Does sipML provide any info about call quality? Something like dropped packets or packets arriving out of order? I have looked at sipML API documenta ...

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 ve ...

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.doub ...

According to this announcement: "As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". ...

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. ...

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 ...

I have got this error: In Asterisk PBX. All registration process with asterisk is successfully done. After that when I hit the call button of my ...

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 bec ...

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 cal ...

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: ...

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 mess ...

I built the source of webrtc2sip, when i run it i got messages: Page with sipML5 connects without errors with websocket to my webrtc2sip gateway. I ...

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 auth ...

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 ...

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 mis ...

I am getting ns_error_unexpected when there are two simultaneous "i_new_call" event occurs. Scenario 1 : When two intercom devices are pressed simulta ...

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-co ...

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 n ...

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 cal ...

I have installed and configure asterisk on my server, everything is working fine, but the problem is when user connected first time following message ...