similar to: WSS over Asterisk

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 terminates the ws connection. I've checked the asterisk debug logs, and the

2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo

2016-02-18 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 (

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 also able to register 2 extensions. I

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

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from the wiki [1]. > So I'm using [2] live demo from

Hi Alexey, This is what works for me: [http.conf]: tlsenable=yes ; enable tls - default no. tlsbindaddr=144.x.y.z:8089 ; address and port to bind to - default is bindaddr and port 8089. tlscertfile=/etc/asterisk/keys/mycert.pem ; path to the certificate file (*.pem) only. tlsprivatekey=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan

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 sound. BUT when I call from browser (I've tried latest Mozilla Firefox

Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'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

Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username:

Hello All, I'm having some trouble figuring out what the clearest way to plot my k-means clustering result on an my existing MDS. First I performed MDS on my distance matrix (note: I performed k-means on the MDS coordinates because applying a euclidean distance measure to my raw data would have been inappropriate) canto.MDS<-cmdscale(canto) I then figured out what would be my optimum

Hi I want to du a sensitivity analysis using Latin Hypercubes. But my parameters have to fulfill two conditions: 1) ranging from 0 to 1 2) have to sum up to 1 So far I am using the lhs package and am doing the following: library(lhs) ws <- improvedLHS(1000, 7) wsSums <- rowSums(ws) wss <- ws / wsSums but I think I can't do that, as after the normalization > min(wss) [1]

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from

Hi I am trying to push populate /home & subdirectories from the puppet server to all the Linux clients. I managed this with cfengine using rsync. But I am not sure how do I achieve this in puppet, do we have any inbuilt function for this. Also, is there a function for userdel like for useradd (user) groupadd(group). Any suggestion is appreciated. -- Deepak

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video