The Future of Voice

WebRTC vs.
Traditional VoIP

Why modern businesses are moving from legacy SIP softphones to WebRTC. Lower latency, mandatory encryption, and HD audio, directly in the browser.

WebRTCDirect Peer-to-Peer
User A
< 50ms
User B
Legacy VoIPServer Relayed
PBX

Security

WebRTC
Mandatory Encryption (DTLS/SRTP)
Legacy VoIP
Optional (Often Unencrypted)
Winner: WebRTC

Network Adaptability

WebRTC
Dynamic Bitrate (Opus Codec)
Legacy VoIP
Static Codecs (G.711)
Winner: WebRTC

Latency

WebRTC
Ultra-low (P2P optimized)
Legacy VoIP
Higher (Server hops)
Winner: WebRTC
SECURITY FIRST

Why WebRTC is More Secure

Traditional VoIP (SIP) was designed in the 90s without encryption as a mandatory requirement. This makes it susceptible to packet sniffing and eavesdropping.

WebRTC was built for the modern web. It requires DTLS (Datagram Transport Layer Security) for key exchange and SRTP (Secure Real-time Transport Protocol) for media. Without encryption, a WebRTC connection simply cannot be established.

  • No VPN required for security
  • End-to-End Encryption capability
  • Sandboxed within the browser

The Opus Codec Advantage

G.711 (Legacy)

Used by traditional phone lines. Static bitrate. If your internet drops slightly, audio cuts out or becomes robotic.

Opus (WebRTC)

Dynamic bitrate. Adapts to network conditions in real-time. Maintains clarity even on 3G/4G connections.

Experience WebRTC Quality

Make a call right now from your browser and hear the difference.

Under the Hood: How WebRTC and SIP Calls Actually Connect

In short

The comparison table above covers the outcomes. This section covers the mechanics: how each technology signals a call, negotiates media, punches through NAT, and encodes audio. Understanding ICE, STUN, TURN, and the Opus codec explains why a browser call often sounds better than a desk phone on the same network.

Signaling: SIP messages versus application-defined channels

Traditional VoIP standardizes signaling with SIP, a text protocol resembling HTTP that carries INVITE, RINGING, and BYE messages between endpoints and a PBX or SIP server, usually over UDP or TCP port 5060. Interoperability is the strength, but exposed SIP ports attract constant scanning and toll fraud attempts, which is why providers wrap them in session border controllers.

WebRTC deliberately leaves signaling undefined. The application exchanges session descriptions over whatever secure channel it already has, typically a WebSocket over HTTPS. There is no standard port to scan and no SIP stack to patch, since the browser only handles the media layer.

The session descriptions themselves use SDP, the same format SIP carries, which is what makes bridging between the two worlds tractable. A gateway can translate a browser's offer into a SIP INVITE and back, so the technologies interoperate even though their signaling philosophies differ.

NAT traversal: why ICE, STUN, and TURN matter

Most devices sit behind network address translation, which breaks the older assumption that an endpoint can be reached at the address it advertises. Classic SIP deployments handled this with ALGs and manual router configuration, a notorious source of one-way audio problems.

WebRTC builds traversal in through ICE. The browser gathers candidate addresses, uses STUN to discover its public address, and tests every candidate pair until one connects. When a strict firewall blocks direct paths, media relays through a TURN server, often over TCP port 443 so it passes as ordinary HTTPS traffic. The result is calls that connect from hotel, office, and mobile networks without any router changes.

Codecs: Opus against G.711 in practice

G.711, the legacy VoIP default, samples audio at 8 kHz and uses a fixed 64 kbps stream, reproducing roughly the frequency range of a 1990s landline. It cannot adapt when the network degrades, so congestion turns directly into dropouts.

Opus, mandatory in WebRTC, supports sampling up to 48 kHz and scales its bitrate dynamically from about 6 to 510 kbps. Combined with built-in packet loss concealment and forward error correction, it delivers wideband audio on a good connection and degrades gracefully rather than abruptly on a poor one. This codec difference is most of why browser calls sound noticeably fuller than traditional phone calls.

Alongside the codec, browsers layer in acoustic echo cancellation, noise suppression, and automatic gain control as part of the WebRTC audio pipeline, processing that a basic hardware SIP phone typically lacks or implements less aggressively.

Reaching ordinary phones from a browser

A pure WebRTC session only connects two internet endpoints. To ring a normal phone, the call must pass through a media gateway that bridges the encrypted WebRTC stream into the public telephone network, transcoding audio and translating signaling along the way.

This is the architecture StartACall runs on. The leg between your browser and the gateway uses WebRTC with DTLS and SRTP encryption, then the gateway hands the call to carrier routes that terminate at any landline or mobile worldwide. You get modern codec quality and encryption on your side of the call without the recipient needing anything beyond a working phone.

Frequently asked questions

Does WebRTC use SIP?+

Not by itself. WebRTC defines media transport and encryption but leaves signaling to the application. Some platforms run SIP over WebSockets to bridge into existing phone systems, while others, including browser calling services, use their own signaling protocols entirely.

What is a TURN server and when is it needed?+

TURN is a relay used when firewalls block a direct media path between endpoints. The call's encrypted audio routes through the relay, often on TCP port 443, which lets WebRTC calls succeed on strict corporate and hotel networks where classic VoIP fails.

Why does Opus sound better than G.711?+

Opus captures a much wider frequency range, up to 48 kHz sampling against 8 kHz for G.711, and adjusts its bitrate to network conditions in real time. It also conceals lost packets, so brief network hiccups cause less audible damage.

Can a WebRTC call reach a regular landline or mobile number?+

Yes, through a media gateway that bridges the browser session into the public telephone network. Services like StartACall operate this bridge, so you dial an ordinary number from a browser tab and the recipient answers a normal call.

Is WebRTC harder to run through a corporate firewall than SIP?+

Usually easier. SIP needs specific ports opened and often misbehaves behind NAT. WebRTC negotiates a path automatically with ICE and can fall back to relaying over port 443, which most corporate firewalls already allow for HTTPS.

Last reviewed June 2026Reviewed by the StartACall calling teamDialing rules cross checked against ITU international dialing procedures
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