Sip unable to make and recieve a call

Asterisk Asterisk Support
1

Hi ,
I am new to the asterisk server . I have installed asterisk 11.11.0 on my centos .All are running on VMware .
I tried simple example of sending and receiving the call from the , example given in the website http://sipjs.com/guides/make-call/. I tried on the firefox browser.
. The user connected to the asterisk server but after few seconds web cam turned off .Moreover , call is also not receiving . Below is the stack trace of the browser .

Sat Oct 15 2016 18:48:26 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | ICE candidate received: candidate:5 2 UDP 1685856254 106.222.36.63 23548 typ srflx raddr 192.168.1.101 rport 61338sip.js (line 2884)Sat Oct 15 2016 18:48:30 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | RTCIceChecking Timeout Triggered after 5000 millisecondssip.js (line 2884)

Sat Oct 15 2016 18:48:30 GMT+0530 (India Standard Time) | sip.inviteclientcontext | closing INVITE session lp19supo8fnugjko5hre9p837cpuhesip.js (line 2884)Sat Oct 15 2016 18:48:30 GMT+0530 (India Standard Time) | sip.invitecontext.mediahandler | closing PeerConnectionsip.js (line 2884)

Sat Oct 15 2016 18:48:30 GMT+0530 (India Standard Time) | sip.transaction.ict | Timer D expired for INVITE client transaction z9hG4bK1339928

I donot know how to trouble shoot . Kindly assist.Below is the server log.

Pasted image1353×415 17.4 KB

Regards
Vivek Solanki

2

As you appear to be using WebRTC how have you configured the SIP peer in sip.conf? As well you will want to use a recent version of Asterisk as fixes have been made to support changes made to WebRTC as it has developed.