Update 7/30/14: A new, easier way to do all this has now been posted, along with technotes for ACCESS/BRIC-Link and STAC-VIP.
I've been teasing Opus and WebRTC for awhile, and I'll have a lot more to say about it after NAB. But until then, let me leave you with an experiment. This will allow you to run a remote broadcast from pretty much any computer (or Android phone) to your ACCESS or BRIC-Link codec using the Opus algorithm via WebRTC. It's also a great exercise in getting familiar with WebRTC. Here's the instructions:
1) You'll need to upgrade your codec to firmware 3.0. Beware this is an unreleased beta test version, so there could be unknown bugs. ACCESS Rack firmware is here and BRIC-Link firmware is here. Instructions on how to upgrade are on our support page.
2) Go to http://ekiga.net, and apply for a free SIP address. You'll be emailed a username and password. Note we would normally recommend our favorite free SIP provider, getonsip.com, but their addresses are incompatible with this service.
3) Enable your codec to accept NACIP/SIP calls. This is the option that lets you accept calls from other codec brands. It's found in the System Settings Tab (advanced options) under N/ACIP SIP Settings->Accept Incoming Connections. Click "enabled". (Hint--uncheck this after use if you don't want your codec to answer random incoming SIP calls)
4) Also on the web UI, enter the SIP parameters for your ekiga.net account:
* Use SIP Proxy -> enabled
* SIP Proxy address-> ekiga.net
* SIP username -> <ekiga supplied username>
*SIP Password -> <ekiga supplied password>
Once applied, you can check registration status on the Connections tab. SIP Status should read "online (200 OK)".
5) Go to http://click2dial.org and choose "sign up". Give a valid email address, and you'll receive an email with two pieces of information:
* Your login password
* A web link. Copy that and store it somewhere. This is what you will send to anyone you want to connect to your codec.
You'll need to validate your click2dial account via the link provided.
6) Go back to click2dial.org and log in with your email address and password. Go to "My Account", and under step 1, choose "add a SIP address". Enter the ekiga address in the following format: sip:<user_name>@ekiga.net
7) The only other thing to worry about is whether the SIP RTP ports to your codec are open. If you're on the public Internet with no router or firewall, you're good to go. If not, and your router isn't kind to SIP traffic, you may need to forward UDP 6014 and 6015 to your codec.
8) To connect, you'll need a computer with a speaker and a microphone. WebRTC won't function without a mic on the dialing computer. A USB headset works best, but WebRTC also has a built-in echo canceller if you're using a laptop speaker/mic. Open Chrome or Firefox and go to the link provided by click2dial. You'll be prompted for permission to use the mic. When you allow that, you should establish a nice, low delay, full-duplex Opus audio link to your codec! Even works on Firefox and Chrome for Android. Sorry, no support for iPhone (ask Apple to support WebRTC!)
Let's have a little fun with this. The first user who successfully completes an Opus call from a browser, and sends us a screenshot of the codec's web UI showing a successful Opus call, gets some nifty Comrex swag. Send your screenie to techies@comrex.com. I'll make an update here when we've gotten the winner.
Update 5/5/14: The challenge has been met! Also I updated the Comrex firmware links to the latest version, which fixes a few WebRTC-related bugs.
Update 7/30/14: A new, easier way to do all this has now been posted, along with technotes for ACCESS/BRIC-Link and STAC-VIP.
No comments:
Post a Comment