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Questions tagged [sipml]

An open source HTML5 SIP (Session Initiation Protocol) client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce sites... without an extension, plugin or gateway.

Source: https://code.google.com/p/sipml5/

This is the world's first open source HTML5 SIP client (May 12, 2012) entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce sites... No extension, plugin or gateway is needed. The media stack rely on WebRTC.

The client can be used to connect to any SIP or IMS network from your preferred browser to make and receive audio/video calls and instant messages. It's also possible to make calls to PSTN or any SIP-legacy network using webrtc2sip.org

The protocol parsers (SIP, SDP...) are highly optimized using Ragel lookup tables and is suitable for embedded systems with limited memory and low computing power.

sipML5 solution also contains webrtc2sip, click-to-call, webrtc4all and SIP TelePresence (Video Group chat) client components.

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Changing a MediaStream of RTCPeerConnection

I want to change from a audio/video stream to a "screensharing" stream: peerConnection.removeStream(streamA) // __o_j_sep... in Screenshots below peerConnection.addStream(streamB) // SSTREAM in Screenshots below streamA is a video/audio stream…
wpp
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Call quality metrics in sipML5

Does sipML provide any info about call quality? Something like dropped packets or packets arriving out of order? I have looked at sipML API documentation, but did not find anything relevant. Also looked into the Developer Tools of Firefox/Chrome,…
Marki555
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SipML5 and Asterisk returning 488 in makeCall

Trying to make a videoaudio call with SipML5 and Asterisk13, one user in Chorme and the other Firefox, but right after "Ringing"(180) the caller receives "Not acceptable here"(488). Asterisk messages: [Jan 23 11:38:27] NOTICE[11127][C-00000004]…
Moisés
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Asterisk 11 Sipml5

When I try to register my extension using sipml5 I get the following in my CLI. Please note that my asterisk and sipml5 are on the same server. [Jan 3 16:48:43] ERROR[10158]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid",…
user3158047
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asterisk Call ID in sipml5

How do I get asterisk call Id (uniqueid in cdr table) (for instance, 1487150355.465) in sipml5 client. As far as I looked, I see only https://www.doubango.org/sipml5/docgen/symbols/SIPml.Session.html#getId which has (afaic) no relation to asterisk…
user3376996
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SIPML 5 Client and SipServlets not works Using WSS

I Have Tomcat run on HTTPS. I have tried to deploy SIPML5 WebSocket Application To into my tomcat. When I tried to connect Sip Servlets using ws : ws://192.168.X.Y:5082 And Sip Servlets Config looks like :
პაატა ლომინაძე
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How can I use sipml5 with 3CX?

I try to call a 3CX extension from browser using sipml5. SIPml.init( function (e) { var stack = new SIPml.Stack({ realm: '192.168.*.**', impi: '2003', impu: 'sip:2003@192.168.2.**', password: '**********', …
irohamca
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Why sipml5 create webRTC invite request with same port for Audio RTP, Audio RTCP, Video RTP and Video RTCP?

Previously I used firefox web browser to initiate webRTC invite request. Then I observed sdp with different port numbers for audio and video channel. And I could easily get the candidates and completed ICE operations. Here I am attaching webRTC…
RajibTheKing
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how can we create connection to Asterisk using SIPml5

I want to implement a WebRTC application to be able to make calls over VoIP. My client is running the SIPml5 and in the server side I have installed and confiured the asterisk. Asterisk was tested with softphone. Please help??
Atul Tiwari
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SIPml5 one sided voice

Below is the code of my dialer. I can register and connect calls successfully with the below code. But, after call is connected only the other end (non sipml5) can hear voice. But, the sipml5 side can not hear anything.However, I could connect and…
Kamrul Khan
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webrtc getUserMedia javascript code

var video = document.querySelector("video"); var constraints = {audio: false, video: true}; navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia; function successCallback(stream) { …
haeminish
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Getting transport Error while trying to connect to Freeswitch using SipML5 api and code

I am trying to call the user 1001 registered on Twinkle using the webpage from chrome. But I am getting Terminated_X_transportError error. JavaScript code is this
Anurag Rana
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sipML5 - Negotiate rtcpMuxPolicy

According to this announcement: "As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". This will affect the next Chrome release, M57." I'm using sipml5 API to make webrtc calls (Back…
Sibin John Mattappallil
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Asterisk sslv3 alert handshake failure

I am using Ubuntu v14.04.3 LTS and Asterisk 13.3.2. When I try to call to my extension from a sipml5 client to just play a demo-congrats audio, my call gets disconnected instantly. When I check asterisk log, I got following error: [2016-08-24…
Ijas Ahamed N
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How to use sipML5 js with Mobicent SIP servlets?

I am using sipML5 for audio and video calls that use web socket. When I register a SIP account with details, it sends a request to the server for authentication with details and the server responds with Status(like 401 or 403 Forbidden etc.) in some…
vipin kumar
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