search for: sipml

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_tran...

Hi, I use chrome and sipml5 to connect to asterisk webrtc interface using TLS. The wss connection seems ok and the SIP REGISTER sent from chrome to asterisk and the SIP response received. In the response, I get a "failed: A server must not mask any frames that it sends to the client" error msg and chrome terminate...

Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to log in to the SIPml...

Hi all ! I am using sipML live demo page (http://sipml5.org/call.htm?svn=224#) in order to test WebRTC setup on my Asterisk PBX. I am using latest SVN version of Asterisk 11 (Asterisk PBX SVN-branch-11-r417677) If I make calls from softphones (Zoiper, X-Lite), which do not support DTLS at all, I can hear the Echo Test s...

...elped me to realize I still have several more things to learn ;-) My setup is the following: - an asterisk server, - a PC, - asterisk server and PC are installed on the same LAN - sipM5 live demo outside my LAN - no NAT/PAT configuration allowing incoming communications from the outside. Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ? What would keep this from working ? > > > > Simon > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://w...

Hi All, I am trying to configure webRTC phone 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 http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and al...

...016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo ( https://www.doubango.org/sipml5/call.htm?svn=241) ? > Dne 18.2.2016 v 15:36 Olivier napsal(a): > > > > 2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > >> >> Is it implied here that...

...gt; connection may not be initiated from a page loaded over HTTPS.* > If I replace ws://123.123.123.123:8088/ws with wss:// > 123.123.123.123:8088/ws, this error message becomes with > *Disconnected: Failed to connet to the server* > > My questions are: > 1. Is wss now required by sipml5 live demo (implying wiki page is not > up-to-date) ? > > Yes, like the error says, you have to use wss on pages served via https. > Furthermore, Chrome requires the use of https when you want to use > getUserMedia. > See here: > https://developers.google.com/web/updates/2015/1...

...9;ll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent things to work. [1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > > i want try jssip > https://github.com/versatica/JsSIP > it looks like a lot "livelier" than sipml5 > > any experience with jssip? > > > Dne 18.2.2016 v 16:01 Olivier napsal(a): > > > > 2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at...