sip_ua 1.1.0

A SIP UA stack for Flutter/Dart, based on flutter-webrtc, support iOS/Android/Destkop/Web.

dart-sip-ua

A dart-lang version of the SIP UA stack, ported from JsSIP.

Overview

• Use pure dart-lang
• SIP over WebSocket && TCP (use real SIP in your flutter mobile, desktop, web apps)
• Audio/video calls (flutter-webrtc) and instant messaging
• Support with standard SIP servers such as OpenSIPS, Kamailio, Asterisk, 3CX and FreeSWITCH.
• Support RFC2833 or INFO to send DTMF.

Currently supported platforms

• ✅ iOS
• ✅ Android
• ✅ Web
• ✅ macOS
• ✅ Windows
• ✅ Linux
• ❌ Fuchsia

Install

Android

• Proguard rules:

-keep class io.flutter.app.** { *; }
-keep class io.flutter.plugin.**  { *; }
-keep class io.flutter.util.**  { *; }
-keep class io.flutter.view.**  { *; }
-keep class io.flutter.**  { *; }
-keep class io.flutter.plugins.**  { *; }

-keep class com.cloudwebrtc.webrtc.** {*;}
-keep class org.webrtc.** {*;}

Quickstart

Run example:

• dart-sip-ua-example
• or add your example.

Register with SIP server:

• Asterisk
• FreeSWITCH
• OpenSIPS
• 3CX
• Kamailio
• or add your server example.

FAQ's OR ISSUES

expand

Server not configured for DTLS/SRTP

WEBRTC_SET_REMOTE_DESCRIPTION_ERROR: Failed to set remote offer sdp: Called with SDP without DTLS fingerprint.

Your server is not sending a DTLS fingerprint inside the SDP when inviting the sip_ua client to start a call.

WebRTC uses encryption by Default, all WebRTC communications (audio, video, and data) are encrypted using DTLS and SRTP, ensuring secure communication. Your PBX must be configured to use DTLS/SRTP when calling sip_ua.

Why isn't there a UDP connection option?

This package uses a WS or TCP connection for the signalling processs to initiate or terminate a session (sip messages). Once the session is connected WebRTC transmits the actual media (audio/video) over UDP.

If anyone actually still wants to use UDP for the signalling process, feel free to submit a PR with the large amount of work needed to set it up, packet order checking, error checking, reliability timeouts, flow control, security etc etc.

SIP/2.0 488 Not acceptable here

The codecs on your PBX server don't match the codecs used by WebRTC

• opus (payload type 111, 48kHz, 2 channels)
• red (payload type 63, 48kHz, 2 channels)
• G722 (payload type 9, 8kHz, 1 channel)
• ILBC (payload type 102, 8kHz, 1 channel)
• PCMU (payload type 0, 8kHz, 1 channel)
• PCMA (payload type 8, 8kHz, 1 channel)
• CN (payload type 13, 8kHz, 1 channel)
• telephone-event (payload type 110, 48kHz, 1 channel for wideband, 8000Hz, 1 channel for narrowband)

NOTE

Thanks to the original authors of JsSIP for providing the JS version, which makes it possible to port the dart-lang.

• José Luis Millán
• Iñaki Baz Castillo
• Saúl Ibarra Corretgé

Sponsors

The first version was sponsored by Suretec Systems Ltd. T/A SureVoIP.

Contributing

The project is inseparable from the contributors of the community.

• SureVoIP - Sponsor
• CloudWebRTC - Original Author
• Robert Sutton - Contributor
• Gavin Henry - Contributor
• Perondas - Contributor
• Mikael Wills - Contributor

License

dart-sip-ua is released under the MIT license.