Hi I am using Asterisk 14.2.1 to run webrtc example provided in https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 I am able to connect to SIPML from asterisk server. But there is the following error seen on console when i try to generate a .call outbound call from Asterisk […

[image] bijumalla_latha: [May 23 15:50:58] ERROR[29341]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“df7jal23ls0d.invalid”, “(null)”, …): Name or service not known [May 23 15:50:58] WARNING[29341]: chan_sip.c:16667 _setaddress_from_contact: Invalid host name in Contact: (can’t resolve in …

To elaborate this further. Please see below asterisk configurations sip.conf [general] udpbindaddr=0.0.0.0:5060 realm=192.168.0.180 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss ; realm=202.65.140.19 ; externip=202.65.140.19 videosupport=yes srvlookup=yes …

[image] bijumalla_latha: [May 24 14:43:46] ERROR[18002][C-0000001a]: res_rtp_asterisk.c:2218 _rtprecvfrom: DTLS failure occurred on RTP instance ‘0xb4626cc4’ due to reason ‘certificate verify failed’, terminating You are either suffering a man in the middle attack, or you do not have a valid …

Installed the mentioned ca.crt on firefox. Anything else needs to be done? SIPML5 is running on Firefox

When it comes to WebRTC you have to learn and gain this knowledge, you can’t depend on help like this for ever as it is complex and deploying it is not easy.

Just a heads up and someone can correct me if I am wrong, but I don’t think browsers (at least Chromium and Firefox) allow insecure websocket connections anymore, so you’ll also need to set up the tls bindings in Asterisk http.conf (note example config only below): tlsenable=yes tlsbindaddr=0.0.0.0…

Thanks, now i could see the call being received on Browser. But still no RTP seen, Audio and Video is failing. See below updated sip.conf, http.conf, rtp.conf. http.conf [general] enabled=yes bindaddr=192.168.0.180 bindport=8088 tlsenable=yes ; enable tls - default no. tlsbindaddr=0…

[image] Troubleshooting WebRTC Issues Asterisk Support Here is a little guide to troubleshoot webrtc issues with Asterisk. Always try to use the latest WebRTC API with the latest Asterisk branch(11 or 12). Be sure you have the icessuport enabled in the rtp.conf ISSUE: I …

Webrtc errors and no audio established

Asterisk Asterisk Support

navaismo May 26, 2017, 8:20am 9

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Webrtc errors and no audio established

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