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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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How to read Call-Info Header from Invite Message using sipml5

I use sipml5 with freeswitch and I need to detect when call should be answered automatically. The only part where I can get it from is SIP Invite message: recv=INVITE sip:username@IP:50598;transport=ws;intercom=true SIP/2.0 Via: SIP/2.0/WSS…
emte
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DTLS-DTLS is not enabled

I built the source of webrtc2sip, when i run it i got messages: SSL is enabled :) DTLS supported: yes DTLS-SRTP supported: yes Page with sipML5 connects without errors with websocket to my webrtc2sip gateway. In SIPml.Stack i'm using option:…
Kamil Robert
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How to remove unnecessary data from SIP Invite in sipML5

How can I remove unnecessary data from SIP invite in sipML5? Now it's too big when i sending it to my server (need only audio). It will accept maximum of 1,500 bytes and it must be on UDP. Could you tell me how to do it ? how to remove some codecs,…
Kamil Robert
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Making requests to ws:// from a website loaded on https

I'm using sipml5 to connect to a sip phone service and one of the setting is the service websocket server URL. the problem is that the server url is not secured (ex. ws://123.123.123.123:9999/ws) and it cannot be accessed on wss://. Because of that,…
Andrei F
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Making SIP Call immediately get 403 after ringing

I am trying to make a phonecall using sipML5 library. The apps can successfully register into the SIP server. How ever when i try to make a phone call, it says 403 Forbidden soon after ringing. Here is some screenshot Does anyone know why i am…
Jeremy
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asterisk sip gone unreachable on sipml5 page load

I have installed and configure asterisk on my server, everything is working fine, but the problem is when user connected first time following message appears on sip debug : [May 27 22:13:25] WARNING[20193]: chan_sip.c:3727 __sip_xmit: sip_xmit of…
Mandeep Singh
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Parameters list send to Event Listener Function in sipml5

Did any one know the parameter list(other than type and session) send to the event listener function when an event occurs.
Ijas Ahamed N
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WebRTC with Asterisk returns "SRTP unprotect" warning

I have this chat system that's using SIP for voice transmission (no phones, only browser to browser in the same server). The sound goes just fine in both ends, but asterisk gives a warning twice (one for each client probably): [Mar 11 09:01:27]…
Moisés
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Using SIPML5 in guest mode

I want to setup a SIPML5 client who can call my server without any authentication. The scenario is that I want my website to call my office without dialing any number or anything. I've been told I need to enable allowguest in my asterisk because if…
Mehran
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Unable to register to FreeSwitch server & unable to call SIP client (XLite) respectively using SIPml5 client

I am unable to register to FreeSwitch server & unable to call to SIP client (XLite) by using SIPml5 SIP client. Following is my HTML5 code: