I am having Asterisk 16.30.0 installation in a EC2 instance.
Asterisk is installed using https://linuxhint.com/install-asterisk-ubuntu-22-04/
Below is the configuration for my Asterisk server running in EC2.
http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pem
sip.conf
[general]
tcpenable=yes
udpenable=yes
tcpbindaddr=0.0.0.0
udpbindaddr=0.0.0.0
bindaddr=0.0.0.0
bindport=5060
realm=127.0.0.1 ; replaced with static IP of EC2
transport=ws,wss
nat=yes
externip=127.0.0.1 ; replaced with static IP of EC2
media_address=127.0.0.1 ; replaced with static IP of EC2
directmedia=no
dtlsenable=yes
webrtc=yes
dtlssetup=actpass
rtcp_mux=yes
disallow=all
allow=ulaw,alaw
[friends_internal](!)
type=friend
host=dynamic
context=from-internal
transport=ws,wss
tcpenable=yes
nat=yes
avpf=yes
force_avp=yes
directmedia=no
qualify=yes
icesupport=yes
encryption=yes
dtlsenable=yes
dtlscipher=ALL
dtlsverify=fingerprint
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass
rtcp_mux=yes
disallow=all
allow=alaw,ulaw
[1000](friends_internal)
username=1000
callerid=1000
secret=somepass
[2000](friends_internal)
username=2000
callerid=2000
secret=somepass
extension.conf
[default]
exten => 1000,1,Dial(SIP/1000)
exten => 2000,1,Dial(SIP/2000)
rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
turnaddr=turn:myturnserver
turnusername=username
turnpassword=pass
The required keys are generated using the command.
./ast_tls_cert -C ec2-ip -O "ec2-ip" -d /etc/asterisk/keys
Required access provided to asterisk user to access keys.
The sip.js 0.16.0 client application running in a Google Chrome browser.
<!DOCTYPE html>
<head>
<script src="https://sipjs.com/download/sip-0.16.0.min.js"></script>
<script>
function sipTest() {
const userUri = SIP.UserAgent.makeURI("sip:1000@ec2-static-ip");
const userAgent = new SIP.UserAgent({
uri: userUri,
logLevel: "debug",
authorizationPassword: somepass,
authorizationUsername: '1000',
rel100: false,
hackViaTcp: false,
hackIpInContact: true,
hackWssInTransport: false,
transportOptions: {
traceSip: true,
server: "wss://ec2-static-ip:8089/ws"
},
delegate: {
onInvite,
onConnect,
onDisconnect
},
sessionDescriptionHandlerFactoryOptions: {
constraints: {
audio: true,
video: false
},
peerConnectionOptions: {
rtcConfiguration: {
iceServers: [
{
urls: "stun:stun.l.google.com:19302",
},
{
urls: "turn:myturn-server",
username: "username",
credential: "password",
}
],
},
},
},
});
const registerer = new SIP.Registerer(userAgent);
userAgent.start().then(() => {
console.log('UserAgent: start');
registerer.register();
});
// Setup myRegisterer state change handler
registerer.stateChange.addListener((newState) => {
switch (newState) {
case SIP.RegistererState.Registered:
console.log('Registered...');
break;
case SIP.RegistererState.Unregistered:
console.log('Unregistered...');
break;
case SIP.RegistererState.Terminated:
console.log('Terminated...');
break;
}
});
function onInvite(invitation) {
console.log('onInvite: invitation: ', invitation);
const callSession = invitation;
// Setup session state change handler
callSession.stateChange.addListener((state) => {
switch (state) {
case SIP.SessionState.Initial:
console.log('SessionState.Initial');
break;
case SIP.SessionState.Establishing:
console.log('SessionState.Establishing');
break;
case SIP.SessionState.Established:
console.log('SessionState.Established');
// We need to check the peer connection to determine which track was added
var pc = callSession.sessionDescriptionHandler.peerConnection;
console.log('PC: ', pc); // Valid
console.log('PC getReceivers: ', pc.getReceivers()); // Empty
console.log('PC getRemoteStreams: ', pc.getRemoteStreams()); // Empty
// Gets remote tracks
var remoteStream = new MediaStream();
pc.getReceivers().forEach(function(receiver) {
if (receiver.track) {
remoteStream.addTrack(receiver.track);
}
});
// I did try both the ways
const remoteAudio = new Audio();
// const remoteAudio = document.getElementById('remoteAudio');
remoteAudio.srcObject = remoteStream;
remoteAudio.play();
break;
case SIP.SessionState.Terminating:
case SIP.SessionState.Terminated:
console.log('SessionState.Terminated');
break;
default:
debugOut("Unknown session state.");
}
});
invitation.accept();
}
</script>
</head>
<body onload="sipTest()">
<audio id="remoteAudio"></audio>
</body>
</html>
Issues I am facing:
No audio is heard on both the ends.
On the client side:
When call is in SessionState.Established I get `pc.getRemoteStreams()` as empty and also `pc.getReceivers()` is empty in the Sip.js client.
And because of that I am not able to add streams to the Audio element.
On the server, I get below error after call is connected.
**When call is established and connected:**
WARNING[84463] chan_sip.c: Retransmission timeout reached on transmission xxxxxx-xxxxxxxx for seqno 123456789 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
WARNING[84463] chan_sip.c: Hanging up call xxxxxx-xxxxxxxx - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
**Also below errors:**
ERROR[92854] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
ERROR[92854] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
ERROR[92854] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
ERROR[92854] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
I have tested the stun and turn serves and they are appropriate resolution and I get relay server IP. Also EC2 instance security group provides access to all the UDP and TCP ports. For testing I have accepted the self signed certificate in using HTTPS Url from the browser where client app is running.
Amazon EC2 service is NATed service, you have private IP.
So if you have any issues with sound, you should ensure you have setuped external IP correctly and RTP ports(10000-20000 usually) are open in amazon firewall.
For see exact issue do
ps nat=yes is not correct for asterisk 1.8+. Use nat=force_rport,comedia