Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 21f12105f1db6745f8c70c6167e8210e023e305e..cef7cdf2becc2cfa786ee5e9e1414f9b752d148c 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1121,6 +1121,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
RTC_DCHECK(!stream_); |
stream_ = call_->CreateAudioSendStream(config_); |
RTC_CHECK(stream_); |
+ UpdateSendState(); |
} |
bool SendTelephoneEvent(int payload_type, int event, int duration_ms) { |