Merge branch 'master' of https://github.com/cloudwebrtc/dart-sip-ua

Author: cloudwebrtc

CHANGELOG.md

@@ -1,6 +1,15 @@
 # Changelog
 
 --------------------------------------------
+[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).
+
 [0.3.1] - 2020.10.18
 
 * fix rport in Via parser.

lib/src/sip_ua_helper.dart

@@ -421,9 +421,9 @@ class Call {
     _session.unmute(audio, video);
   }
 
-  void sendDTMF(String tones) {
+  void sendDTMF(String tones, [Map<String, dynamic> options]) {
     assert(_session != null, 'ERROR(sendDTMF): rtc session is invalid!');
-    _session.sendDTMF(tones);
+    _session.sendDTMF(tones, options);
   }
 
   String get remote_display_name {

pubspec.yaml

@@ -1,5 +1,5 @@
 name: sip_ua
-version: 0.3.1
+version: 0.3.2
 description: A SIP UA stack for Flutter/Dart, based on flutter-webrtc, support iOS/Android/Destkop/Web.
 homepage: https://github.com/cloudwebrtc/dart-sip-ua
 environment:
@@ -9,14 +9,15 @@ environment:
 dependencies:
   crypto: ^2.1.2
   flutter_webrtc: ^0.5.7
+  intl: ^0.16.0
+  logger: ^0.9.0
   parser_error: ^0.1.1
-  sdp_transform: ^0.2.0
-  uuid: ^2.0.2
+  path: ^1.6.4
   random_string: ^2.0.0
-  logger: ^0.9.0
-  intl: ^0.16.0
   recase: ^3.0.0
-  path: ^1.6.4
+  sdp_transform: ^0.2.0
+  uuid: ^2.0.2
+  
 
 dev_dependencies:
   test: ^1.6.7