📞 SIP phone (Experimental)

Use the built-in SIP phone panel in MiroTalk SFU to register a SIP account from inside a room, then call an extension, a phone number or a SIP address.

This feature is currently marked as experimental and is intended as a practical MVP for SIP interop testing.

MiroTalk SFU SIP phone panel

🧪 Experimental SIP demo (Kamailio)

A minimal, self-contained SIP demo is available in sip/:

Kamailio in Docker (REGISTER + INVITE routing)
Simple Node.js web SIP client example

Quick start:

Bash

cd sip
chmod +x run.sh
./run.sh

Open http://localhost:8088. Full instructions: sip/README.md.

You can also test the in-room SIP phone MVP directly in MiroTalk SFU:

Start MiroTalk SFU normally.
Start the SIP stack (./sip/run.sh).
Join a room with two browsers/users.
Open Settings → SIP phone, register random users, and call each other.

The in-room SIP phone uses this demo server by default (ws://<host>:5066, SIP domain <host>).

WebSocket contact aliasing

The demo Kamailio config aliases WebSocket contacts (add_contact_alias / handle_ruri_alias) so in-dialog ACK/BYE/re-INVITE requests reach browser clients, which advertise unroutable Contacts (e.g. *.invalid / 192.0.2.x). Apply the same approach on your own Kamailio/OpenSIPS setup if calls connect but hang-up or media renegotiation fails.

Compatibility

The SIP phone works with the bundled demo and is designed to be compatible with most SIP servers that expose WSS (WebSocket Secure) transport (for example Kamailio, Asterisk, FreeSWITCH, OpenSIPS, and similar SIP platforms configured for WebRTC clients).

For non-demo deployments, ensure your SIP server provides:

wss:// SIP WebSocket endpoint reachable from browsers
Correct SIP domain/realm for registration
WebRTC-compatible SIP profile/transports
DTLS-SRTP / ICE / NAT handling suitable for browser endpoints
A PSTN trunk (SIP provider or PBX gateway) if you want to call real phone numbers

Experimental scope

SIP interoperability can vary by server configuration, NAT topology, auth policy, and codec/transcoding rules. Validate your full call flow in your own environment before production usage.

SIP phone panel

The panel is organized in a few sections:

Status bar: shows the current state (idle, registering, registered, incoming, in call, error) with a color indicator and a live call timer during active calls.
Incoming call banner: shows the caller with Accept / Reject buttons (a toast notification is also shown).
Account: SIP registration settings. The card collapses automatically after a successful registration and shows the registered identity (sip:user@domain); fields are locked while registered.
Dialer: call target input, keypad, call controls and recent numbers.

Account fields

Field	Description	Example
WS server	SIP WebSocket endpoint used by the browser SIP stack.	`ws://127.0.0.1:5066` or `wss://sip.example.com:7443`
SIP domain	SIP realm/domain sent in SIP registration and INVITE flows.	`127.0.0.1`
SIP username	SIP account/extension to register. Use the inline buttons to generate a random user or copy the current one.	`user8165`
Auth username	Authentication username (optional, defaults to SIP username).	`user8165`
SIP password	Optional in open/local test setups, required when server auth is enabled.	`••••••••`
Contact IP compatibility hack	Toggles SIP.js `hackIpInContact` for better interoperability across SIP servers.	on/off

Advanced compatibility (collapsible):

Field	Description	Example
Display name	Optional caller display name.	`Alice`
Outbound proxy / route set	Optional outbound proxy or route set.	`sip:proxy.example.com;lr`
Phone number gateway	Domain used for phone-number calls (defaults to SIP domain). Extensions keep using the SIP domain.	`pstn.provider.com`
Register expires (s)	Registration expiry in seconds.	`600`

Use Register / Unregister to control the SIP registration lifecycle.

Dialer

The call target input accepts an extension, a phone number or a SIP address. The detected type and the resulting SIP URI are previewed under the input:

You type	Detected as	Called URI
`1001`, `alice`	Extension	`sip:1001@<SIP domain>`
`+1 (555) 123-4567`, `0039 06 1234 5678`	Phone number	`sip:+15551234567@<phone number gateway or SIP domain>`
`tel:+15551234567`	Phone number	same as above
`bob@example.org`, `sip:bob@example.org`	SIP address	used as-is

Phone-number formatting (spaces, dashes, dots, parentheses) is stripped.
Values starting with + or with 7+ digits are treated as phone numbers; shorter ones as extensions.
Invalid targets (e.g. 12+34) are rejected.

Dialer controls:

Keypad: tap digits, * and #; hold 0 to insert +.
Backspace: delete the last character; hold to clear the input.
Enter in the target input places the call.
Call / Mute / Hang up: start a call, mute/unmute your microphone, and end the call.
DTMF: during an active call, the keypad (or keyboard keys 0-9, *, #) sends DTMF tones (RTP, with SIP INFO fallback).
Recent: the last 6 dialed targets are shown as quick-dial chips.

Calling real phone numbers

Calling a real phone number requires a SIP provider or PBX with a PSTN trunk. Set Advanced compatibility → Phone number gateway to the provider's domain to route phone numbers there, while extensions keep using the SIP domain.

With the Kamailio demo (no PSTN trunk), a phone number only rings if a user registered with that number, e.g. register as 5551234567 and dial (555) 123-4567.

Saved settings

Server settings (WS server, SIP domain, display name, outbound proxy, phone number gateway, register expires, contact hack) are remembered in the browser (localStorage).
The SIP username and auth username are kept per tab (sessionStorage), so two tabs never register the same user.
The SIP password is never stored.

Recommended test flow

Launch SFU and SIP demo stack.
Open room in Browser A and Browser B.
In both browsers, open Settings → SIP phone.
Click Random user (if needed), then Register on both sides.
Copy Browser B user and paste into Browser A Call target (or type it on the keypad).
Press Call in Browser A.
In Browser B, click Accept on the incoming call banner.
Verify two-way audio, Mute, DTMF and the call timer, then end the call with Hang up.

To test phone-number dialing with the demo, register Browser B as 5551234567 and dial (555) 123-4567 from Browser A.

Troubleshooting

Registration fails

Confirm SIP server is running and listening on the configured WS/WSS endpoint.
Verify SIP domain, username, auth username, and password.
Check browser console and SIP server logs for auth (401/403) or transport errors.

Call does not connect

Confirm both users are successfully registered first.
Verify target extension exists and is reachable.
Check the detected target type and URI preview under the call target input.
For phone numbers, verify the Phone number gateway and your provider's PSTN trunk / dial plan.
Check SIP routing rules (INVITE path) in your SIP server.

Call connects but hang-up / ACK fails

Browser clients advertise unroutable Contacts; make sure your proxy aliases WebSocket contacts (see the Kamailio note above) so in-dialog requests reach the browser.

No media / one-way audio

Review NAT traversal and ICE candidate reachability.
Validate DTLS-SRTP and RTP profile compatibility.
Confirm UDP/TCP media ports and firewall policies allow media traffic.

DTMF not received

Ensure your SIP server/IVR accepts RFC 4733 RTP events or SIP INFO (application/dtmf-relay).

WSS errors in browser

Use valid TLS certificates on SIP WSS endpoint.
Ensure hostname in certificate matches the WS/WSS URL used by the client.
Avoid mixed-content issues (https app pages calling insecure ws:// endpoints).

Notes

SIP phone is currently intended for experimental and integration testing.
For internet-facing usage, prefer wss:// with trusted certificates.
Keep SIP logs enabled while validating third-party interop profiles.