How WebRTC Signaling Works

WebRTC itself has no signaling protocol — it only defines how to establish media. The signaling (how you say “call this number” or “I’m ringing”) is up to the application. Here’s how the Telnyx WebRTC SDK does it.

The Signaling Path

Key components:

Component	Role	Protocol
Your App	User interface	JavaScript
TelnyxRTC SDK	SDK logic	Internal
VSP	Voice SDK Proxy — translates WebSocket ↔ SIP	WebSocket + SIP
SIP Proxy	Routes SIP messages to carriers	SIP/UDP
B2BUA-RTC	Media gateway (WebRTC ↔ RTP)	DTLS/SRTP + RTP

VSP handles signaling only. B2BUA-RTC handles media only. They are separate systems.

WebSocket Connection

The SDK opens a single persistent WebSocket to rtc.telnyx.com:

const client = new TelnyxRTC({ login_token: jwt });
client.connect();

// WebSocket lifecycle:
// 1. DNS resolves rtc.telnyx.com → nearest VSP instance
// 2. TLS handshake (port 443)
// 3. WebSocket upgrade
// 4. Login message sent (JWT or credential)
// 5. Server responds with session info
// 6. telnyx.ready fires — you can make/receive calls

What the DNS resolution does

rtc.telnyx.com resolves to the nearest VSP based on DNS-based geo-routing:

Region	VSP DC
North America	NJ1
Europe	AMS3, FR5
Asia-Pacific	CN1

If the DNS routes to a suboptimal VSP (e.g., an Indian client hitting FR5 instead of CN1), call latency increases. See Configure Network & Firewall for troubleshooting.

Outbound Call Flow

When you call client.newCall(): What the SDK does at each step:

newCall() — Creates a Call object, starts ICE gathering
SIP INVITE — SDK sends invite message over WebSocket, VSP translates to SIP
SDP negotiation — Codec selection (OPUS, PCMU, PCMA), ICE candidates exchanged
Ringing — Remote party’s phone is ringing. call.state === 'ringing'
Answer (200 OK) — Remote party picked up. call.state === 'active'
Media flows — Audio transmitted via WebRTC (separate from signaling)

Inbound Call Flow

When someone calls your WebRTC client: What the SDK does:

Incoming INVITE — VSP receives SIP INVITE, pushes to SDK over WebSocket
callUpdate notification — notification.call.state === 'ringing'
Your app decides — Call call.answer() or call.hangup()
Answer — SDK sends 200 OK, establishes WebRTC media
Media flows — Two-way audio established

Session Description Protocol (SDP)

During call setup, both sides exchange SDP (Session Description Protocol) to agree on:

SDP Negotiates	Example
Audio codecs	OPUS (preferred), PCMU (G.711u), PCMA (G.711a)
Codec parameters	OPUS with FEC, stereo/mono, sample rate
ICE candidates	How to reach each other for media
DTLS fingerprint	For encrypting media (SRTP)
Media direction	Sendrecv (two-way), sendonly, recvonly
Bandwidth	Maximum bitrate

The SDK handles SDP negotiation automatically. You don’t need to construct SDP manually.

Codec Priority

The SDK’s default codec priority:

OPUS — Best quality, handles packet loss well, variable bitrate
PCMU — G.711μ-law, universal compatibility, 64kbps
PCMA — G.711A-law, European PSTN standard, 64kbps

OPUS is strongly preferred — it handles jitter and packet loss better than G.711, and uses less bandwidth.

WebSocket Reconnection

If the WebSocket drops, the SDK automatically reconnects: See Handle Reconnection for the full reconnection behavior and how to handle it in your app.

Custom Headers

You can pass custom SIP headers in both directions:

Outbound (your app → carrier)

const call = client.newCall({
  destinationNumber: '+12345678900',
  customHeaders: [
    { name: 'X-My-Header', value: 'value1' },
    { name: 'X-Another', value: 'value2' },
  ],
});

These appear as SIP headers in the INVITE.

Inbound (carrier → your app)

Inbound custom headers are available in the notification:

client.on('telnyx.notification', (notification) => {
  if (notification.type === 'callUpdate' && notification.call.state === 'ringing') {
    const customHeaders = notification.call.customHeaders;
    // e.g., { 'X-Caller-Name': 'John' }
  }
});

What Signals What

Action	SDK Method	SIP Message	Who Initiates
Make a call	`newCall()`	INVITE	Client
Answer a call	`call.answer()`	200 OK	Client
Reject a call	`call.hangup()`	487 Request Terminated	Client
End a call	`call.hangup()`	BYE	Either side
Put on hold	`call.hold()`	re-INVITE with sendonly	Client
Resume from hold	`call.unhold()`	re-INVITE with sendrecv	Client
Send DTMF	`call.sendDigits()`	INFO (RFC 2833)	Client
Mute audio	`call.mute()`	(local only, no SIP)	Client

Mute is a local operation — it stops sending audio from your microphone but doesn’t send any SIP signal. The remote party doesn’t know you’re muted (unless you tell them via your app).

Getting Started

Authentication

Push Notifications

Tutorials

Use Cases

Troubleshooting

SDKs

API Reference

For AI Agents

How WebRTC Signaling Works

How WebRTC Signaling Works

The Signaling Path

WebSocket Connection

What the DNS resolution does

Outbound Call Flow

Inbound Call Flow

Session Description Protocol (SDP)

Codec Priority

WebSocket Reconnection

Custom Headers

Outbound (your app → carrier)

Inbound (carrier → your app)

What Signals What

See Also

​How WebRTC Signaling Works

​The Signaling Path

​WebSocket Connection

​What the DNS resolution does

​Outbound Call Flow

​Inbound Call Flow

​Session Description Protocol (SDP)

​Codec Priority

​WebSocket Reconnection

​Custom Headers

​Outbound (your app → carrier)

​Inbound (carrier → your app)

​What Signals What

​See Also

How WebRTC Signaling Works

The Signaling Path

WebSocket Connection

What the DNS resolution does

Outbound Call Flow

Inbound Call Flow

Session Description Protocol (SDP)

Codec Priority

WebSocket Reconnection

Custom Headers

Outbound (your app → carrier)

Inbound (carrier → your app)

What Signals What

See Also