WebRTC Outbound Calls

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azrobert:
Usetheforceobiwan showed us how to call our OBi's via WebRTC.
See: http://www.obitalk.com/forum/index.php?topic=9234.msg61177#msg61177

If you log into your GetOnSip account you can pass an outbound number to your OBi. Add code to your X_InbounCallRoute to bridge the call out a provider. I had to use STUN to fix a one-way audio problem. I also port forward the UserAgent and RTP ports.

Log into your GetOnSip account using Chrome here:
https://www.getonsip.com/webrtc/

Enter  18005551212@xx.xx.xx.xx:5061
Click the phone symbol.

Replace xx.xx.xx.xx:5061 with your public IP address and OBi trunk port number.

This worked using Chrome on Window 7 and Android.

QBZappy:
Using these WebRTC Outbound Calls methods with the obi as a sip/pstn gateway with the ability to make audio/video calls basically makes voip apps not relevant anymore. The obi used as a gateway can ring any other endpoint reproducing somewhat what Google Voice does. This might be interesting for people who can not obtain a GV number directly.

Usetheforceobiwan:
The only caveat to this is with respect to one stage dialing for calls from the Getonsip web portal  via the non-login method as referenced above.  While the name you have to enter in order to place the call is forwarded to the Obi as the CNAME (Peer Name), the CID (Peer Number) comes through as "anonymous".  Last time I messed with this, and unless something has changed recently, you cannot route calls via Peer Name, it can only be done with Peer Number.  So in order to facilitate one stage dialing from non-login Getonsip calls,  you have to be prepared to permit one stage dialing for all calls inbound.  At least that is how I remembered it was when I messed around with this 6 months ago.

Edit:  I just remembered another issue (duh), because the logged-in or non logged-in Getonsip web portal does not have facilities to generate DTMF tones, you cannot place calls via the Auto Attendant because you have no way to generate the dialing codes the AA needs.  So the only way to set up redirects through the AA via your Getonsip account is to call your Gos account from another Gos account (login required) which will generate a non anonymous CID which you can route to a speed dial or other pre-defined end point.

Usetheforceobiwan:
I just did a search for an online HTML-5 base WEBRTC-SIP client (to allow DTMF dialing) and it appears one does not exist.  While it should be possible, beyond prototypes and demo's, evidently the capability does not exist currently.

azrobert:
I logged into my GetOnSip account for the above test.
My OBi call history shows my GetOnSip username as Peer Number.
I did not test for Peer Number in my testing.
I bridged all inbound calls out SP1, but since it shows in the call history I'm sure Peer Number can be tested for routing.

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