This document describes how to configure the H.323 Gateway in a Cisco Meeting Server (CMS) or an Acano server deployment. The H.323 Gateway was added in version R1.7 and allows to receive/send H.323 calls.
There are no specific requirements for this document.
This document is not restricted to specific software and hardware versions.
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
The Acano solution is very modular, let's discuss two common deployments:
Single combined server deployment:
Scalable and resilient deployment:
Step 1. On the Acano server Command line interface (CLI)
1. Secure Shell (SSH) to the MMP with the use of the admin credentials.
2. Configure the interface on which the H.323 Gateway should listen for H.323 calls:
For example, if you chose interface a to be the listening interface, then use this command:
h323_gateway h323_interfaces a
3. Configure the interface on which the Gateway listens for incoming SIP calls from the Call Bridge:
Note: The Gateway can listen on the same interface for both SIP and H.323 calls from the Call Bridge.
h323_gateway sip_interfaces a
4. Configure the port for the SIP interface to listen for SIP connections from the Call Bridge, by default the Gateway listens on port 6061:
h323_gateway sip_port 6061
Note: If the call bridge and the H.323 Gateway are colocated on the same server, you can change the Gateway's SIP port from 6061 to other values other than 5061.
It is recommend to deploy the H.323 Gateway with a Gatekeeper. This is because the Gatekeeper is responsible for the further call routing as the Gateway is limited in this functionality.
If your deployment doesn't include a Gatekeeper, omit this step.
5. Configure the nexthop of the H.323 Gateway. The nexthop should be the Gatekeeper's (for example, VCS-C) IP address:
h323_gateway h323_nexthop <IP_address>
6. Configure the SIP Proxy. The SIP Proxy is the part of the deployment that handles the SIP call leg in the H.323-SIP call.
If the Gateway and the SIP Proxy are on the same server, the IP address used must be 127.0.0.1, for example:
h323_gateway sip_proxy 127.0.0.1
If not, this should be the IP address of the Call bridge used as the SIP Proxy.
h323_gateway sip_proxy <IP_address>
7. Assign the certificate to be used by the H.323 gateway. This is required as the gateway always connects to and accepts connection from the Call bridge securely. For this reason the gateway needs to verify the Call Bridge certificate, so this needs to be in the H.323 Gateway's trust store.
"[<cert-bundle>]" in the command allows to add the CB certificate onto the Gateway's trust store. If you have multiple call bridges, this cert-bundle needs to contain the certificates of all the call bridges in the deployment.
Use this command to configure the certificates to use:
8. The H.323 SIP domain is appended onto outbound interworked calls from the H.323 gateway. If this is not set, the far-end would see the calling SIP URL as the username/DN@IP-address of H.323 gateway.
Set the H.323 SIP domain with this command:
h323_gateway sip_domain <domain>
9. Enable the H.323 Gateway component with this command:
Step 2. On the Call Bridge WebUI:
1. Connect to the WebUI of the Call Bridge with the admin credentials.
2. Single combined server deployment:
a Go to Configuration > Outbound calls b. Configure the destination domain for example h323.vc.alero.local c. Under SIP Proxy to use, set the loopback IP and SIP Port confirgured, for example 127.0.0.1:6061 d. Under Local from domain use the domain of call bridge.
3. Scalable and resilient deployment:
a Go to Configuration > Outbound calls b. Configure the destination domain for example h323.example.com c. Under SIP Proxy to use, set the IP and SIP Port confirgured, for example 10.48.36.76:6061 d. Under Local from domain use the domain of call bridge
Call Flow Example
This example details a typical call flow in a Scalable and resilient deployment. The same is true for a Single combined Server deployment, except for the SIP Proxy address being 127.0.0.1.