![]() ![]() To define the logging settings for the policy, configure the settings in the Logging section.įor more information, see Set Logging and Notification Preferences.See Configure Static NAT (SNAT) and Configure Server Load Balancing. You can also configure static NAT or configure server load balancing.Connections are - Specify whether connections are Allowed, Denied, or Denied (send reset) and define who appears in the From and To list (on the Policy tab of the proxy definition).You can use the settings on this tab to set logging, notification, automatic blocking, and timeout preferences. The Settings tab also shows the port and protocol for the policy, as well as an optional description of the policy. On the Settings tab, you can set basic information about a proxy policy, such as whether it allows or denies traffic, create access rules for a policy, or configure static NAT or server load balancing. You must disable NAT on your VoIP devices if you configure an H.323 or SIP-ALG. The H.323 and SIP-ALGs also perform this function. Many VoIP devices and servers use NAT (Network Address Translation) to open and close ports automatically. This allows softphone users to see peer status. ![]() Supports SIP presence through the use of the SIP Publish method.Generates log messages for auditing purposes.Makes sure that VoIP connections use standard SIP protocols.Opens the ports necessary to make and receive calls, and to exchange audio and video media.You do not have to complete any additional configuration steps to use IM with the SIP-ALG. ![]() The SIP-ALG supports page-based instant messaging (IM) as part of the default SIP protocol. This can help you to troubleshoot any problems. We recommend you make sure that VoIP connections work successfully before you add an H.323 or SIP-ALG. It can be difficult to coordinate the many components of a VoIP installation. It can support both the SIP Registrar and the SIP Proxy when used with a call management system that is external to the Firebox. The WatchGuard SIP-ALG supports SIP trunks. The WatchGuard SIP-ALG opens and closes the ports necessary for SIP to operate. Together, these components manage connections hosted by the call management system. In the SIP standard, two key components of call management are the SIP Registrar and the SIP Proxy. The call management system can be self-hosted, or hosted by a third-party service provider. In a peer-to-peer connection, each of the two devices knows the IP address of the other device and connects to the other directly without the use of a proxy server to route their calls.Ĭonnections managed by a call management system (PBX). It is important to understand that you usually implement VoIP with either: ![]() For instructions to add the SIP-ALG to your Firebox configuration, see Add a Proxy Policy to Your Configuration.įor supported deployment configurations, see Example VoIP Network Diagrams. Before you configure SIP-ALG, you must create a proxy policy to handle the traffic. There is no default policy for SIP-ALG traffic. To determine which ALG you need to add, consult the documentation for your VoIP devices or applications. You can use both H.323 and SIP-ALGs at the same time, if necessary. H.323 is commonly used on videoconferencing equipment. These ALGs have been created to work in a NAT environment to maintain security for privately-addressed conferencing equipment behind the Firebox. An ALG is created in the same way as a proxy policy and offers similar configuration options. I've noticed also that on Watchguard there is a SIP ALG which is equivalent to a Cisco Fixup Protocol, but I think that just facilitates opening of ports, not prioritizing traffic.If you use Voice-over-IP (VoIP) in your organization, you can add a SIP (Session Initiation Protocol) or H.323 ALG (Application Layer Gateway) to open the ports necessary to enable VoIP through your Firebox. I can do that, but I'd like to investigate further before we go that route. Problem is that users complain of call quality issues.Īt the moment I have setup 'Traffic Management' which uses a firewall rule and allows you to allocate/dedicate bandwidth, so I've set aside 512Kbps (10 users at site, maybe 6 calls active at once top) so this should be totally overkill.Ĩx8 says that if I can't disable SPI on the firewall, then they recommend a second firewall/gateway that doesn't use SPI, and point all phones to it. We are using Virtual Office solution from 8x8 which is a hosted SIP PBX solution. We have a site which has a 10Mb (up and down) circuit using a Watchguard x330 running 11.6.1 software. ![]()
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