similar to: Asterisk 11.9 with webRTC demo integration

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 am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

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

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After

Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? -- Thanks, Bhavik Patel -------------- next part -------------- An HTML

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

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, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

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

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, "David Cunningham" <dcunningham at voisonics.com> wrote: > Hello, > > Can anyone recommend a particular online WebRTC phone for testing with > Asterisk? > > We tried: > > - JsSIP, but even with the "enable video" checkbox disabled it sends video >

tengo la siguiente pagina pero no se como seguir despues del punto 22 http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html gracias! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141112/59751a87/attachment.html>

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

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

I'm having the same issue! The difference in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00

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

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia