Evolución de la arquitectura de red para los servicios de Internet, VoIP e IPTV 1 Servicio de Internet

The following figure shows the SIP Settings tab.

Jitter buffer Auto None Small Medium Large Default: Auto

Select the size of jitter buffer for your system.

G.729 payload size (ms)

10, 20, 30, 40, 50, 60 Default: 30

Assign the maximum required payload size for each codec, for the VoIP calls sent over H.323 trunks. As well, the Payload size on the IPT must match. For H.323 endpoints, the suggested payload size is 30 ms. Networks that support a mix of SIP and H.323 have less interoperability issues if both protocols use the same payload size.

Attention: Payload size can also be set for Nortel IP telephones. G.723 payload size (ms) 30 G.711 payload size (ms) 10, 20, 30, 40, 50, 60 Default: 30 Incremental payload size

<check box> If you select the check box, the system advertises a variable payload size (40, 30, 20, 10 ms).

Enable T.38 fax <check box> If you select the check box, the system supports T.38 fax over IP.

Attention: Fax tones that broadcast through a

Attention: telephone speaker may disrupt calls at other telephones using VoIP trunks in the vicinity of the fax machine. Here are some suggestions to minimize the possibility of your VoIP calls being dropped because of fax tone interference:

Locate the fax machine away from other telephones. Turn the speaker volume on the fax machine to the lowest level, or off, if that option is available.

Force G.711 for 3.1k Audio

<check box> Default: Unselected

If you select the check box, the system forces the VoIP trunk to use the G.711 codec for 3.1k audio signals such as modem or TTY machines.

Attention: You can use this setting for fax machines if T.38 fax is not enabled on the trunk.

Table 20 H323 Media Parameters field descriptions

Figure 39 SIP Settings tab

The following table describes the fields of the SIP Settings tab.

Table 21 SIP Settings field descriptions

Attribute Value Description

Telephony Settings Fallback to circuit- switched Enabled-All Enabled-TDM Disabled Default: Enabled-All

Your choice determines how the system handles calls if the IP network cannot be used:

• Enabled-All: All calls are rerouted over specified PSTN trunks lines.

• Enabled-TDM: All TDM (digital telephones) voice calls are rerouted over specified PSTN trunks lines.

• Disabled: Calls are not rerouted.

SIP Settings

Local Domain <alphanumeric> Local domain of the SIP network. Call signaling port Default: 5060 This is the listening port for the BCM.

Attention: If you change this value, the system restarts Functional Endpoint Proxy Server.

Actions

Modify 1. Click Modify on the SIP Settings panel to modify the Call Signaling Port. 2. Change the Call Signaling Port, and press OK. This dialog box warns you that if you change the Call Signaling Port value, the system drops all SIP calls and restarts FEPS. See Figure 35 on page 120.

The following figure shows the Modify dialog box used to modify the Call Signaling Port.

Figure 40 Modify Call Signaling Port

SIP Proxy

The following figure shows the SIP Proxy tab.

Figure 41 SIP Proxy tab

Dynamic Payload _96–127 Default: 120

Assign 0 to disable RFC2833 functionality.

Status <read-only> Indicates the status of the gateway, for example, Gateway is running.

Table 21 SIP Settings field descriptions

The following table describes the fields of the SIP Proxy tab.

The following table shows the Outbound Proxy Table values. Table 22 SIP Proxy field descriptions

Attribute Value Description

SIP Proxy

Domain <alphanumeric> This attribute is mandatory.

This is the SIP domain handled by the proxy. If it is also a DNS resolvable hostname of the proxy, a DNS lookup is done to route the messages.

Otherwise, an IP address should be provided in either the legacy routing box or in the Outbound Proxy table.

Route all calls using proxy

<check box> Default: Unselected

If you use the default, the system first checks the routing table before routing all SIP calls.

If you select the check box, the system uses the SIP Proxy for all SIP calls.

MCDN Protocol None CSE

Default: None

Use CSE to interoperate with other Nortel devices (BCM or CS 1000).

Optional IP Address for legacy routing

IP Address Format 0.0.0.0 <7-24>

This attribute is optional.

The system uses the IP Address and Port to route the message if the Outbound Proxy is not configured.

The IP Address and Port are used in message headers. If supplied, the IP Address is used in the maddr= section of message headers.

The system uses these attributes to interoperate with NRS. Port <numeric>

Default: 0

This attribute is optional.

If the port is 0, the system uses the well-known SIP port 5060.

Otherwise, the system uses the port you enter here.

Outbound Proxy Table Actions

Add 1. On the Outbound Proxy Table subpanel, click Add to add an entry.

2. In the Outbound Proxy table, type the information as described. See Outbound Proxy table field descriptions (page 113).

3. Click OK to add the entry.

Delete 1. On the Outbound Proxy Table subpanel, click an entry to delete. 2. Click Delete to delete the entry.