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Questions tagged [sipml5]

SIPML5 is the world’s first open source HTML5 SIP client entirely written in JavaScript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites. No extension, plugin or gateway is needed. The media stack rely on WebRTC project. 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. Use it for questions about SIPML5.

After the world’s first SIP video clients for Android and iOS (early 2009), Doubango Telecom open sourced the SIPML5 Project.

list of supported features:

  • Works on Chrome, Firefox, IE, Safari, Opera and Bowser
  • Audio / Video call
  • Screen/Desktop sharing from Chrome to any SIP client
  • Instant messaging
  • Presence
  • Call Hold / Resume
  • Explicit Call transfer
  • Multi-line and multi-account
  • Dual-tone multi-frequency signaling (DTMF) using SIP INFO
  • Click-to-Call
  • SIP TelePresence (Video Group chat)
  • 3GPP IMS standards

For more information read: SIPML5 Project.

16 questions
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Video Conference MCU NAT Traversal not work

I have successfully compiled source code of doubango opentelepresence system (An open source video conference MCU) and successfully tested. I can make video calls through desktop version and using webrtc (sipml5 client) with Chrome and Firefox (with…
M.Mahdipour
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2 answers

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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1 answer

How to configure REFER call in SIPML5 WebRTC?

I am trying to make a web client for my SIP call request. I have done invite call successfully from browser. But, I am not getting how to refer to the 3rd party call through Javascript. I am using WebRTC and SIPML5. On trying…
Shachi
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1 answer

SipML5 with Kamailio as Sip Server return 488 in make Call

I have setup Kamailio with websocket module. When I register with sipML5 its going well. But returns 488 Not Acceptable Here when I trying to call. 488 means: Some aspect of the session description or the Request-URI is not acceptable, or Codec…
wanz
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failed Connection closed before receiving a handshake response sipml5

I have an application that connect to freepbx server. The application can't connect to websocket on my freepbx server. the error is : 'failed: Connection closed before receiving a handshake response sipml5' This case just happened sometime. When…
Almugezh
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vote
1 answer

Kamailio and sipML5 and unable to get a connection

I have attempted using sipML5 instead of tryit jsip but I have not been able to figure out the configuration. I am having kamailio listen on port 15000 so I changed my sipML5 to wss://21.1.1.32:15000. I see the request coming through to kamalio but…
sw007
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sipml5 do not hang up automatically at the end of a call

I have a sipml5 web client and I can successfully make a call to it. But when a caller (softphone like linphone) hangs up, the web client is not hanging the call. and the event of sipml.session.call son't show that the caller is already hang up, How…
Mehdi
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0
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1 answer

603 Declined event is coming while hangup on incoming call with sipML5 but after 4-5 sec again I am getting incoming call

I am using asterisk 15.5 as voip server and twillio trunk to make outgoing and incoming call but when I hangup on an incoming call to sip client then 603 Declined event is coming to asterisk but after 4-5 sec again I am getting incoming call…
sirvisuresh
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0
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1 answer

WebRTC + Adapter.js giving addremoteCandidate error while connecting audio call in MS Edge

I am trying to connect audio call through sipML5 API in MS edge using webrtc and adapter.js, but it gives error Timeout for addremoteCandidate. Consider sending an end-of-candidates notification. I already tried sending addIceCandidate(null) as…
Abeer Waseem
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0
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1 answer

No video in WebRTC call on smartphone(Android) via Asterisk

I had built a WebRTC system based on Asterisk and sipml5, and I could make audio calls on my smartphone(Android), but when I enables the video, the caller can get callee's video for about 5sec, and the callee cannot get video at all. Is there any…
Chino Kafuu
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0
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1 answer

Call established but no audio on both ends in sipml5

I tried to call on sipml5 through 2 browsers. Even though the call has been initiated, we can't listen anything from another side. How to tackle this issue? By the way, the browsers that I have used are; Chrome (version 42)-can place call, but…
Evan Murphy
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1 answer

can I use indexDB to store sipml5 client objects

regarding this problem: Call get disconnected while I am refreshing the SIPML5 demo page . can be found here https://groups.google.com/forum/#!msg/doubango/BlAww-8Wq4U/79Rupoa4BwAJ;context-place=searchin/doubango/page$20refresh%7Csort:date I am…
Louay Alosh
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0
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2 answers

No audio in sipML5 with Firefox 58

With the recent release of Firefox Version 58, I have encountered a no audio issue using sipML5, I suspect it has to do with the change they did where they completely removed mozSrcObejct and they recommend to use SrcObeject instead: The prefixed…
Yasser Gutierrez
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1 answer

How can I get in SIPML5 the remote display name?

I have an issue getting the remote display name with the sipML5 library. When I register the user, I set in the stack object the display_name. display_name: local_username After when I make the call I can see in my SIPml.Session.Call object the…
DustInTheSilence
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sipml5 mobile broswer use speakers instead of loud speaker

Guy, we are using sipml5 for web calling. As we need voice through speaker instead of loud speaker on mobile browser. can anyone please let me know how to fix the issue. Thanks in advance.
andrew
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