ejabberd WebRTC integration pairs ejabberd, an Erlang-based XMPP server, with WebRTC for peer-to-peer audio and video. ejabberd handles signaling, presence, and authentication through the XMPP Jingle extension, while WebRTC transports media directly between browsers, with STUN/TURN handling NAT traversal. For multi-party video, pair ejabberd with an SFU or use a managed platform like VideoSDK that bundles signaling, media, and TURN into one SDK.
Real-time communication has quietly become table stakes for modern applications. Telehealth platforms, live customer support, collaborative education tools, and social apps all need sub-second audio and video, and developers keep reaching for the same two building blocks: WebRTC for media transport and a battle-tested signaling server to coordinate it. That is where ejabberd WebRTC integration enters the picture.
ejabberd, written in Erlang and maintained by ProcessOne, has powered large-scale XMPP deployments for two decades. Its per-process concurrency model handles enormous numbers of simultaneous connections, which is exactly what signaling-heavy video applications demand. WebRTC supplies the peer-to-peer media layer, but it deliberately leaves signaling undefined, so you need a server to exchange session descriptions and relay connectivity candidates. ejabberd, with its mature XMPP implementation and the Jingle extension, is a strong candidate for that job.
By the end of this guide you will understand the full ejabberd WebRTC architecture, how to prepare the server, how STUN/TURN solves NAT traversal, how Jingle signaling flows end to end, and what it takes to run the stack in production.

What is ejabberd and how does it relate to WebRTC?

ejabberd is defined as an open-source, Erlang-based messaging server that implements XMPP for real-time chat, presence, and multi-user rooms, with additional support for MQTT and SIP. It is known for clustering, hot code upgrades, and the ability to sustain hundreds of thousands of concurrent connections on modest hardware, a direct consequence of Erlang's lightweight process model.
WebRTC, by contrast, is a browser and mobile API standard for peer-to-peer audio, video, and data exchange. It handles media capture, encoding, encryption, and transport, but it does not define how two peers find each other or negotiate a session. That negotiation, exchanging offers, answers, and ICE candidates, must travel over a signaling channel you build yourself.
The synergy is straightforward: ejabberd handles signaling, presence, and authentication while WebRTC transports media. ejabberd already gives you user accounts, TLS-encrypted client connections, roster management, and multi-user chat rooms, which map naturally onto video calling concepts like participants, presence, and conference rooms. Developers who have shipped XMPP-based products consistently find that reusing ejabberd's identity and presence layer removes a large chunk of the plumbing a custom signaling server would require.
If you would rather not assemble signaling, TURN, and media routing yourself, a managed video calling SDK abstracts the entire stack behind a room-based API.

Architecture Overview: How ejabberd WebRTC Fits Together

A complete ejabberd WebRTC deployment separates concerns cleanly: browsers exchange media directly (or through a relay), while ejabberd carries only the lightweight signaling traffic. Understanding which component owns which responsibility is the key to debugging and scaling the system later.
The flow works like this. Browser A wants to call Browser B. A generates a WebRTC offer (an SDP payload describing codecs and capabilities) and sends it as a Jingle message through ejabberd to Browser B. B responds with an answer, and both sides exchange ICE candidates describing possible network routes. Once connectivity is established, media flows directly between the browsers, or through a TURN relay if direct paths fail, or through an SFU/MCU for group calls. ejabberd never touches the media itself.
Each component has a distinct responsibility:
  • Client browsers capture, encode, encrypt, and send media using WebRTC, and generate the SDP offers and ICE candidates that must be signaled.
  • ejabberd authenticates users, routes Jingle signaling messages, tracks presence, and can host room state for multi-party calls.
  • STUN/TURN servers solve NAT traversal: STUN helps peers discover their public address; TURN relays media when direct paths are blocked by firewalls or symmetric NATs.
  • SFU/MCU (optional) routes media for group calls, since full mesh peer-to-peer degrades quickly beyond four or five participants.
The critical insight: signaling volume is tiny compared to media volume. A signaling server can be lightweight even when the calls it coordinates carry gigabits of video, because the media never passes through it.

Preparing ejabberd for WebRTC

Preparing an ejabberd server for WebRTC work means enabling the right modules, exposing the right listeners, and locking down authentication. The goal is a server that can carry Jingle signaling and help clients with NAT discovery.

Choose your installation path

You have two main options. A source installation from the ejabberd repository gives you the newest modules and full control, at the cost of managing Erlang dependencies yourself. A containerized deployment is faster to stand up and easier to reproduce across environments, which is why most teams prototype in containers and graduate to source builds when they need custom modules. Both paths end at the same place: a running ejabberd node with an editable configuration file.

Enable the required modules

Three module areas matter for WebRTC. The Jingle-related support lets ejabberd route session negotiation stanzas between participants. The STUN service lets the server itself answer address-discovery requests, so a small deployment can skip a separate STUN box. The TURN service, if you enable it on ejabberd, relays media for clients behind restrictive networks; many teams instead run a dedicated TURN implementation such as coturn alongside ejabberd, which keeps media relay off the signaling node entirely.

Configure listeners and TLS

Your configuration must expose the standard client-to-server port for XMPP, ideally wrapped in TLS from the very first test, because browsers treat WebRTC as secure context and mixing plaintext signaling into a secure app invites trouble. Declare your server certificate, enable the authentication backend you plan to use (internal database for prototypes, external or LDAP-backed for production), and make sure the STUN/TURN listener section references valid credentials if you run TURN on the same box.
A practical tip from production deployments: keep a staging node with the same configuration shape as production. Configuration drift between a laptop prototype and a cloud deployment is one of the most common sources of "it worked locally" ICE failures.

Configuring STUN/TURN for NAT Traversal

STUN and TURN are not optional extras; they are what make video calls work across the real internet. Roughly 10 to 20 percent of real-world connections cannot establish direct peer-to-peer paths because of symmetric NATs or corporate firewalls, and without a TURN relay those calls simply fail.
STUN (Session Traversal Utilities for NAT) lets a client ask a server "what does my address look like from the outside?" The client uses that answer to build ICE candidates. TURN (Traversal Using Relays around NAT) goes further: when no direct path works, the client sends media to the TURN server, which relays it to the peer. TURN is expensive in bandwidth, so treat it as a fallback, not a primary route.
The key parameters you will set are the STUN server address clients should query, the TURN server address and port, the TURN credentials (a static username and secret, or better, time-limited credentials generated from a shared secret), and the allowed transport protocols, since TURN over TLS on port 443 survives networks that block UDP.
Best practices for public-cloud deployment:
  • Run TURN on a dedicated instance so relay bandwidth does not compete with signaling.
  • Use time-limited TURN credentials rather than static ones, so leaked credentials expire.
  • Expose TURN over both UDP and TLS-on-443 to survive restrictive networks.
  • Monitor TURN allocation counts; a spike means many clients are falling back to relay, and your bandwidth bill will follow.
If you would rather never think about TURN capacity planning, managed platforms such as VideoSDK operate TURN infrastructure and network-adaptive streaming as part of the service.

Signaling with XMPP Jingle

Jingle is the XMPP extension that structures peer-to-peer session negotiation, and it is the glue between ejabberd and WebRTC. Rather than inventing a custom message format, Jingle defines a small set of session actions that map cleanly onto the WebRTC offer/answer model.
ejabberd implements Jingle primarily as a router: it authenticates the sending user, verifies the recipient is reachable, and delivers the stanza. The intelligence lives in the clients, which is the correct division of labor for a signaling server.

The typical signaling exchange

A call proceeds through four core actions:
  1. Session initiate. The caller sends a Jingle session-initiate stanza carrying the SDP offer, describing supported codecs, resolutions, and transport candidates.
  2. Transport info. Both sides exchange ICE candidates as they discover them, sent as Jingle transport-info stanzas. These arrive continuously during early connection setup.
  3. Session accept. The callee answers with an SDP answer wrapped in a session-accept stanza, locking in the negotiated codecs and parameters.
  4. Session terminate. Either side ends the call with a session-terminate stanza, which also covers busy, decline, and error cases.
Presence and chat messages complement this flow: a ringing indicator can be a presence-based chat state, and call metadata can travel in the message body.

Using existing XMPP libraries

You do not need to hand-roll the client side. Strophe.js in the browser, and XMPP libraries for Android, iOS, and server-side environments, all support sending arbitrary stanzas, which is enough to carry Jingle payloads. The pattern is consistent: establish an authenticated XMPP session over TLS, attach to the ejabberd server, then serialize your WebRTC offer, answer, and ICE candidates into Jingle stanzas and exchange them with the peer. The SDP payload itself is opaque to the signaling layer; ejabberd just moves it.
One gotcha that bites teams here: candidates often arrive before the remote description is set. Queue incoming candidates on the client until the session-accept has been applied, or you will drop early connectivity paths and slow down call setup.

Managing Media Streams and Room Logic

One-to-one calls and multi-party rooms are different engineering problems, and your ejabberd design should acknowledge that early.
For one-to-one calls, direct peer-to-peer media is ideal: lowest latency, lowest cost, and the signaling load on ejabberd is trivial. ejabberd's multi-user chat rooms still add value even here, providing presence, ringing semantics, and a natural place to persist call history metadata.
For multi-party rooms, full mesh peer-to-peer collapses quickly. Each participant uploads their stream to every other participant, so a five-person call means four simultaneous uplinks per client, which overwhelms typical home connections. The standard answer is a Selective Forwarding Unit: each participant sends one stream up, and the SFU forwards the right streams back down. ejabberd can remain the coordination brain, storing room state, participant lists, and roles, while a dedicated media server handles the streams.
ejabberd's room model supports role-based access naturally. Room moderators can be mapped to call hosts, voice-enabled members to active speakers, and visitors to view-only participants. If you need viewer-to-speaker promotion, the kind of interactive live streaming where audience members join the stage, that role-switching pattern is well established in platforms like VideoSDK's Interactive Live Streaming, and you can approximate it in ejabberd by combining room role changes with SFU publish permissions.

Scaling and Performance Considerations

ejabberd's clustering model is its superpower for signaling scale. Nodes in an ejabberd cluster share routing tables and user state, so you can add capacity by adding machines, and a client connected to any node can reach any other user. Erlang's process-per-connection design means thousands of concurrent XMPP sessions per node are routine, and large public deployments have demonstrated the pattern at far greater scale.
Two scaling rules keep the architecture healthy:
  • Load-balance signaling, never media, through ejabberd. Put a TCP load balancer in front of the cluster for client connections, and keep media entirely off the signaling nodes.
  • Offload media to dedicated infrastructure. TURN relays and SFUs should scale independently of the signaling cluster, because their bottlenecks (bandwidth) are unrelated to the signaling bottleneck (connection count).
A quick production readiness checklist:
  • Cluster size validated against your target concurrent-session count.
  • TURN capacity provisioned for the 10 to 20 percent of clients that will need relay.
  • TLS everywhere, including TURN-over-TLS on port 443.
  • Monitoring on session counts, stanza throughput, and TURN allocations.
  • A documented fallback plan for the day you outgrow self-managed media, such as migrating to a managed real-time communication platform.

Security and Authentication

Security in an ejabberd WebRTC stack has three layers, and each needs deliberate attention.
First, protect the signaling channel. Enforce TLS on all client-to-server XMPP connections, and use SASL authentication so credentials never cross the wire in plaintext. ejabberd supports token-based and external authentication backends, which let your application issue short-lived credentials rather than storing passwords client-side.
Second, protect the relay layer. Restrict TURN access with time-limited credentials derived from a shared secret, so a leaked credential pair expires instead of granting indefinite relay bandwidth. Bind TURN allocations to reasonable quotas.
Third, protect the media. WebRTC mandates encryption for all audio and video via DTLS-SRTP, so media in transit is encrypted by default. For end-to-end encryption where the server cannot see media, you need insertable streams or an SFU design that never decrypts, which is a meaningful additional engineering commitment. If E2EE is a hard requirement, evaluate whether a platform with built-in E2EE support, like VideoSDK's encrypted calling, saves you the custom cryptography work.

Common Pitfalls and Troubleshooting

Most ejabberd WebRTC failures cluster into three buckets, and each has a reliable diagnostic path.
ICE failures show up as calls that ring but never connect. Check that STUN and TURN addresses are reachable from the client network, that TURN credentials are valid and not expired, and that the TURN server's public IP is correctly advertised. Browser console ICE state logs tell you whether candidates are being gathered at all; if the candidate list is empty, the problem is configuration, not network.
Mismatched codecs appear as a connected call with black video or silent audio. Compare the offered and answered SDP payloads in the browser's internal WebRTC state, and confirm both sides actually support the negotiated codec. Hardware-dependent codecs on mobile are a frequent culprit.
TURN timeouts usually mean the relay is unreachable on the configured port, or the allocation quota is exhausted. On the server side, ejabberd's log output will show authentication failures and delivery errors for signaling problems, while TURN-specific logs reveal allocation issues. The habit that pays off: reproduce failures with the browser console open on both endpoints simultaneously, since signaling problems appear on one side and media problems on the other.

Real-World Use Cases

The ejabberd plus WebRTC pattern shows up across industries. A telehealth platform can use ejabberd for patient-provider presence and chat, escalating to video when a consultation starts. A live-support product can embed a video call button in an existing XMPP chat, with the agent's availability driven by ejabberd presence. A collaborative education tool can use ejabberd rooms for class rosters and hand-raising, with WebRTC carrying tutor-to-student video. In each case the common thread is that chat and presence already exist on XMPP, and video is a natural extension rather than a parallel system.

Future Trends: WebRTC and ejabberd in 2026

The transport layer under real-time communication is evolving. Work on QUIC-based media transport and the broader WebRTC evolution often grouped under the "WebRTC 2.0" label aims to improve congestion control, multiplexing, and deployment flexibility. Because ejabberd's role is signaling rather than media transport, its architecture is insulated from these shifts: as long as session negotiation can be expressed as offer/answer, Jingle over XMPP remains a viable carrier. ejabberd's modular design, with its extension points and hot-code-upgrade heritage, positions it to absorb new signaling requirements without rearchitecting. Meanwhile, managed platforms are already shipping QUIC-era transport innovations, so teams should watch both paths.

Definitions Glossary

ejabberd: An open-source messaging server written in Erlang that implements XMPP, with support for MQTT and SIP, known for clustering and very high concurrent-connection capacity.
XMPP Jingle: An XMPP extension that structures peer-to-peer session negotiation, defining actions like session-initiate, transport-info, session-accept, and session-terminate for exchanging SDP and ICE data.
STUN: Session Traversal Utilities for NAT, a protocol that lets a client discover its public network address so it can build ICE candidates for direct connection.
TURN: Traversal Using Relays around NAT, a protocol that relays media through a server when direct peer-to-peer paths fail, typically needed by 10 to 20 percent of real-world clients.
SFU: Selective Forwarding Unit, a media server that receives one stream per participant and forwards the appropriate streams back out, the standard architecture for multi-party video.
ICE: Interactive Connectivity Establishment, the framework WebRTC uses to gather, exchange, and test candidate network routes between peers.

Key Takeaways

  • ejabberd WebRTC integration divides responsibilities cleanly: ejabberd carries signaling, presence, and authentication over XMPP Jingle, while WebRTC transports media peer-to-peer or through relays.
  • STUN/TURN configuration is mandatory for production, since a meaningful fraction of clients cannot establish direct paths through NATs and firewalls.
  • Multi-party video requires a media server (SFU or MCU); full mesh peer-to-peer degrades beyond a handful of participants, while ejabberd remains the coordination and room-state layer.
  • ejabberd's Erlang clustering model scales signaling to thousands of concurrent sessions, but media infrastructure should always be scaled separately from signaling.
  • If assembling signaling, TURN, SFU, and room logic yourself is more than your roadmap allows, a managed platform like VideoSDK provides rooms, recording, transcription, and adaptive streaming as a single SDK.

Conclusion

ejabberd WebRTC integration is a compelling path for teams that already live in the XMPP world or want full control over their signaling layer: ejabberd brings two decades of Erlang-proven scalability, and WebRTC brings encrypted peer-to-peer media. Start with a minimal setup, one server, two browser clients, a Jingle exchange, and a TURN fallback, then grow toward clustering and an SFU as your usage demands. When you want video calling without owning the infrastructure, explore the VideoSDK quickstart or grab free credits at app.videosdk.live. What are you building with ejabberd and WebRTC? Drop a comment, I'd love to hear what kind of real-time communication use case you're working on.

Step 5: Implement Participant View

In this step, we will implement the participant view, which displays the video and audio streams of all users in the call. This involves handling multiple WebRTC peer connections and dynamically updating the user interface as participants join or leave the call.

Building the Participant View

We have already defined the HTML and CSS for the participant view. Now, let's focus on the JavaScript functionality required to manage the participant video streams.

[a] Handle Remote Streams

First, we need to update the handleOffer function to properly handle remote streams and add video elements dynamically for each participant.
JavaScript for Handling Remote Streams (main.js):
JavaScript
1function handleOffer(connection, from, offer, localStream, peerConnections) {
2    const peerConnection = new RTCPeerConnection();
3    peerConnections[from] = peerConnection;
4
5    peerConnection.ontrack = (event) => {
6        const remoteVideo = document.createElement('video');
7        remoteVideo.srcObject = event.streams[0];
8        remoteVideo.autoplay = true;
9        remoteVideo.id = `remote-video-${from}`;
10        document.getElementById('remote-videos').appendChild(remoteVideo);
11    };
12
13    peerConnection.onicecandidate = (event) => {
14        if (event.candidate) {
15            connection.send($msg({ to: from, type: 'candidate' }).c('body').t(JSON.stringify(event.candidate)));
16        }
17    };
18
19    localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream));
20
21    peerConnection.setRemoteDescription(new RTCSessionDescription(offer)).then(() => {
22        return peerConnection.createAnswer();
23    }).then((answer) => {
24        return peerConnection.setLocalDescription(answer);
25    }).then(() => {
26        connection.send($msg({ to: from, type: 'answer' }).c('body').t(JSON.stringify(peerConnection.localDescription)));
27    });
28}

[b] Update Local Stream Handling

Ensure that the local video stream is correctly added to each new peer connection. We have already set up this functionality in the startWebRTC function, but let's make sure it covers all scenarios.
JavaScript for Local Stream Handling (main.js):
JavaScript
1async function startWebRTC(connection, room) {
2    const localVideo = document.getElementById('local-video');
3    const remoteVideos = document.getElementById('remote-videos');
4
5    // Get user media
6    const localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
7    localVideo.srcObject = localStream;
8
9    connection.addHandler((msg) => {
10        const from = msg.getAttribute('from');
11        const type = msg.getAttribute('type');
12        
13        if (type === 'offer') {
14            const offer = JSON.parse(msg.getElementsByTagName('body')[0].textContent);
15            handleOffer(connection, from, offer, localStream, peerConnections);
16        } else if (type === 'answer') {
17            const answer = JSON.parse(msg.getElementsByTagName('body')[0].textContent);
18            handleAnswer(from, answer, peerConnections);
19        } else if (type === 'candidate') {
20            const candidate = JSON.parse(msg.getElementsByTagName('body')[0].textContent);
21            handleCandidate(from, candidate, peerConnections);
22        }
23        
24        return true;
25    }, null, 'message', 'chat');
26
27    connection.send($pres().c('x', { xmlns: 'http://jabber.org/protocol/muc' }).c('history', { maxstanzas: 0 }));
28
29    // Send offer to new participants
30    localStream.getTracks().forEach(track => {
31        for (const peerId in peerConnections) {
32            peerConnections[peerId].addTrack(track, localStream);
33        }
34    });
35
36    // Create offer
37    for (const peerId in peerConnections) {
38        const peerConnection = peerConnections[peerId];
39        const offer = await peerConnection.createOffer();
40        await peerConnection.setLocalDescription(offer);
41
42        connection.send($msg({ to: peerId, type: 'offer' }).c('body').t(JSON.stringify(offer)));
43    }
44}

[c] Handling Participant Disconnection

We need to manage the UI and clean up resources when participants leave the call. Update the endCall function to handle this scenario properly.
JavaScript for Handling Disconnection (main.js):
JavaScript
1function endCall() {
2    const localVideo = document.getElementById('local-video');
3    const remoteVideos = document.getElementById('remote-videos');
4
5    // Stop local media tracks
6    const localStream = localVideo.srcObject;
7    localStream.getTracks().forEach(track => track.stop());
8
9    // Close peer connections
10    for (const peerId in peerConnections) {
11        peerConnections[peerId].close();
12        delete peerConnections[peerId];
13    }
14
15    // Clear video elements
16    localVideo.srcObject = null;
17    while (remoteVideos.firstChild) {
18        remoteVideos.removeChild(remoteVideos.firstChild);
19    }
20
21    // Redirect to join screen
22    window.location.reload();
23}

[d] Handling Peer Disconnections Gracefully

To handle peers leaving the call gracefully, you should listen for ICE connection state changes and remove video elements when peers disconnect.
JavaScript for Handling Peer Disconnections (main.js):
JavaScript
1function handlePeerDisconnection(peerId) {
2    const remoteVideo = document.getElementById(`remote-video-${peerId}`);
3    if (remoteVideo) {
4        remoteVideo.srcObject = null;
5        remoteVideo.parentNode.removeChild(remoteVideo);
6    }
7    if (peerConnections[peerId]) {
8        peerConnections[peerId].close();
9        delete peerConnections[peerId];
10    }
11}
12
13function handleOffer(connection, from, offer, localStream, peerConnections) {
14    const peerConnection = new RTCPeerConnection();
15    peerConnections[from] = peerConnection;
16
17    peerConnection.ontrack = (event) => {
18        const remoteVideo = document.createElement('video');
19        remoteVideo.srcObject = event.streams[0];
20        remoteVideo.autoplay = true;
21        remoteVideo.id = `remote-video-${from}`;
22        document.getElementById('remote-videos').appendChild(remoteVideo);
23    };
24
25    peerConnection.onicecandidate = (event) => {
26        if (event.candidate) {
27            connection.send($msg({ to: from, type: 'candidate' }).c('body').t(JSON.stringify(event.candidate)));
28        }
29    };
30
31    peerConnection.oniceconnectionstatechange = () => {
32        if (peerConnection.iceConnectionState === 'disconnected' || peerConnection.iceConnectionState === 'failed' || peerConnection.iceConnectionState === 'closed') {
33            handlePeerDisconnection(from);
34        }
35    };
36
37    localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream));
38
39    peerConnection.setRemoteDescription(new RTCSessionDescription(offer)).then(() => {
40        return peerConnection.createAnswer();
41    }).then((answer) => {
42        return peerConnection.setLocalDescription(answer);
43    }).then(() => {
44        connection.send($msg({ to: from, type: 'answer' }).c('body').t(JSON.stringify(peerConnection.localDescription)));
45    });
46}
By implementing the participant view functionality, we ensure that all video and audio streams are displayed correctly and dynamically updated as participants join or leave the call. The code handles remote streams, local stream integration, and disconnections gracefully.
In the next step, we will run our code and test the complete functionality of the Ejabberd WebRTC application.

Step 6: Run Your Code Now

In this final step, we will compile and run the Ejabberd WebRTC application, testing the complete functionality to ensure everything works as expected. We will also cover some basic troubleshooting steps in case you encounter any issues.

Compiling and Running the Application

[a] Start Ejabberd Server

Ensure that your Ejabberd server is running and properly configured to handle WebRTC signaling. You can start the Ejabberd server with the following command:
bash
1ejabberdctl start

[b] Compile the Erlang Code

Navigate to your project directory and compile the Erlang code using Rebar3:
bash
1cd /path/to/your/ejabberd-webrtc-app
2rebar3 compile

[c] Run the Application

After compiling, you can start the Erlang application:
bash
1erl -pa _build/default/lib/*/ebin -sname ejabberd_webrtc -s main
This command will start the Erlang shell with your application running. Ensure there are no errors during the startup process.

[d] Access the Web Application

Open a web browser and navigate to the HTML file of your application. If you are running a local server, you might access it via http://localhost/your-app-path.

Testing the Application

Join a Room

Enter a username and room name on the join screen and click the "Join" button. This should connect you to the Ejabberd server and initialize the WebRTC session.

Verify Local Video

Ensure that your local video stream appears on the screen. This confirms that your webcam is working and the media stream is being captured correctly.

Add Participants

Open the application in multiple browser tabs or devices and join the same room with different usernames. Verify that remote video streams are displayed correctly for all participants.

Test Controls

Use the mute, video, and end call buttons to test their functionality:
  • Mute/Unmute: Check that your microphone can be muted and unmuted.
  • Start/Stop Video: Verify that your video can be started and stopped.
  • End Call: Ensure that the call ends correctly and the UI resets.

Troubleshooting Common Issues

Connection Issues

If you cannot connect to the Ejabberd server, verify the following:
  • Ensure the Ejabberd server is running and reachable.
  • Check your ejabberd.yml configuration for WebSocket and BOSH settings.
  • Confirm that the Erlang/OTP environment is correctly set up.

Media Stream Issues

If you face issues with capturing or displaying media streams:
  • Ensure that your browser has permissions to access the webcam and microphone.
  • Check the console for errors related to getUserMedia or WebRTC APIs.

Signaling Issues

If WebRTC connections are not established:
  • Verify that the signaling messages are correctly exchanged between peers via Ejabberd.
  • Check for errors in the JavaScript console related to ICE candidates or SDP exchange.

Conclusion

In this guide, we walked through the steps to set up, configure, and run an Ejabberd WebRTC application. We covered the initial project setup, handling user inputs, managing media streams, and implementing essential call controls. By now, you should have a functional real-time communication application using Ejabberd and WebRTC.

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