Asterisk sip.js remote call

Asterisk Asterisk Support
1

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?

2

Check the debug on both sides the chrome is debug and the asterisk pjsip or sip debug. See if the RTP is using a wrong address, if so, check your Nat and stun settings.

3

What is correct settings for nat and stun? Need I turn server for remote call?

4

[quote]What is correct settings for nat and stun? [/quote]Depends on your setup.

[quote] Need I turn server for remote call?[/quote] No, isn´t required.

5

Thank you for your help. I try to explan my scenario.

My asterisk is in global. I have one softphone(zoiper) and one sip.js client, they connect to asterisk by global IP.

sip.js code is here

[code] 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();
});[/code]

Can you tell me simple configuration for this scenario?