What is WebRTC? WebRTC is a new videoconferencing standard being added to web browsers. By design, without any other software installed, WebRTC can enable video calls between computers using browsers. In this way, it can emulate (and even out-perform) the functions of programs like Skype and Facetime. Tell me more about the audio format used. The algorithm is called Opus, and has generated quite a lot of buzz in the IP telephone community. It has very low delay and rivals some of the best codecs in quality. And since it doesn t carry any licensing costs, it can be used in free applications like WebRTC. It delivers full bandwidth (20KHz) voice and music over data channels as low as 64 Kb/s. Sounds great! What are the limitations? Figure 1 Here s a secret: the engine to deliver WebRTC is already built into many browsers (maybe the one you re using now) and if you open a link to a WebRTC server, special javascript code can wake it up, and allow you to make or accept calls from other WebRTC users. OK, it s a great video conferencing tool. How does it affect my talk shows? If you re not interested in WebRTC s video capabilities, you can turn them off. But when used for audio, the default encoding algorithm used provides excellent fidelity and low delay. When you use this to talk to your Comrex STAC VIP, you can make high-quality calls into your station from most web browsers! First of all, not all browsers support WebRTC yet. Currently, on Mac and Windows, the list includes Google Chrome, Mozilla Firefox, and Opera. On Android, Chrome and Firefox support it. Plug-ins are available for you die-hard Internet Explorer users as well. What about my iphone? Sorry, but Apple has chosen not to participate in WebRTC ios support yet, and has restricted browser apps or plug-ins that try to support it. Hopefully they ll have a change of heart soon. So do I just make a WebRTC call to my STAC VIP? Not quite. STAC VIP doesn t natively speak WebRTC, but as of firmware 1.1p4, it does speak Opus. STAC VIP can encode, decode and deliver Opus over the SIP protocol (the standard protocol used for Voice over IP). And you can convert from WebRTC-Opus to SIP-Opus using free online conversion servers.
This sounds complicated. Explain. First, you ll create a new account at Getonsip.com It s not so bad. We re going to recommend use of a SIP provider called Getonsip.com. They will provide you free credentials that you can key into your STAC VIP. They will also provide you a web link you can post or email to anyone. Once your STAC VIP is registered with Getonsip, anyone who clicks the link in a compatible browser can connect to your station sounding great. Figure 2 As shown in Figure 2, Getonsip handles the SIP portion of the call (to the STAC VIP) and the WebRTC portion of the call (to the browser) and does a translation of signaling. But the Opus audio flows through untouched and is encoded/decoded on each side. Figure 3 You ll need to confirm your email address used, then log in to your Getonsip account using the email and password you provided. Can you take me through it? You ll need: 1) A computer with a microphone (built in is OK for testing, a USB headset will help performance a lot) 2) A Comrex STAC VIP with firmware 2.0 or higher with unassigned lines for incoming calls. Figure 4
You ll need to have at least a microphone on your computer in order to use the Getonsip.com dashboard. If the system can t detect a mic, it will give you this error until it s resolved: Next, click on the section called View Profile. Figure 7 Figure 5 Then you ll see the Getonsip dashboard page. This will display all your credentials that you will need in order to configure your STAC VIP. Copy all these. We re done with the Getonsip dashboard, so log out of your account. Now, log in to the web interface of your STAC VIP configuration page (as described in the manual). Figure 6 There s a couple of pieces of information you ll need to copy from this dashboard page. First, note the link that appears in the upper right corner under Receive incoming calls. Copy that and paste it somewhere safe (like an email to yourself). This is the link you will post or email to people when you want them to call into your station from their browser. Figure 8
Maneuver to the provider config page by selecting Line Configuration->VoIP Providers If all these parameters are correct, Choose Restart System. Once the system returns (about 10 seconds), check SIP Registration status as shown: Figure 9 Choose Add Provider->SIP Provider to create a new provider entry. Make sure the following entries are set correctly: 1) Name->WebRTC (or any name you like) 2) Color Code->(your choice, chooses banner color on caller interface page) Figure 10 If any error messages appear here, carefully check the credentials as they were input. Also, be sure UDP Port 5060 is not blocked for outgoing data. Now you must assign this provider to one or more unused lines, so calls can be received. This is done by backing out to the main configuration menu and selecting Line Configuration->Line Assignments. 3) SIP Provider->OnSIP 4) Account Username -> Username (from credentials, probably your login name) 5) Account Password-> SIP Password (from credentials, long string of characters) 6) Auth Username-> Auth Username (from credentials, probably getonsip_login name) 7) Proxy/Domain-> (Domain from credentials, probably getonsip.com) 8) Outgoing Enabled (your choice, WebRTC will be primarily incoming)
As shown in the example, the first four lines have been assigned to the Main Provider and the last two have been assigned to the WebRTC function. You may use as many lines as you like for WebRTC. You ll be presented with the calling screen, and the browser will ask for permission to use your microphone. (This may look differently on different browsers, but you must give permission for the call to work.) Figure 13 Key in a familiar name (this will appear on the incoming call name on the STAC VIP web interface) and press the phone icon to make the call. Figure 11 Assuming your STAC VIP is registered with Getonsip, try opening the web link they provided in a compatible browser, on a computer with an active microphone. Figure 12 Figure 14
The call should start ringing on the STAC VIP and indicate an Opus audio connection. The call can be ended by either end at any time. Figure 15