I am working with Asterisk 12 and sip.js . I am trying to call chrome browser from zoiper (android phone )
my pears are
[6004]
context=default
secret=6004
type=friend
host=dynamic
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1060 ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
videosupport=no
nat=force_rport,comedia
and My JS code is
var session;
var mediaStream;
var audio = new Audio('ring.mp3');
var config = {
// Replace this IP address with your Asterisk IP address
uri: 'sip:1060@XXX:9091',
// Replace this IP address with your Asterisk IP address,
// and replace the port with your Asterisk port from the http.conf file
ws_servers: 'ws://XXX:9092/ws',
// Replace this with the username from your sip.conf file
authorizationUser: '1060',
// Replace this with the password from your sip.conf file
password: '1060',
// HackIpInContact for Asterisk
hackIpInContact: true
};
var ua = new SIP.UA(config);
ua.on('invite', function(incomingSession) {
session = incomingSession;
audio.play();
prepareToanswer();
});
I can get invite but when I accept it , i can not get audio stream. Can anybody help me?