WebRTC Softphone Security Explained: Encryption, Browser Risks, Best Practices

In the rapidly evolving landscape of VoIP, the WebRTC softphone has emerged as the gold standard for business communication. By eliminating the need for plugins and native desktop apps, WebRTC (Web Real-Time Communication) has democratized voice and video calling, embedding it directly into browsers and CRMs. However, this accessibility brings a critical question to the forefront of every CTO's mind: Is it secure?

While WebRTC is touted as "secure by design," the reality is more nuanced. The protocol enforces mandatory encryption, but the surrounding infrastructure like signaling servers, browser implementations, and NAT traversal mechanisms—can introduce significant vulnerabilities if not properly fortified.

For enterprises deploying custom softphones (like those built by Sheerbit), understanding these risks is not just a compliance checkbox; it is a fundamental requirement to protect proprietary data and maintain user trust. This guide provides a deep technical dive into WebRTC security, exposing the hidden risks in browsers and offering a battle-tested checklist for securing your softphone infrastructure.

1. The Foundation: How WebRTC Encryption Actually Works

Unlike legacy VoIP protocols that often transmitted media in cleartext (RTP), WebRTC forces security at the standard level. It is impossible to establish a WebRTC connection without encryption. Here is the technical breakdown of the protocols that form this shield.

DTLS (Datagram Transport Layer Security)

WebRTC signaling happens over TCP (usually), but the media itself travels over UDP for speed. Standard TLS (used in HTTPS) doesn't work well with UDP because it expects reliable packet delivery.

• The Role of DTLS: DTLS is essentially "TLS for UDP." It handles the initial handshake between two softphone clients (peers). During this handshake, they exchange cryptographic keys securely.
• Self-Signed Certificates: In a peer-to-peer (P2P) context, there is no central certificate authority to validate every peer. Instead, WebRTC endpoints generate self-signed certificates and exchange their "fingerprints" via the signaling server to ensure the device you are talking to is the same one that initiated the call.

SRTP (Secure Real-Time Transport Protocol)

Once DTLS establishes the keys, the heavy lifting is handed over to SRTP.

• Media Encryption: SRTP encrypts the actual payload—your voice and video data. Even if a hacker captures the UDP packets flowing through a public Wi-Fi network, they will hear nothing but static.
• Replay Protection: SRTP includes mechanisms to prevent "replay attacks," where an attacker captures a legitimate conversation stream and re-transmits it later to impersonate a user or disrupt a session.

Key Takeaway: The "DTLS-SRTP" handshake is the bedrock of WebRTC security. If your softphone implementation fails to handle the key exchange correctly, or if the signaling server leaks these fingerprints, the encryption is rendered useless.

2. Browser Risks: The Vulnerabilities You Can’t Ignore

Because WebRTC softphones live inside the browser (Chrome, Firefox, Safari), they inherit the security model of that browser. While generally robust, this introduces specific attack vectors that native apps don't always face.

The ICE Candidate IP Leak

To connect two peers directly, WebRTC uses the ICE (Interactive Connectivity Establishment) framework to find the best path. This involves gathering "candidates"—possible IP addresses where the device can be reached.

• The Risk: A browser queries all network interfaces to find these candidates. This can inadvertently reveal a user’s private local IP address (behind the NAT) or their true public IP even if they are using a VPN.
• The Impact: For businesses requiring anonymity or operating in high-security zones, leaking a private IP topology is a major breach.
• Mitigation: Modern softphones must configure the iceTransportPolicy to relay (forcing traffic through a TURN server) or use browser extensions/policies that disable non-proxied UDP.

Permission Persistence & "Clickjacking"

Browsers require user permission to access the microphone and camera.

• Persistent Permissions: If a user clicks "Always Allow" for a domain, a compromised script injected into that domain (via XSS) could theoretically activate the microphone without a new prompt.
• The Fix: Custom softphone interfaces should always provide visual indicators (like a distinct UI "On Air" light) separate from the browser’s native indicator, and sessions should be timed out aggressively.

Cross-Site Scripting (XSS) in Signaling

WebRTC does not define how signaling (call setup) happens. Most developers use WebSockets.

• The Attack: If your signaling server or the web page hosting the softphone is vulnerable to XSS, an attacker can inject malicious JavaScript. This script can’t decrypt the media (thanks to DTLS-SRTP), but it can reroute the call to a different destination or silently record the call metadata.

3. Securing the Signaling Plane: SIP over WebSockets

For many business softphones, WebRTC is just the access point; the backend is often a standard SIP server (like Asterisk, FreeSWITCH, or Kamailio). Securing the bridge between these two worlds is critical.

Mandatory HTTPS and WSS

You cannot run WebRTC on an insecure origin (HTTP), but you can mistakenly leave your WebSocket connection insecure (WS instead of WSS).

• WSS (WebSocket Secure): Just like HTTPS, WSS encrypts the signaling traffic using TLS. This protects the SIP credentials (username/password) and the SDP (Session Description Protocol) packets from being intercepted during the handshake.
• SIP Authentication: Ensure your SIP server enforces Digest Authentication even for WebSocket connections. Never rely solely on the fact that the request came from your web server.

Validating SIP Traffic

Since the softphone client is JavaScript running on a user's machine, the client should never be fully trusted.

• Input Sanitization: A malicious user could modify the JavaScript to send malformed SIP headers or giant SDP packets intended to crash the SIP server (DoS attack). Your SIP proxy must validate and rate-limit all incoming WebSocket frames before processing them.

4. Infrastructure Security: TURN Servers and SFUs

In a professional VoIP environment, Peer-to-Peer (P2P) isn't always possible or desirable. You will likely use TURN servers (for relaying media) or SFUs (Selective Forwarding Units) for conference calls.

The "Encryption Gap" in SFUs

This is one of the most misunderstood aspects of WebRTC security.

• P2P is E2EE: In a direct 1:1 call, encryption is truly end-to-end. Only the two users have the keys.
• SFU Decryption: In a conference call, the media goes to a server (SFU) which distributes it to other participants. Standard SFUs decrypt the media to inspect packets and route them, then re-encrypt it for the receiving participants.
• The Risk: Technically, the SFU server has access to the unencrypted media for a brief microsecond in memory. If the server is compromised, the calls are compromised.
• The Solution: Use "Insertable Streams" (E2EE for conferences) if you are building a highly sensitive application (like healthcare or finance), or ensure your SFU infrastructure is hardened, isolated, and strictly monitored.

Securing TURN Servers

TURN servers relay traffic when P2P fails. They are publicly accessible by necessity, making them targets for abuse.

• Authentication is Non-Negotiable: Never run an "open relay" TURN server. Use REST API time-limited credentials. The softphone requests a token from your backend, and that token grants access to the TURN server for only the duration of that specific call.

5. WebRTC Softphone Security Checklist (2025)

For CTOs and developers, here is the implementation roadmap to ensure your softphone is enterprise-ready.

Category	Action Item	Priority
Signaling	Enforce WSS:// (TLS 1.2 or 1.3) for all WebSocket connections.	Critical
Media	Implement Perfect Forward Secrecy (PFS) in DTLS configuration.	High
Privacy	Set iceTransportPolicy: 'relay' for high-security users to hide IP.	High
Access Control	Use Time-Limited Tokens (ephemeral credentials) for TURN access.	Critical
SIP	Validate Origin Headers on the SIP server to reject unauthorized domains.	Medium
Client-Side	Implement Content Security Policy (CSP) to prevent XSS script injection.	Critical
Infrastructure	Regularly rotate shared secrets used for TURN authentication.	Medium

Conclusion

WebRTC has revolutionized the softphone market by lowering barriers to entry, but it has raised the bar for security responsibility. "Secure by design" does not mean "secure by default." The encryption protocols (DTLS/SRTP) are robust, but they are only as strong as the signaling path that negotiates them and the browser environment that hosts them.

For businesses relying on VoIP for critical operations, using a generic or off-the-shelf solution often leaves these configuration gaps wide open. The future belongs to custom-built, security-first softphones that treat metadata protection with the same rigor as media encryption.

Ensure your development strategy—whether in-house or through a specialized partner—accounts for these hidden risks. In the world of real-time communication, trust is your most valuable currency.