Network Interface

For information only. This is the server interface used by the gateway service.

Local IP Address

For information only. This is the current IP address associated with the selected Network Interface.

Gateway Listen Port

This is the port on which the gateway listens for any incoming calls from the IP Office system. This setting is used when configuring an application server for operation with an IP500 V2.

SIP Trunk Listen Port

This is the port on which the gateway listens for SIP trunk connections from the IP Office system. Not currently used.

Logging Level

This sets the level of logging used by the gateway. The log files, prefixed WebRTCGateway, can be downloaded through the server's web control/platform view menus (LogsDownload). The default setting is Info.

Allow Origins

This field sets the domains and/or IP addresses from which the WebRTC gateway service will accept web socket (IP Office service) connections. Multiple entries can be added, each separated by ; semi-colon.

Configuration Mode

For Server Edition servers, the Automatic setting can be used. That automatically configures the gateway to match other IP Office service settings. For an application server, select Manual.

Domain Name

Set this field to match the domain name configured in the SIP Registrar settings of the IP Office system.

Private IP Address

Set this to the address of the IP Office system configured as the SIP registrar for WebRTC client users.

Private TCP Port

Private UDP Port

Private TLS Port

Set these fields to match the protocol ports configured for the SIP registrar on the IP Office.

Public IP Address

Leave this set to 0.0.0.0 to use the application server's IP address.

Public TCP Port

Public UDP Port

Public TLS Port

Use these fields to set the ports that should be used for each protocol by client applications.

Transport Type

Select the protocol that the gateway and clients should use. This must match the Layer 4 Protocol settings of the IP Office SIP Registrar .

RTP Port Range (Private)

These fields set the minimum and maximum RTP ports for connections between the gateway services and the IP Office system.

RTP Port Range (Public)

These fields set the minimum and maximum RTP ports for connections from the WebRTC clients. If supporting external clients, these ports should be allowed for routing to the gateway server in the customer's external firewalls. Ensure that these do not overlap with the RTP port range configured for the IP Office SIP registrar.

Codecs - Audio

Use this list to adjust the order of codec preference. It is recommended that both the PCM codec choices are kept at the top of the list.

Codecs - Video

Currently VP8 is the only supported video codec.

DTMF Payload Type

Default = 101

This field set the default value for RFC2833 payload negotiation. This value is used with clients and services that do not support dynamic payload negotiation.

STUN/TURN Settings

The following setting allow the media gateway to be used with external clients via STUN and TURN servers. If enabled, the settings need to match the STUN/TURN server. For details of doing this with an Avaya Session Border Controller for Enterprise, refer to the "IP Office SIP Phones with ASBCE" manual.

STUN Server Address

Default = 0.0.0.0 (Disabled)

The gateway service can use STUN to attempt to resolve issues caused by network address translation (NAT) being applied to traffic between it and external clients. The gateway attempts to use STUN if a STUN server address is set.

STUN Server Port

Sets the port used for connection to the STUN server. The default is 3478.

TURN Server Address

Default = 0.0.0.0 (Disabled)

The gateway service can use TURN to attempt to resolve issues caused by network address translation (NAT) being applied to traffic between it and external clients. Unlike STUN, all traffic is routed via a TURN server. The gateway attempts to use TURN if a TURN server address is set.

TURN Server Port

Sets the port used for connection

TURN User Name

TURN Password

Enter the name and password of the account on the TURN server if authentication is being used.