InstaCall operates on that premise, continued Wolpov. to us and use the service anywhere,” he said. “Customers can outsource the smarts of a traditional phone system - the extension dialing, the routing, etc. The company’s offerings are aimed at small to medium-sized businesses that want to avoid the acquisition and ongoing maintenance expense of a corporate phone service, Rob Wolpov, OnSIP’s cofounder, told CRM Buyer. OnSIP introduced InstaCall along with its OnSIP Network, a signaling Platform as a Service for developers to build WebRTC applications. Google launched WebRTC as an open source project it supports audio, video and data transfer applications in the browser. Sip-0.9.2.It is a Web Real-Time Communications, or WebRTC, application that allows a company to add a button for service or sales assistance to a website or mobile device via the browser. Sip-0.9.2.js:807 Mon 22:32:19 GMT+0000 (GMT Standard Time) | | acquired local media streams PeerConnection.onaddstream sip-0.9.2.js:8977 Logger.(anonymous function) sip-0.9.2.js:818 LoggerFactory.(anonymous function) sip-0.9.2.js:824 Sip-0.9.2.js:807 Mon 22:32:19 GMT+0000 (GMT Standard Time) | | Using deprecated stream API Sip-0.9.2.js:807 Mon 22:32:19 GMT+0000 (GMT Standard Time) | | New peer connection created Sip-0.9.2.js:807 Mon 22:32:19 GMT+0000 (GMT Standard Time) | sip.dialog | dialog changed to CONFIRMED state I notice the following in the webrtc-internals: I think it is a problem getting the packets to the local speaker !! I think you are right, it is not an ICE/STUN issue at all. I use the speaker and microphone on the webrtc client device (a laptop) all the time and they work. Using the chrome internals and looking at the stats tables under "CONN-AUDIO-1-0) GoogCandidatePair - I can see the bytesreceived/bytessent counts both incrementing while the call is live. The googRemoteAddress is (.92) which is the asterisk machine and it is bridging the call between the deskphone and the webrtc client. The googLocalAddress is the local machine (.211). I also ran chome://webrtc-internals and that shows me the active connection. 211 machine and I can see the RTP packets on the LAN going from. I looked at the rtp debug output on Asterisk it is relaying the rtp packets between (.121) the IP phone. I am new to webrtc so apologies if I am missing something very obvious. Should there be a STUN server on the local LAN so that it would provide a direct route to the desk ip phone since there is one ? Or, is there any way to say to the webrtc client - don't bother with STUN as this session is not going to traverse a NAT device so it will not be needed? It does not appear to have any effect anyway. I have seen the stunServers parameter in some versions of sipjs, but not in more recent ones. Replace this with the password from your sip.conf file Replace this with the username from your sip.conf file and replace the port with your Asterisk port from the nf file Uri: ' Replace this IP address with your Asterisk IP address, Replace this IP address with your Asterisk IP address Is there a way to stop the STUN/ICE process happening with sip-0.8.4.js or sip-0.9.0.js? It doesn't need to as the asterisk, the ip deskphone and the browser running the webrtc app are all on the same LAN. I believe the ICE/STUN interaction is happening on the browser side and the external WAN IP address is being presented as an candidate in the SDP to the asterisk and it is using this as the remote audio location. I have also registered the browser containing the WEBRTC button.īoth the IP phone and the browser running the simple webrtc button are on the same LAN.Įverything works as I would expect - the browser registers on the asterisk, I can click the INVITE button on the simple browser webrtc application and the desk phone rings, I answer the phone and the call is setup successfully but I can only hear audio going from browser to deskphone - I cannot get audio going the other way. I have set up an asterisk server on the LAN with a standard desk IP phone registered to the asterisk server. I have set up a simple WEBrtc demo using the onsip developer demo "Embed a WebRTC" button.
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