notification

screen.

Send

Notification to

Notifications are sent to the e-mail address specified in this field. If this field is

left blank, notifications will not be sent via e-mail.

Missed Call

Email Title

Type a title that you want to be in the subject line of the e-mail notifications that

the Device sends.

Early Media

Select this option if you want people to hear a customized recording when they

call you.

IVR Play

Index

Select the tone you want people to hear when they call you.

This field is configurable only when you select

Early Media

. See

Section 21.10

on page 252

for information on how to record these tones.

Music On Hold

Select this option to play a customized recording when you put people on hold.

IVR Play

Index

Select the tone to play when you put someone on hold.

This field is configurable only when you select

Music On Hold

. See

Section

21.10 on page 252

for information on how to record these tones.

Apply

Click this to save your changes and to apply them to the Device.

Cancel

Click this to set every field in this screen to its last-saved value.

Table 112

VoIP > SIP > SIP Account > Add new accoun/Edit (continued)

LABEL

DESCRIPTION

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Each field is described in the following table.

21.4.1

The SIP Service Provider Add/Edit Screen

Use this screen to configure a SIP service provider on the Device. Click the

Add new provider

button or an

Edit

icon in the

VoIP > SIP > SIP Service Provider

to open the following

screen.

Table 113

VoIP > SIP > SIP Service Provider

LABEL

DESCRIPTION

Add new provider

This is the index number of the entry.

SIP Service

Provider Name

This shows the name of the SIP service provider.

SIP Server

Address

This shows the IP address or domain name of the SIP server.

Address

This shows the IP address or domain name of the SIP register server.

SIP Service

Domain

This shows the SIP service domain name.

Modify

Click the

Edit

icon to configure the SIP service provider.

Click the

Delete

icon to delete this SIP service provider from the Device.

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Chapter 21 Voice

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Note: Click

to see all the fields in the screen. You don’t necessarily need to use all

these fields to set up your account. Click

less

to see and configure only the fields

needed for this feature.

Figure 146

VoIP > SIP > SIP Service Provider > Add new provider/Edit

Each field is described in the following table.

Table 114

VoIP > SIP > SIP Service Provider > Add new provider/Edit

LABEL

DESCRIPTION

SIP Service Provider Selection

Service

Provider

Selection

Select the SIP service provider profile you want to use for the SIP account you configure in

this screen. If you change this field, the screen automatically refreshes.

General

SIP Service

Provider Name

Enter the name of your SIP service provider.

SIP Local Port

Enter the Device’s listening port number, if your VoIP service provider gave you one.

Otherwise, keep the default value.

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SIP Server

Address

Enter the IP address or domain name of the SIP server provided by your VoIP service

provider. You can use up to 95 printable ASCII characters. It does not matter whether the

SIP server is a proxy, redirect or register server.

SIP Server Port

Enter the SIP server’s listening port number, if your VoIP service provider gave you one.

Otherwise, keep the default value.

Server Address

Enter the IP address or domain name of the SIP register server, if your VoIP service provider

gave you one. Otherwise, enter the same address you entered in the

SIP Server Address

field. You can use up to 95 printable ASCII characters.

Server Port

Enter the SIP register server’s listening port number, if your VoIP service provider gave you

one. Otherwise, enter the same port number you entered in the

SIP Server Port

field.

SIP Service

Domain

Enter the SIP service domain name. In the full SIP URI, this is the part after the @ symbol.

You can use up to 127 printable ASCII Extended set characters.

RFC Support

Support

Locating SIP

Server

(RFC3263)

Select this option to have the Device use DNS procedures to resolve the SIP domain and

find the SIP server’s IP address, port number and supported transport protocol(s).

The Device first uses DNS Name Authority Pointer (NAPTR) records to determine the

transport protocols supported by the SIP server. It then performs DNS Service (SRV) query

to determine the port number for the protocol. The Device resolves the SIP server’s IP

address by a standard DNS address record lookup.

The

SIP Server Port

and

fields in the

General

section above are

grayed out and not applicable and the

Transport Type

can also be set to

AUTO

if you

select this option.

RFC

3262(Require:

100rel)

PRACK (RFC 3262) defines a mechanism to provide reliable transmission of SIP provisional

response messages, which convey information on the processing progress of the request.

This uses the option tag 100rel and the Provisional Response ACKnowledgement (PRACK)

method.

Select this to have the the peer device require the option tag 100rel to send provisional

responses reliably.

VoIP IOP Flags

Select the VoIP inter-operability settings you want to activate.

Replace dial

digit '#' to

'%23' in SIP

messages

Replace a dial digit “#” with “%23” in the INVITE messages.

Remove ‘:5060’

and

'transport=udp'

from request-

uri in SIP

messages

Remove “:5060” and “transport=udp” from the “Request-URI” string in the REGISTER and

INVITE packets.

Remove the

'Route' header

in SIP

messages

Remove the 'Route' header in SIP packets.

Don't send re-

Invite to the

remote party

when there are

multiple codecs

answered in the

SDP

Do not send a re-Invite packet to the remote party when the remote party answers that it

can support multiple codecs.

Bound Interface Name

Table 114

VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued)

LABEL

DESCRIPTION

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Bound

Interface Name

If you select

LAN

Any_WAN

, the Device automatically activates the VoIP service when

any LAN or WAN connection is up.

If you select

Multi_WAN

, you also need to select two or more pre-configured WAN

interfaces. The VoIP service is activated only when one of the selected WAN connections is

up.

Outbound Proxy

Outbound

Proxy Address

Enter the IP address or domain name of the SIP outbound proxy server if your VoIP service

provider has a SIP outbound server to handle voice calls. This allows the Device to work

with any type of NAT router and eliminates the need for STUN or a SIP ALG. Turn off any SIP

ALG on a NAT router in front of the Device to keep it from re-translating the IP address

(since this is already handled by the outbound proxy server).

Outbound

Proxy Port

Enter the SIP outbound proxy server’s listening port, if your VoIP service provider gave you

one. Otherwise, keep the default value.

RTP Port Range

Start Port

End Port

Enter the listening port number(s) for RTP traffic, if your VoIP service provider gave you this

information. Otherwise, keep the default values.

To enter one port number, enter the port number in the

Start Port

and

End Port

fields.

To enter a range of ports,

•

enter the port number at the beginning of the range in the

Start Port

field.

•

enter the port number at the end of the range in the

End Port

field.

SRTP Support

When you make a VoIP call using SIP, the Real-time Transport Protocol (RTP) is used to

handle voice data transfer. The Secure Real-time Transport Protocol (SRTP) is a security

profile of RTP. It is designed to provide encryption and authentication for the RTP data in

both unicast and multicast applications.

The Device supports encryption using AES with a 128-bit key. To protect data integrity, SRTP

uses a Hash-based Message Authentication Code (HMAC) calculation with Secure Hash

Algorithm (SHA)-1 to authenticate data. HMAC SHA-1 produces a 80 or 32-bit

authentication tag that is appended to the packet.

Both the caller and callee should use the same algorithms to establish an SRTP session.

Crypto Suite

Select the encryption and authentication algorithm set used by the Device to set up an SRTP

media session with the peer device.

Select

AES_CM_128_HMAC_SHA1_80

AES_CM_128_HMAC_SHA1_32

to enable

both data encryption and authentication for voice data.

Select

AES_CM_128_NULL

to use 128-bit data encryption but disable data authentication.

Select

NULL_CIPHER_HMAC_SHA1_80

to disable encryption but require authentication

using the default 80-bit tag.

DTMF Mode

Control how the Device handles the tones that your telephone makes when you push its

buttons. You should use the same mode your VoIP service provider uses.

RFC2833

- send the DTMF tones in RTP packets.

PCM

- send the DTMF tones in the voice data stream. This method works best when you are

using a codec that does not use compression (like G.711). Codecs that use compression

(like G.729 and G.726) can distort the tones.

SIP INFO

- send the DTMF tones in SIP messages.

Transport Type

Select the transport layer protocol

UDP

TCP

(usually UDP) used for SIP.

Table 114

VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued)

LABEL

DESCRIPTION

Rate

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