similar to: One Way Audio with WebRTC (with external asterisk)

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_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =

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

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

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, 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

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

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

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

hey guys, I am trying to add mapping functionality to the example application from the book. The mapping (mapquest''s openAPI) uses javascript to generate maps. So I was thinking that I''ll have a partial view to which i''ll pass the addresses and it''ll plot them. I''ll then put the partial view in the main view. Is this a good way of implementing this

Why would the samba list be looking for a reverse host name via IPV6? IFAIK I only have an IPv4 addr and reverse addr. I don't know if the original went to the person in france or not. Anyone else getting bounces like this? -------- Original Message -------- Subject: Returned mail: see transcript for details Date: Tue, 7 Apr 2015 13:19:19 -0700 From: Mail Delivery Subsystem

I have wondershaper to limit my upload at 400kilobits (my line is 600kbps). I do a lot of torrent seeding and I dont want my pings killed when I''m uploading so I set low prority source ports as follows (by the way, I have bittornet to only use ports 6881-6910): NOPRIOPORTSRC="6881 6882 6883 6884 6885 6886 6887 6888 6889 6890 6891 6892 6893 6894 6895 6896 6897 6898 6899 6900 6901

Hi i have a debian box working as a router.. it works quite well, now i want to give more than 1 ip.. is it possible to do it? some of them must be an open ip.. i mean.. all ports opened is it possible? how should i do it? Here is my nat.sh script just in case someone wants it.. (comments r in spanish.. and not right) Thanks in advance, #!/bin/sh echo "AthoS LaN Generando

I have a new method defined in a existing controller. All the methods I called in these are workable. But somehow the new method is not recognizable. ------------------new method I added-------------------------------------------- def push # if fair begin @product = Product.find(params[:id]) rescue logger.error("Attempt to access invalid product #{params[:id]}")

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

Why is someone not setting up an email server conform RFC's and seems to be you have an ipv6.. so set the ptr record. Hai Linda, I do have ipv4 and ipv6 on my server, and im never blocked.. but i did setup conform all rfcs.. i block servers in the same way. so yes, i did set my reverse for ipv4 and ipv6.. what most people do wrong.. 1) the hostname must have A and RR (PTR) and MX

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

I've now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients: http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc In particular, the focus is on the use of packages because that makes it faster for more people to deploy identical working systems. To get the demo running for the WebSocket client, I really only needed

I've just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to Asterisk Asterisk successfully answers the call using SAVPF, SRTP and ICE. The client is greeted by the demo This was tested in the Asterisk 11 environment described in my earlier email about SRTP build issues on the asterisk-users list. This is quite useful because it proves

Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. I made a webrtc relay with recording and dumped the SDP requests and RTP packets into files. Then I made a java decoder based on jitsi. Although the files contain all the needed info: encription keys, codec info, timestamps, etc., I could only decode one side in one of 2 conversations. For example, the RTP payload is decrypted successfully, but opus_packet_get_nb_samples() or opus_decode()