By default, Wowza Streaming Engine requires that RTMP-based encoders such Cisco Meeting Server provide a source username and password before they can connect to a live application and publish a live stream. Complete the following steps to create a source account and manage source authentication.
First to connect to Wowza Streaming Engine Manager, From the Admin-PC open a web browser and type the url http://10.1.5.128:8088.
In Wowza Streaming Engine Manager, click Server and then click Source Authentication.
Click Add Source.
Add Source UserName : cmsuser and Password : cisco. Click Add. This account will be used later on the cospaces of Cisco Meeting Server. This source account is used to authenticate connections from Cisco Meeting Server to live applications in Wowza Streaming Engine.
In the section Application , click Source Security, and then click Edit.
Under the source type RTMP, select Require password authentication.
Under Client Restrictions, You can control which IP addresses encoders can connect from. Click Save.
To connect the Cisco Meeting Server to Wowza Streaming Engine and publish a live stream, we need a stream URL.
To retrieve this URL, we need the following application connection settings:
- Server URL : rtmp://10.1.5.128:1935/live
- Stream Name : a unique name to identify the stream for example clcnf.
- Username : cmsuser
- password : cisco
On the Cisco Meeting Server we need to configure the stream URL in the following format instead:
In this scenario, it should be : rtmp://cmsuser:email@example.com:1935/live/clcnf
To retrive the stream URL’s informations, see in the Application Connection Settings information in the right of the web page and use this information in the CMS streamer ‘s configuration to connect it to Wowza Streaming Engine.
The new streamer component requires to listen to SIP connections , the streamer server must have a valid certificate for TLS connection.
Configure the listening interface of the streamer and the SIP TCP and TLS ports to listen on using the following command.
hq-cms> streamer sip listen a 6000 6001
Configure the certificates to be used for the SIP streamer. Specify the key file, certificate, and CA trust bundle.
hq-cms> streamer sip certs cmscert.key cmscert.cer Chain-CA.cer
Optionally select the quality for streaming
hq-cms> streamer sip resolution 720p
Enable the streamer. All message must show “SUCCESS” as below displayed below.
hq-cms> streamer enable
Verify the streamer status.
Configuring the Stream URI via the API.
Once the new SIP streamer is enabled, it can be configured and used in the Call Bridge using the sipStreamerUri API parameter specified in the API call profile object.
On the web GUI of the CMS navigate to Configuration > API.
In the Filter section, type “callprofile”
Click the Create new button to create a new CallProfile or edit the existing profile and populate the following information.
- streamingMode: select manual from the drop menu
Add the created callProfile above to the /system/profiles. This is a global configuration and the configured “sipStreamerUri” will be used for streamer operation.
Click return to object field.
In the Filter section, type “system”.
Click the View or edit button for system profiles.
Then select Choose Next to the field for callProfile, and then Select the callProfile previously created
In the Filter section, type “cospace”.
Click the object ID for jdoe Meeting Space.
Modify the space to add “StreamURL”. The ‘streamURL’ in the following format: rtmp://cmsuser:firstname.lastname@example.org:1935/live/clcnf
Configure an outboundDialPlan rule to match the domain used in streamerUri to route.
On CMS GUI navigate to Configuration > Outbound calls.
Create a new Outbound rules with the following informations:
- Domain: cms-str.lab.local
- SIP proxy to use: 10.1.5.42:6000
- Encryption: Unencrypted
- click And New
- Domain: cms-str.lab.local
- SIP proxy to use: 10.1.5.42:6001
- Encryption: Encrypted
- click And New
For SIP streamer, if default ports for SIP (5060,5061) are not used, then It is mandatory to specify ports in the configuration of the streamer and include the following port number to connect to the “sip proxy to use” field when outboundDialPlanRule is configured for the service.
From the jdoe-pc, access the jdoe space and click the Join button.
From the jsmith-pc, access the jsmith space, and click the Join a meeting button.
Enter the Meeting ID of the jdoe space 51001. Click the Join meeting button.
From the jdoe-pc, Click the Meeting Control Icon on the right-hand side of the video screen.
This will expand to show the Recording and Streaming Control. Click the Streaming button.
After a couple of seconds, notice the Streaming button will go to a solid blue dot to indicate the streaming has started.
To verify that Wowza Streaming Engine is receiving the published stream, complete the following steps in Wowza Streaming Engine Manager.
Click Incoming Streams in the the live application.
You should see the stream clcnf listed with a status of Active.
On the cmayer-pc, open the CMS Live Stream using VLC, enter the stream URL rtmp://cmsuser:email@example.com:1935/live/clcnf.
Meeting content is being streamed to cmayer-pc.
From the HQ-CMS, navigate to Logs > Event Logs. Verify that an outgoing SIP call is sent to CMS Streamer. The stream URL is sent via SIP header in the contact field.
Navigate to Status > Calls, verify that in addition to jdoe and jsmith calls, the sip streaming call is also connected.