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DoubangoTelecom
The world's first HTML5 SIP client (WebRTC)
Call terminated when i press hold button in Mozilla firefox latest stable update (48). And in console i see this error: TypeError: this.o_pc.removeStream is not a function What can i...
Hi, i m using asterisk 13.10 with sipml5 on chrome52. User registration works ok. but when try to call gives error "Media stream permission denied" Pls help.
When new INVITE the display name is lost and then getRemoteFriendlyName does not return the callerID(name).
not getting audio after call is established.
I am using Asterisk 13 and i am trying to dial an extension on asterisk server which playbacks an demo file. now i can hear the sound in case of...
No errors, browser says that it's accessing the speaker and the microphone, but no sound is coming out of my speakers. This is an internal app with no outside access,...
I am trying to run sipml5 call to free switch on Mac OS High Sierra 10.13.3.. I am using latest sipml5-master code from https://github.com/DoubangoTelecom/sipml5 It is working chrome browser on...
The Google api have a method 'Performing Streaming Speech Recognition on an Audio Stream' wich can use a microphone audio. How to realize speech recognizing from incoming stream audio?
488 Not acceptable here hi i got this error, when i try to make call. What can be a reason? Thanks.
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The world's first HTML5 SIP client (WebRTC)