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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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sipml5 client one side not hanging up

I have a sipml5 web client and I can successfully make a call to it. But when a caller hangs up, the web client is not hanging the call. I think I missed an event for this in the sipml5 API. I got this error in chrome…
Sibin John Mattappallil
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sipml5 givin ns_error_unexpected in firefox 36.0.4 on two simultaneous incoming calls

I am getting ns_error_unexpected when there are two simultaneous "i_new_call" event occurs. Scenario 1 : When two intercom devices are pressed simultaneously, i receive two "i_new_call" event, after processing the events two icons are displayed on…
user1900266
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Trying to setup Asterisk for voice chat between website users with sipjs. But unable to configure DTLS certificates. getting "hostname: Unknown host"

I'm trying to setup Asterisk Voice chat for users with the Help of Sipjs follows the instruction given on SIPJS docs http://sipjs.com/guides/server-configuration/asterisk. Users are created and also connected. They can call each other through…
K Ravi
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Asterisk goes mute in android but works on PC

I have an asterisk for web(calls in the website only) working, on PC, but when I run the same client on my android 5 no sound, at least not in my android, it does send voice to PC, but just cant reproduce the incomming voice. Everthing seems to be…
Moisés
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no audio issue on one side of SIPml5 demo

I am using two SIPml5 demo + asterisk to make a call each other. I can hear the sound from one end but can't from the other end. I succeed once and suddenly lost one side after some changes i don't remember at all. I think I am using the same…
haeminish
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Call from web client to softphone Twinkle is received but gets disconnected at very next moment

I am using trying to call from web client SipML5 live demo page to a registered user at freeswitch. Now there are two problems. 1. Sometimes User 1002 is successfully connected and is able to make a call to user 1001 on twinkle. But call is…
Anurag Rana
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SIpml5 demo not working with asterisk 11.9.0

I am trying to configure an example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support. I have asterisk 11.9.0 installed and downloaded source of SIPml5 from…
Bhavik Patel
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sipml5 Asterisk 11.7 one way audio

I have integrated sipml5 with my asterisk server (which is publicly accessible).Am facing a one way audio on call between two sipxml5 clients. Additional Information 1.Works perfectly if one client is xlite and other is sipml5 2.ICE enabled in both…
hks1233
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WebRtc2SIP: No video is been received/transmitted when made call between chrome and a SIP client

I am a newbie to webrtc2sip. I have setup my webrtc2sip gateway and registered to sip2sip.info as my domain. The problem is when I make video calls from chrome to any SIP client(ekiga/jitsi) the call gets connected but I am unable to see videos on…
Shyam Sundar Kulkarni
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Issue on SIPML5 plugin integration on AWS with Asterisks server- 13 using WebRTC

I have faced an issue on integrating the demo of SIPML5 plugin on the Asterisks server. The Asterisks server version is "Asterisk 13.14.0". The new version of the asterisks server supports SRTP module. The plugin demo files are taken from Doubango's…
Sarath Kumar
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SIP invite is not received until 180 seconds

I am using asterisk 11.9 + Chrome 56.0 + SIPML5, Scenario: 1. Chrome receives "new_call" event from asterisk, it renders ringing icon on the screen 2. Do not answer the call, press F5 or CTRL+F5 to refresh the browser 3. Wait for 180 seconds,…
user1900266
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Automatic terminating a call when call made

here is my Console log of asterisk server [Feb 15 12:17:49] WARNING[3558][C-00000000]: res_rtp_asterisk.c:2141 dtlsetup: Could not set policies when setting up DTLS-SRTP on '0x7fd64400caa0 [Feb 15 12:17:49] WARNING[3558][C-00000000]:…
Suhani Mendapara
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Javascript :Ringtone is not playing

In java script i add function startRingTone() { try { ringtone.play(); } catch (e) { } } function stopRingTone() { try { ringtone.pause(); } catch (e) { } } …
Suhani Mendapara
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FreeSwitch WebRTC Call Termination

I have FreeSwitch working with SIP Clients for Extension to Extension Call Extension to PSTN / Gateway Call PSTN/DID to Extension Call I have configured WebRTC with SIPML5 clients and it is working on following scenarios Extension to…
For Guru
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WebRTC to PSTN call established but no audio

Basically i set up an asterisk server, connected to a sip provider to make calls to pstn or mobile networks. I have configured SIP to SIP properly because when i make calls from softphone e.g. Zoiper - Asterisk - Sip provider - Mobile network, the…
somedude27
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