In traditional Unified CM telephony, ad-hoc conferencing is a call by which two participants are talking directly to one another when one party (using a device registered to the Unified CM) presses the Conference button, calls a different person, and after talking to that 3rd party, presses the Conference button again to join everyone into a 3-party conference. The exact steps may vary depending on the endpoint (e.g. Jabber vs. a physical phone endpoint), but it is always a non-scheduled escalation of a point-to-point call to a multi-party conference.
What makes this feature different than a scheduled or permanent conference is that an adhoc conference is not simply a SIP call to the CMS, as you will see from the configuration on the Unified CM. When the conference initiator presses the Conference button the second time to bring everyone into the same meeting, Unified CM must make an API call to CMS to create a conference on the fly, to which all the calls are then transferred. All this happens transparently to the participants.
This means that Unified CM needs to have the API credentials and Web Admin address/port configured, as well as a SIP trunk directly to the CMS server in in order to extend the call.
When required, CUCM can then dynamically create a Space on CMS, so that each call can be extended directly to CMS and match an Incoming call rule that targets Spaces. Therefore direct SIP trunks to each CMS node in the cluster are required on the Unfied CM. This is required for any Unified CM Integration.
You should see:
Note that the domain portion of the Request-URI in the SIP INVITE to a Call Bridge must match one of the configured inbound rules. In this lab, you configured the SIP Trunk Destination on the Unified CM server with the FQDN of the CMS server, cms1a.pod8.cms.lab for the case of cms1a. If the SIP trunk were configured with an IP address, for example, then you would need to add the IP addresses of your CMS servers to the list of domains in the Call matching rules and make sure Target Spaces is set to Yes.
In order to configure the Conference Bridge, a SIP trunk to CMS must exist. We'll configure that first.
For a secure SIP trunk, you first need to create a SIP Trunk Security Profile that allows for encrypted calls.
Now you can configure the SIP trunks from the UCM cluster to each of the CMS servers.
Parameter | Value |
---|---|
Device Name | CMS1A-SIP-TRUNK |
Device Pool | Default |
SRTP Allowed | Checked |
Run On All Active Unified CM Nodes | Checked |
Calling Search Space (under the Inbound Calls section) | CSS_INBOUND_CMS This Calling Search Space is used to limit the calls that can originate from CMS. CMS should be able to call other video endpoints, but should not have PSTN access in this deployment to avoid any potential toll-fraud issues. |
Calling and Connected Party Info Format | Deliver URI and DN in connected party, if available |
Destination | cms1a.pod8.cms.lab NOTE: You will only configure one destination per trunk. This is because later on, for the ad-hoc conferencing feature, Unified CM needs the ability to send an API message to a specific Call Bridge server and then use a SIP trunk that can only reach that specific server. Also, if for some reason you see 16 blank lines for entering a destination, this is a browser problem (there should only be 1 shown by default). Use the '-' button to remove the lines 2 through 16 before saving the SIP trunk. |
Destination Port | 5061 |
SIP Trunk Security Profile | Encrypted SIP Trunk Profile for CMS1 |
Rerouting Calling Search Space | CSS_CMS_REROUTING Note: This parameter is needed for Call Bridge Groups, to be discussed later. |
SIP Profile | LTRCOL-2250 SIP Profile |
Normalization Script | cisco-meeting-server-interop Note that this is Unified CM 11.5. In older versions of Unified CM, the cisco-meeting-server-interop script did not exist, but could be manually uploaded or you could use the older conductor-interop script that existed in older versions of UCM. |
Now you must repeat this configuration to add trunks pointing to cms1b and cms1c. If you would like to do so manually, continue below. If you would like to have these trunks created automatically for you, click HERE to have them added for you automatically. You should see another window pop up with the results. If the addition is successful, you can skip this section and move on to the Adhoc Conference Bridge Configuration for cms1a configuration. If you would like to configure these manually, click the button below to expand the detailed steps.
Parameter | Value |
---|---|
Device Name | CMS1B-SIP-TRUNK |
Device Pool | Default |
SRTP Allowed | Checked |
Run On All Active Unified CM Nodes | Checked |
Calling Search Space | CSS_INBOUND_CMS |
Calling and Connected Party Info Format | Deliver URI and DN in connected party, if available |
Destination | cms1b.pod8.cms.lab |
Destination Port | 5061 |
SIP Trunk Security Profile | Encrypted SIP Trunk Profile for CMS1 |
Rerouting Calling Search Space | CSS_CMS_REROUTING |
SIP Profile | LTRCOL-2250 SIP Profile |
Normalization Script | cisco-meeting-server-interop |
Finally, configure the trunk to cms1c.
Parameter | Value |
---|---|
Device Name | CMS1C-SIP-TRUNK |
Device Pool | Default |
SRTP Allowed | Checked |
Run On All Active Unified CM Nodes | Checked |
Calling Search Space | CSS_INBOUND_CMS |
Calling and Connected Party Info Format | Deliver URI and DN in connected party, if available |
Destination | cms1c.pod8.cms.lab |
Destination Port | 5061 |
SIP Trunk Security Profile | Encrypted SIP Trunk Profile for CMS1 |
Rerouting Calling Search Space | CSS_CMS_REROUTING |
SIP Profile | LTRCOL-2250 SIP Profile |
Normalization Script | cisco-meeting-server-interop |
You should now have three SIP trunks configured. If you navigate to Device > Trunk in Unified CM Administration and click Find, you should see the trunks go in service as the SIP OPTIONS messages are answered. This can take a couple of minutes. Sometimes the admin page will report that SIP OPTIONS are not enabled until it goes to Full Service.
Start by configuring a conference bridge resource on the Unified CM cluster pointing to cms1a by following these steps:
If you reload the web page after a few seconds, the status of Conference Bridge device should eventually indicate it is registered. You may have wait up to 30 seconds and reload the page for the status to update.
The registration status only indicates the health of the API connectivity between Unified CM and the CMS server. The status does not have anything to do with the status of the SIP trunk. To become registered, the Unified CM connects to the CMS using the address, port, and credentials supplied. It then sends a GET request to the destination configured on the SIP trunk on the configured port, to check the CMS' system status. For the conference resource you just configured, this would be: https://cms1a.pod8.cms.lab:8443/api/v1/system/status. If you like, you can check this status from your browser using the GET /system/status (username: admin with password: ) request. The results will something like this:
If there is no response, or the Call Bridge isn't active, or any other error is returned, the conference bridge resource will not show as being registered. There is also one other important consideration: Remember that this connection always uses HTTPS. For that reason, the Unified CM MUST trust the certificate, specifically the Web Admin certificate, that the CMS presents when Unified CM connects to it.
Now you must repeat this configuration to the other two CMS servers as adhoc conference bridge resources. If you would like to do so manually, continue below. If you would like to have these CFB resources created automatically for you, click HERE to have them added for you automatically. You should see another window pop up with the results. If the addition is successful, you can skip this section and move on to the Media Resource Group / List configuration.
Now repeat the same configuration for the other two CMS servers, starting with CMS1B
Finally, configure the adhoc conference bridge for CMS1C
If you go back to Media Resources > Conference Bridge, and click the Find button, you should see that all three conference bridges are registered. If you don't the the last one you just configured registered yet, wait a few seconds and click Find again. Ignore the two devices named CFB_2 and CFB_3. These are software conference resources that are not being used.
Now that you have configured the Conference Bridge resources, you must configure Unified CM to make them available to users. By default, any device not assigned to a Media Resource Group is available to any user, however it is best practice to assign media resources to a Media Resource Group (MRG) and Media Resource Group List (MRGL) and then use the list to control how those resources are used.
Start by following these steps to add the Media Resource Group:
Now that you have created the Media Resource Group, proceed to create the Media Resource Group List by following these steps:
The last step is to assign the MRGL to devices that are allowed to use it. You could apply this MRGL to a device directly, however in most cases, it is best to assign the MRGL to a Device Pool to make it available to all the devices in the Device Pool. Follow these steps to add the MRGL you created to the Default Device Pool that is being used by your devices.
Now that you have everything configured, you can test making an adhoc conference call. From the Jabber endpoint on your laptop (Pod8 User4), you will place a call to Pod8 User1 by dialing +19195080001. This is a Jabber client on one of your Remote Desktop sessions (PC1). After the call is connected, you will then call Pod8 User2 (Remote Desktop session to PC2) and join the two calls into a conference. There are a few ways to perform this task this in Jabber, but you'll use the way that most closely replicates the sequence you would use on a physical phone endpoint.
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You've finished this portion of the lab. We have multiple other optional lab sections, depending on your time and interest, to continue on.