sipml5
                                
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                        The world's first HTML5 SIP client (WebRTC)
Hi, when i try to hold the call i have this error: > `NotSupportedError: removeStream not yet implemented` > > tmedia_session_jsep.prototype.close() > tmedia_session_jsep.prototype.__stop() > tmedia_session.prototype.stop() > tmedia_session_mgr.prototype.stop() > __tsip_dialog_invite_onterm() >...
I am using latest sipml5 with lastest git checkout version of freeswitch. After some trying, now I can call from freeswitch, and other part , which is linphone can ring...
I am getting this error when calling `c.pluginObj.bindEventListener(` in attachMediaStream method in the SIPml-api.js file. ``` attachMediaStream = function(c, g) { console.log("Attaching media stream"); //.. if (c.pluginObj) { c.pluginObj.bindEventListener("play", function(k)...
Hello, I would like to ask, I used sipml5 + webrtc2sip initiate a call when the call is connected, no sound, I tried to crawl local media streams, but found,...
Hi there i'm trying to make only audio connect but it's to big for my server, it must be maximum 1500 bytes. How can i make my invite smaller ?...
Using the SIPml API and a self hosted instance of webrtc2sip I am able to make incoming calls fine, but when I attempt to make an outgoing call I get...
RTCP?
Hi, I am seeing some rtcp related messages in debug , does anyone have any information regarding RTCP in sipml5 ? Is it working, broken or not really implemented?
Hi, How can I remove opus from the SDP? a=rtpmap:111 opus/48000/2 Many thanks
I was connected to the sipml5 with freeswitch. when the number of users connected to freeswitch up to 60 or so, it will appear in "Stack not started". I would...
``` What steps will reproduce the problem? The problem is reproduced unstable. The series of calls can be successful before failure. What is the expected output? What do you see...