JsSIP clients not entering freeswitch conference with 'conference_set_auto_outcall' - freeswitch

Question

I am having an issue with setting up a conference in FreeSwitch with JsSIP clients. I have the conference_set_auto_outcall set in the dialplan to call 1000 and 1001. When using MicroSIP (software SIP phone) everything works as expected and the 2 clients pick up the conference however when the originating phone is a JsSIP instance none of the clients are placed in the conference (running conference 23456 play <wav-file> doesn't play on any of them).
Checking the remote stream that is recieved by the remote clients shows the video track has no dimension details (height or width) and a framerate of 0.
Calls from the one web-app to the other work as expected and using originate <user> &conference(<conf-id>) to place all of the users into the conference also works as expected.
Any help would be greatly appreciated, please let me know if there is any other information that might be helpful!
The dialplan can be seen below
<extension name="Demonstrate Conference">
<condition field="destination_number" expression="^(23456)$">
<action application="answer" />
<action application="set" data="conference_auto_outcall_timeout=30" />
<action application="set" data="conference_utils_auto_outcall_flags=mute"/>
<action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
<!-- Set caller ID for incoming calls -->
<action application="set" data="conference_auto_outcall_caller_id_name=Emergency" />
<action application="set" data="conference_auto_outcall_caller_id_number=9876543210" />
<action application="set" data="conference_auto_outcall_profile=default" />
<!-- When the call initiator leaves the conference ends -->
<action application="set" data="[conference_member_flags=endconf]user/${caller_id_number}#$${domain}" />
<!-- Autocall 1000 and 1001 -->
<action application="conference_set_auto_outcall" data="user/1000#$${domain}"/>
<action application="conference_set_auto_outcall" data="user/1001#$${domain}"/>
<action application="conference" data="$1#default+flags{endconf}" />
</condition>
</extension>
Both instances share most of the same userAgent event detection
userAgent.on('newRTCSession', function (data) {
console.info('onNewRTCSession:', data);
//Call incoming
if (data.originator == 'remote') {
console.info("incomingSession, answer the call----------------------");
incomingSession = data.session;
var status = document.getElementById("status");
status.innerHTML = "Incoming call";
setTimeout(() => {
data.session.answer({
'mediaConstraints': get_constraints(), 'mediaStream': null
})
updateUI();
}, 1000);
}else {
console.info("outgoingSession");
outgoingSession = data.session;
outgoingSession.on('connecting', function (data) {
console.info('onConnecting-', data.request);
currentSession = outgoingSession;
outgoingSession = null;
});
}
data.session.on('accepted', function (data) {
console.info('onAccepted-', data);
if (data.originator == 'remote' && currentSession == null) {
currentSession = incomingSession;
incomingSession = null;
console.info("setCurrentSession-", currentSession);
}
});
data.session.on('confirmed', function (data) {
console.info('onConfirmed-', data);
if (data.originator == 'remote' && currentSession == null) {
currentSession = incomingSession;
incomingSession = null;
console.info("setCurrentSession-", currentSession);
}
});
data.session.on('sdp', function (data) {
console.info('onSDP, type-', data.type, 'sdp-', data.sdp);
});
data.session.on('progress', function (data) {
console.info('onProgress-', data.originator);
if (data.originator == 'remote') {
console.info('onProgress, response-', data.response);
}
});
data.session.on('peerconnection', function (data) {
console.info('onPeerconnection-', data.peerconnection);
data.peerconnection.onaddstream = function (ev) {
console.info('onaddstream from remote -----------', ev);
videoView.srcObject = ev.stream;
remote_stream = ev.stream;
};
});
data.session.on('ended', function (data) {
console.info('onEnded-', data);
if (data.originator == 'remote') {
console.info('onEnded, response-', data.response);
}
currentSession = null;
updateUI();
});
});
Full logs and HTML pages can be seen in the pastebins below
Logs
Call placer - https://pastebin.com/5HdjPADH
Call reciever - https://pastebin.com/64eb9jUT
FreeSwitch Logs - https://pastebin.com/p7MtTxvm
HTML Pages
Call placer - https://pastebin.com/f7E71wJh
Call reciever - https://pastebin.com/22wD9NRp
EDIT: If one of the client machines (1000) is a microSIP client, 1001 will be able to see the camera of 1000 however the call initiator will not be able to see anything.
EDIT 2: Having implemented the same pages in sipJS the same issue is occuring suggesting the problem might be with WebRTC in FreeSwitch.