A dart-lang version of the SIP UA stack.

Overview

dart-sip-ua

Financial Contributors on Open Collective pub package slack

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

Overview

  • Use pure dart-lang
  • SIP over WebSocket (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 and FreeSWITCH.
  • Support RFC2833 or INFO to send DTMF.

Currently supported platforms

  • iOS
  • Android
  • Web
  • macOS
  • Linux
  • Windows
  • 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:

Register with SIP server:

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

NOTE

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

Sponsors

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

Contributing

The project is inseparable from the contributors of the community.

License

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

Comments
  • RFC8599 support

    RFC8599 support

    OpenSIPS has just added this. Amazing. Do we need anything on the UA I wonder.

    https://www.linkedin.com/posts/opensips_opensips31-release-opensource-activity-6670654698992795648-qaJn

    https://tools.ietf.org/html/rfc8599

    enhancement wontfix 
    opened by ghenry 31
  • Audio call terminates after 1 second

    Audio call terminates after 1 second

    Describe the bug

    Audio call terminates after 1 second. Also, library didn't handle any inbound calls (I even don't see any SIP log).

    To Reproduce Steps to reproduce the behavior:

    1. Create Telnyx account
    2. Make SIP calls to any ordinary phone number. For example audio call to sip:[email protected]
    3. Call starts successfully
    4. I hear the voice from both sides
    5. Call terminates with one of the several errors: call failed with cause: Code: [488], Cause: Bad Media Description, Reason: Not Acceptable Here or call failed with cause: Code: [503], Cause: SIP Failure Code, Reason: Service Unavailable

    Log files: android_call_interupted.txt ios_call_interupted.txt

    Expected behavior Calls shouldn't terminate after 1 second

    System Infomation() Flutter SDK Version: 1.13.7-pre.26 Target OS and Version: Android 9 / iOS 13 Host OS and Version: Ubuntu 19.04 / Mac Os Mojave

    I am working on Flutter client for Telnyx. It helps connect VoIP and ordinary phones. SIP related docs - https://sip.telnyx.com/. It looks like library incompatibility (implementation or config) with Telnyx SIP/WebRTC. I am ready to fix everything myself but I don't have experience with SIP/WebRTC and I need direction where to start debugging this issue.

    PM me If you need Telnyx credentials to test.

    Thanks for good SIP library!

    bug wontfix 
    opened by xal 25
  • Different SDP on iOS vs Android

    Different SDP on iOS vs Android

    Describe the bug I'm getting Incompatible SDP when doing calls from iOS. The only difference I can see is some UUIDs are uppercase. Why are they different?

    Not working top part of SDP from iOS:

      | v=0 o=- 6079434066994874248 2 IN IP4 x.x.x.x s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS DD2D5D7A-C8DF-4955-B92D-FFCBD9B01DE0 m=audio 55698 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126 c=IN IP4 x.x.x.x a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:102 ILBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105
    -- | --
    

    Working top part of SDP from Android 11:

      | v=0 o=- 8464561152007567274 2 IN IP4 x.x.x.x s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS 4ecf41c6-9195-40ed-a4b5-bf44bda73aca m=audio 48391 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 110 112 113 126 c=IN IP4 x.x.x.x a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:102 ILBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105
    -- | --
    
    

    To Reproduce Steps to reproduce the behavior:

    1. Call FreeSWITCH echo feature code

    Expected behavior Works

    System Infomation() Flutter SDK Version: 2.0.3 Target OS and Version: iOS 14.4.1 iPhone 8 Host OS and Version: macOS 10.15: Catalina (Jazz)

    bug 
    opened by ghenry 21
  • Replace events for pr

    Replace events for pr

    This is a very large PR...

    I've replaced the logger with a new logger which outputs the time, log level, source file and line number of the log statement that created the logger line.

    [2019-10-11 12:18:26.910] Level.debug WebSocketInterface.dart:161 ::: send()

    I've replaced events2 (the message bus) with a completely new and fully typed (as much as is currently possible) event bus called EventManager, this has been a very invasive change and as such I have spent a lot of time making sure things still work and have undergone a full internal code review of every line of changed code.

    It is now possible to statically determine which parameters are available when receiving an event and which parameters must be passed when emitting an event.

    handlers.on(EventOnSuccessResponse(), (EventOnSuccessResponse event) { logger.warn("$event.response"); });

    opened by rlsutton1 18
  •  CERTIFICATE_VERIFY_FAILED: unable to get local issuer

    CERTIFICATE_VERIFY_FAILED: unable to get local issuer

    Hi, thanks for this implementation webrtc of flutter

    When I try to register to my webrtc error I get following error.

    Log output:

    I/flutter (21450): [2019-11-18 16:03:16.324] Level.warning sip_ua_helper.dart:127 ::: UA instance already exist!, stopping UA and creating a new one... I/flutter (21450): [2019-11-18 16:03:16.330] Level.debug ua.dart:282 ::: stop() I/flutter (21450): [2019-11-18 16:03:16.337] Level.debug transport.dart:138 ::: close() I/flutter (21450): [2019-11-18 16:03:16.343] Level.debug websocket_interface.dart:147 ::: disconnect() I/flutter (21450): [2019-11-18 16:03:16.348] Level.debug websocket_interface.dart:204 ::: WebSocket wss://kam13.x2one.com:8443 closed I/flutter (21450): [2019-11-18 16:03:16.354] Level.debug sip_ua_helper.dart:158 ::: disconnected => Code: [0], Cause: disconnect, Reason: close by local I/flutter (21450): [2019-11-18 16:03:16.362] Level.debug websocket_interface.dart:32 ::: new() [url:wss://kam13.x2one.com:8443] I/flutter (21450): [2019-11-18 16:03:16.372] Level.debug websocket_interface.dart:44 ::: SIP URI: sip:kam13.x2one.com:8443;transport=ws I/flutter (21450): [2019-11-18 16:03:16.378] Level.debug ua.dart:103 ::: new() [configuration:Instance of 'Settings'] I/flutter (21450): [2019-11-18 16:03:16.384] Level.info config.dart:246 ::: Check mandatory parameter => sockets. I/flutter (21450): [2019-11-18 16:03:16.409] Level.info config.dart:246 ::: Check mandatory parameter => uri. I/flutter (21450): [2019-11-18 16:03:16.418] Level.debug config.dart:252 ::: Check optional parameter => authorization_user. I/flutter (21450): [2019-11-18 16:03:16.427] Level.debug config.dart:252 ::: Check optional parameter => user_agent. I/flutter (21450): [2019-11-18 16:03:16.431] Level.debug config.dart:252 ::: Check optional parameter => connection_recovery_max_interval. I/flutter (21450): [2019-11-18 16:03:16.436] Level.debug config.dart:252 ::: Check optional parameter => connection_recovery_min_interval. I/flutter (21450): [2019-11-18 16:03:16.441] Level.debug config.dart:252 ::: Check optional parameter => contact_uri. I/flutter (21450): [2019-11-18 16:03:16.446] Level.debug config.dart:252 ::: Check optional parameter => display_name. I/flutter (21450): [2019-11-18 16:03:16.451] Level.debug config.dart:252 ::: Check optional parameter => instance_id. I/flutter (21450): [2019-11-18 16:03:16.455] Level.debug config.dart:252 ::: Check optional parameter => no_answer_timeout. I/flutter (21450): [2019-11-18 16:03:16.460] Level.debug config.dart:252 ::: Check optional parameter => session_timers. I/flutter (21450): [2019-11-18 16:03:16.465] Level.debug config.dart:252 ::: Check optional parameter => session_timers_refresh_method. I/flutter (21450): [2019-11-18 16:03:16.469] Level.debug config.dart:252 ::: Check optional parameter => password. I/flutter (21450): [2019-11-18 16:03:16.475] Level.debug config.dart:252 ::: Check optional parameter => realm. I/flutter (21450): [2019-11-18 16:03:16.479] Level.debug config.dart:252 ::: Check optional parameter => ha1. I/flutter (21450): [2019-11-18 16:03:16.484] Level.debug config.dart:252 ::: Check optional parameter => register. I/flutter (21450): [2019-11-18 16:03:16.490] Level.debug config.dart:252 ::: Check optional parameter => register_expires. I/flutter (21450): [2019-11-18 16:03:16.496] Level.debug config.dart:252 ::: Check optional parameter => registrar_server. I/flutter (21450): [2019-11-18 16:03:16.501] Level.debug config.dart:252 ::: Check optional parameter => use_preloaded_route. I/flutter (21450): [2019-11-18 16:03:16.507] Level.debug transport.dart:53 ::: new() I/flutter (21450): [2019-11-18 16:03:16.516] Level.debug ua.dart:163 ::: start() I/flutter (21450): [2019-11-18 16:03:16.520] Level.debug transport.dart:111 ::: connect() I/flutter (21450): [2019-11-18 16:03:16.524] Level.debug ua.dart:844 ::: Transport connecting I/flutter (21450): [2019-11-18 16:03:16.529] Level.debug sip_ua_helper.dart:146 ::: connecting => Instance of 'EventSocketConnecting' I/flutter (21450): [2019-11-18 16:03:16.534] Level.debug websocket_interface.dart:104 ::: connect() I/flutter (21450): [2019-11-18 16:03:16.549] Level.debug websocket_interface.dart:115 ::: connecting to WebSocket wss://kam13.x2one.com:8443 I/flutter (21450): [2019-11-18 16:03:16.620] Level.error websocket_interface.dart:139 ::: HandshakeException: Handshake error in client (OS Error: I/flutter (21450): CERTIFICATE_VERIFY_FAILED: unable to get local issuer certificate(handshake.cc:352)) I/flutter (21450): #0 WebSocketInterface.connect (package:sip_ua/src/websocket_interface.dart:117:18) I/flutter (21450): I/flutter (21450): #1 Transport.connect (package:sip_ua/src/transport.dart:132:19) I/flutter (21450): #2 UA.start (package:sip_ua/src/ua.dart:166:23) I/flutter (21450): #3 SIPUAHelper.start (package:sip_ua/src/sip_ua_helper.dart:197:16) I/flutter (21450): I/flutter (21450): #4 _MyRegisterWidget._handleSave (package:dart_sip_ua_example/src/register.dart:137:12) I/flutter (21450): #5 _MyRegisterWidget.build. (package:dart_sip_ua_example/src/register.dart:320:44) I/flutter (21450): #6 _InkResponseState._handleTap (package:flutter/src/material/ink_well.dart:654:14) I/flutter (21450): #7 _InkResponseState.build. (package:flutter/src/material/ink_well.dart:729:32) I/flutter (21450): #8 GestureRecognizer.invokeCallback (package:flutter/src/gestures/recognizer.dart:182:24) I/flutter (21450): #9 TapGestureRecognizer._checkUp (package:flutter/src/gestures/tap.dart:365:11) I/flutter (21450): #10 TapGestureRecognizer.handlePrimaryPointer (package:flutter/src/gestures/tap.dart:275:7) I/flutter (21450): #11
    I/flutter (21450): [2019-11-18 16:03:16.628] Level.error websocket_interface.dart:223 ::: WebSocket wss://kam13.x2one.com:8443 error: HandshakeException: Handshake error in client (OS Error: I/flutter (21450): CERTIFICATE_VERIFY_FAILED: unable to get local issuer certificate(handshake.cc:352))

    Thanks, Oleg

    opened by olegshentz1 14
  • Initiating a call takes very long on most recent commit

    Initiating a call takes very long on most recent commit

    Describe the bug When starting a call, there's a very long delay until the actual call is made. It seems fine at ae529ee5985b7bfb5f04f547e0c3df08defea5ef, but is slow at 46f74b0bbcf468956fb1f40362bca369ae54a54f.

    Expected behavior Calls should work quicker, as in previous versions of the library.

    bug wontfix 
    opened by wilkomanger 12
  • Unable to make a call - [488] Not Acceptable Here

    Unable to make a call - [488] Not Acceptable Here

    Hi there, I recently added the WebRTC support configurations to my asterisk server following this tutorial. I'm able to successfully register an account but when making a call, it disconnects after 4 secs. Not sure where the problem is, but here are the logs:

    I/flutter ( 6828): [2020-11-08 20:21:29.657] Level.debug rtc_session.dart:2215 ::: receiveInviteResponse() I/flutter ( 6828): [2020-11-08 20:21:29.670] Level.debug websocket_interface.dart:164 ::: Received WebSocket message I/flutter ( 6828): [2020-11-08 20:21:29.677] Level.debug transport.dart:314 ::: received text message: I/flutter ( 6828): I/flutter ( 6828): SIP/2.0 488 Not Acceptable Here I/flutter ( 6828): Via: SIP/2.0/WSS ri49964770ws.invalid;rport=25927;received=154.160.30.221;branch=z9hG4bK3196774880000000 I/flutter ( 6828): Call-ID: 08v07i4mcclzclhciae3 I/flutter ( 6828): From: "Tester One" sip:[email protected];tag=snyrikchig I/flutter ( 6828): To: sip:[email protected];tag=0466db44-87a7-4ad7-ba49-6dff7fa39a94 I/flutter ( 6828): CSeq: 4932 INVITE I/flutter ( 6828): Server: Asterisk PBX 17.3.0 I/flutter ( 6828): Content-Length: 0 I/flutter ( 6828): I/flutter ( 6828): [2020-11-08 20:21:29.722] Level.debug rtc_session.dart:2215 ::: receiveInviteResponse() I/flutter ( 6828): [2020-11-08 20:21:29.727] Level.debug rtc_session.dart:2868 ::: session failed I/flutter ( 6828): [2020-11-08 20:21:29.733] Level.debug rtc_session.dart:2871 ::: emit "_failed" I/flutter ( 6828): [2020-11-08 20:21:29.738] Level.debug rtc_session.dart:1450 ::: close() D/FlutterWebRTCPlugin( 6828): onConnectionChangeCLOSED I/flutter ( 6828): [2020-11-08 20:21:29.744] Level.debug rtc_session.dart:2882 ::: emit "failed" I/org.webrtc.Logging( 6828): NetworkMonitor: Stop monitoring with native observer 492335202624 I/org.webrtc.Logging( 6828): NetworkMonitorAutoDetect: Unregister network callback D/ConnectivityManager( 6828): unregisterNetworkCallback; CallingUid : 10060, CallingPid : 6828 I/flutter ( 6828): [2020-11-08 20:21:29.750] Level.debug sip_ua_helper.dart:200 ::: call failed with cause: Code: [488], Cause: Incompatible SDP, Reason: Not Acceptable Here I/flutter ( 6828): [2020-11-08 20:21:29.757] Level.debug invite_client.dart:91 ::: Timer D expired for transaction z9hG4bK3196774880000000 D/RTCAudioManager( 6828): stop`

    Any help will be greatly appreciated. Thanks alot and great work!

    opened by mrkobby 11
  • call failed with cause: Code: [488], Cause: Incompatible SDP, Reason: Not Acceptable Here

    call failed with cause: Code: [488], Cause: Incompatible SDP, Reason: Not Acceptable Here

    When trying to make a call, I am getting the error call failed with cause: Code: [488], Cause: Incompatible SDP, Reason: Not Acceptable Here. I emailed my provider, Telnyx, and they responded...

    "So a 488 response is generally related to a bad media parameter in your SIP INVITEs SDP body, which our system does not like. Upon inspection, your SDP is quite big and you can see the pasted version below. It's hard to tell which one's could be the parameters we don't like but what catches my attention straight away are the candidates. it appears this SIP Client is listing candidates in IPV6 and I'm pretty sure we don't support this. The other candidate, that is IPV4, is a private IP address as-well. If you can remove the IPV6 candidates or change them into public IPV4 addresses, that should do the trick."

    Is it possible to force IPV4 use in the config when initiating a call? Thanks!

    opened by stuartwk 11
  • Interactive testing with popular open source SIP servers.

    Interactive testing with popular open source SIP servers.

    Test the following servers in order, register/un-register/(audio/video)call/hold/unhold, etc.

    • [x] Asterisk Dockerfile https://github.com/flutter-webrtc/dockers/tree/main/asterisk
    • [x] FreeSWITCH
    • [x] OpenSIPS
    • [x] reSiprocate
    git clone https://github.com/resiprocate/resiprocate.git
    cd resiprocate
    autoreconf --install
    ./configure --with-ssl --with-repro CXXFLAGS="-I`pwd`/contrib/cajun/include"
    cd repro
    make
    ./repro
    
    • [ ] Kamailio
    help wanted wontfix 
    opened by cloudwebrtc 11
  • migrate to null safety

    migrate to null safety

    I have updated all project files to support null safety. This merge still needs to be checked and tested for errors. But the project is already started and running.

    opened by viplifes 10
  • running the exmaple results in a fatal error while calling

    running the exmaple results in a fatal error while calling

    Describe the bug Registration was successfull, When starting the call the call dialog comes and then the whole app crashes with following log

    I/flutter (28112): [2021-01-29 23:21:23.418] Level.debug rtc_session.dart:1572 ::: emit "peerconnection" I/flutter (28112): [2021-01-29 23:21:23.421] Level.debug rtc_session.dart:2826 ::: newRTCSession() I/flutter (28112): [2021-01-29 23:21:23.428] Level.debug sip_ua_helper.dart:155 ::: newRTCSession => Instance of 'EventNewRTCSession' I/org.webrtc.Logging(28112): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(28112): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(28112): EglBase14Impl: Using OpenGL ES version 2 I/org.webrtc.Logging(28112): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(28112): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(28112): EglBase14Impl: Using OpenGL ES version 2 I/ViewRootImpl@6f40bb8MainActivity: MSG_WINDOW_FOCUS_CHANGED 0 1 D/InputMethodManager(28112): prepareNavigationBarInfo() DecorView@b02cecc[MainActivity] D/InputMethodManager(28112): getNavigationBarColor() -855310 D/InputTransport(28112): Input channel destroyed: 'ClientS', fd=93 I/ViewRootImpl@6f40bb8MainActivity: stopped(false) old=false I/FlutterWebRTCPlugin(28112): getUserMedia(audio): mandatory: [], optional: [googNoiseSuppression: true, googEchoCancellation: true, echoCancellation: true, googEchoCancellation2: true, googDAEchoCancellation: true] E/rtc (28112): E/rtc (28112): E/rtc (28112): # E/rtc (28112): # Fatal error in: ../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/pc/peer_connection.cc, line 1989 E/rtc (28112): # last system error: 0 E/rtc (28112): # Check failed: IsUnifiedPlan() E/rtc (28112): # GetTransceivers is only supported with Unified Plan SdpSemantics. F/libc (28112): Fatal signal 6 (SIGABRT), code -1 (SI_QUEUE) in tid 28801 (signaling_threa), pid 28112 (_sip_ua_example)


    To Reproduce Steps to reproduce the behavior:

    1. Run the example on flutter dev channel (had to remove androidx from the create script)
    2. Run the app
    3. Register with the sip server
    4. Enter the number to call and press call
    5. Crash happens

    Expected behavior Call to be connected without a crash

    System Infomation() Flutter SDK Version: 1.26.0-17.1.pre Target OS and Version: android 10 Host OS and Version: Mac OS X 10.13.6

    bug 
    opened by shamis 10
  • Websocket message queue using streams and Delay between messages

    Websocket message queue using streams and Delay between messages

    After INVITE from an app, server sends proxy-authorization challenge and app sends ACK and INVITE with proxy-authorization. However sometimes the Server doesn't recieve the second invite(It recieves the ACK tho). The delay fixes this issue, possibly because server needs delay between these two messages.

    opened by avnotaklu 0
  • Can we have the Option to disable the RTP_Timeout?

    Can we have the Option to disable the RTP_Timeout?

    we have the requirement to not get disconnected. for that we need to Commented the RTP_Timeout line in flutter pub code. if we have the option to disable/enable the RTP_Timeout Feature as method. then it will helpful for other developer.

    opened by pixcelnettechnology 0
  • call failed with cause: Code: [603], Cause: Rejected, Reason: Decline

    call failed with cause: Code: [603], Cause: Rejected, Reason: Decline

    Whenever changing to custom screen and even if the status is REGISTERED, getting the logs for call initiated and connecting but not redirecting to call screen. Gives call failed with cause: Code: [603], Cause: Rejected, Reason: Decline error when re calling after the navigation to custom screen and getting back to dialer screen again

    opened by PrarthanaBhatt 0
  • Default logger is broken in 0.5.3

    Default logger is broken in 0.5.3

    Describe the bug Not defining a custom Logger makes it use the PrettyPrinter of the logger package instead MyLogPrinter provided by this package. Also loggingLevel has no effect because of this.

    To Reproduce Do anything that causes a log. Do not use a custom logger. Also notice that changing loggingLevel doesn't do anything

    Expected behavior I expect it to work like in 0.5.2 or before. Now, it's not using the provided methods of the Log class anymore, (like debug, warn etc.) therefore bypassing the autoInit and the factories.

    System Infomation() Flutter SDK Version: Flutter (Channel stable, 3.3.8) Target OS and Version: (flutter web) Chrome Version 107.0.5304.107 (Official Build) (64-bit) Host OS and Version: Microsoft Windows [Version 10.0.19044.2251]

    bug 
    opened by IvoB1987 2
  • can't send keepAlive request to server

    can't send keepAlive request to server

    Describe the bug A clear and concise description of what the bug is.

    To Reproduce Steps to reproduce the behavior:

    1. Go to '...'
    2. Click on '....'
    3. Scroll down to '....'
    4. See error

    Expected behavior A clear and concise description of what you expected to happen.

    System Infomation() Flutter SDK Version: Target OS and Version: Host OS and Version:

    bug 
    opened by kidusG12 0
Releases(0.5.3)
  • 0.5.3(Oct 19, 2022)

    [0.5.3] - 2022.10.19

    • Bump version for flutter-webrtc
    • Fix nullability in subscriber
    • Fix flutter test
    • Fix subscription parsing grammar
    • Added ability to supply custom logger
    • Added ability to get call statistics
    Source code(tar.gz)
    Source code(zip)
  • 0.5.2(Aug 5, 2022)

  • 0.5.1(Feb 12, 2022)

  • v0.5.0(Feb 7, 2022)

  • 0.4.0(Oct 13, 2021)

  • 0.3.9(Sep 27, 2021)

  • 0.3.8(Sep 26, 2021)


    [0.3.8] - 2021.09.26

    • Fix ice delay.
    • Don't run ready if the session has been terminated (#226)
    • Support IceRestart when IceStateDisconnected (#218)
    • Add options to the hangup (#224)
    • Adaptive when answering audio or video calls.
    Source code(tar.gz)
    Source code(zip)
  • 0.3.7(Aug 25, 2021)

  • 0.3.6(Aug 25, 2021)

    [0.3.6] - 2021.08.24

    Support custom MediaStream for call/answer. Fix the exception caused by speaker operation in web mode. bump dependencies (#216) Fix the parameters with double quotes in the Authentication header, and the unknown parameters are saved to auth_params. updated crypto and uuid versions (#188) Update dependency sdp_transform to ^0.3.0 Fixed mute audio for unified-plan Add remote_has_audio/video method for Call. Configuring via_transport.

    Source code(tar.gz)
    Source code(zip)
  • 0.3.5(Feb 3, 2021)

    [0.3.5] - 2021.02.03

    • Upgrade flutter-webrtc to 0.5.8.
    • Set sdpSemantics (plan-b or unfied-plan) to unfied-plan by default.
    • Add correct transport param to contact uri. close #161, close #160.
    • Let the user override the call options by extending SIPUAHelper (#170).
    Source code(tar.gz)
    Source code(zip)
  • 0.3.2(Nov 11, 2020)

    [0.3.2] - 2020.11.11

    • Added dtmf options to Call (#154)
    • Fix bug for digest authentication.
    • Fix rport parse (#144).
    • Support RFC2833.
    • Upgrade flutter_webrtc to 0.4.1.
    • Fix incorrect register assert (#139).
    Source code(tar.gz)
    Source code(zip)
  • 0.3.1(Oct 18, 2020)

    [0.3.1] - 2020.10.18

    • fix rport in Via parser.

    [0.3.0] - 2020.10.18

    • Upgrade flutter_webrtc to 0.4.0
    • Get more pub points (#138)
    • Fix Uri.parse
    • Force use case sensitivity in Websocket Upgrade to be compatible with old SIP servers
    • Expose Register Expires setting and if Register at all (Thanks ghenry@SureVoIP)
    • extraContactUriParams now working and tested against OpenSIPS 3.1 that has RFC8599 support (Thanks ghenry@SureVoIP)
    Source code(tar.gz)
    Source code(zip)
  • 0.2.4(Aug 25, 2020)

  • 0.2.3(Aug 25, 2020)

    [0.2.3] - 2020.08.25

    • Add display_name for Call.
    • Add WebSocketSettings.
    • Fix the invalid extraHeaders in Registrator.
    • Exposed local_identity for Call.
    • Fixed Sec-WebSocket-Key keys are not 24 bytes.
    Source code(tar.gz)
    Source code(zip)
  • v0.2.2(Jul 17, 2020)

    [0.2.2] - 2020.07.16

    • Refactor call API, move answer, hangup, hold etc methos to Call class.
    • Add SIP message listener to listen the new incoming SIP text message.
    • Expose ha1 in UaSettings.
    Source code(tar.gz)
    Source code(zip)
  • v0.2.1(Jun 12, 2020)

  • v0.2.0(May 27, 2020)

    [0.2.0] - 2020.05.27

    • Fixed bug for incoming call.
    • Just wait for 3 seconds for ice gathering.
    • Upgrade flutter-webrtc version to 0.2.8.
    • Prevent sharing of config between different UA instances.
    Source code(tar.gz)
    Source code(zip)
Owner
Flutter WebRTC
About flutter's voip, webrtc related solutions.
Flutter WebRTC
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