WebRTC demo not working

[devel...@gmail.com]

Hi,

Trying the demo.

All local. The goal is to use telepresence as an audio chat room.

The www.doubango.org/conf-call test application gives:

    This appears to be Firefox SIPml-api.js:1:861
    SIPML5 API version = 2.0.2 SIPml-api.js:1:24763
    [TELEPRESENCE] tp.init() SIPml-api.js:1:24763
    User-Agent=Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:42.0) Gecko/20100101 Firefox/42.0 SIPml-api.js:1:24763
    WebSocket supported = yes SIPml-api.js:1:24763
    Navigator friendly name = firefox SIPml-api.js:1:24763
    OS friendly name = unknown SIPml-api.js:1:24763
    Have WebRTC = yes SIPml-api.js:1:24763
    Have GUM = yes SIPml-api.js:1:24763
    Engine initialized SIPml-api.js:1:24763
    [TELEPRESENCE] realm=[192.168.5.40], impi=[2003],ws_url=[ws://192.158.5.40:20060],ice_servers=null,video_size={},bandwidth={} SIPml-api.js:1:24763
    s_websocket_server_url=ws://192.158.5.40:20060 SIPml-api.js:1:24763
    s_sip_outboundproxy_url=(null) SIPml-api.js:1:24763
    b_rtcweb_breaker_enabled=no SIPml-api.js:1:24763
    b_click2call_enabled=no SIPml-api.js:1:24763
    b_early_ims=yes SIPml-api.js:1:24763
    b_enable_media_stream_cache=no SIPml-api.js:1:24763
    o_bandwidth={} SIPml-api.js:1:24763
    o_video_size={} SIPml-api.js:1:24763
    SIP stack start: proxy='ns313841.ovh.net:10062', realm='<sip:192.168.5.40>', impi='2003', impu='"2003"<sip:20...@192.168.5.40>' SIPml-api.js:1:24763
    Connecting to 'ws://192.158.5.40:20060' SIPml-api.js:1:24763
    [TELEPRESENCE] stack event = starting SIPml-api.js:1:24763
    SecurityError: The operation is insecure.

The sipml5 test application gives:

    This appears to be Firefox SIPml-api.js:1:861
    SIPML5 API version = 2.0.2 SIPml-api.js:1:24763
    Media resource https://www.doubango.org/sipml5/sounds/dtmf.wav could not be decoded. call.htm
    location=https://www.doubango.org/sipml5/call.htm?svn=241 call.htm:161:1
    User-Agent=Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:42.0) Gecko/20100101 Firefox/42.0 SIPml-api.js:1:24763
    WebSocket supported = yes SIPml-api.js:1:24763
    Navigator friendly name = firefox SIPml-api.js:1:24763
    OS friendly name = unknown SIPml-api.js:1:24763
    Have WebRTC = yes SIPml-api.js:1:24763
    Have GUM = yes SIPml-api.js:1:24763
    Engine initialized SIPml-api.js:1:24763
    Use of getPreventDefault() is deprecated.  Use defaultPrevented instead. jquery.js:2:0
    s_websocket_server_url=ws://192.158.5.40:20060 SIPml-api.js:1:24763
    s_sip_outboundproxy_url=udp://192.158.5.40:20060 SIPml-api.js:1:24763
    b_rtcweb_breaker_enabled=no SIPml-api.js:1:24763
    b_click2call_enabled=no SIPml-api.js:1:24763
    b_early_ims=yes SIPml-api.js:1:24763
    b_enable_media_stream_cache=no SIPml-api.js:1:24763
    o_bandwidth={} SIPml-api.js:1:24763
    o_video_size={} SIPml-api.js:1:24763
    SIP stack start: proxy='ns313841.ovh.net:14062', realm='<sip:192.168.5.40>', impi='2003', impu='<sip:20...@192.168.5.40>' SIPml-api.js:1:24763
    Connecting to 'ws://192.158.5.40:20060' SIPml-api.js:1:24763
    ==stack event = starting

Server: default `telepresence.cfg`, except the debug level / audio loopback.

The ports react with some logs when tested directly with netcat.

Ubuntu 15.04, libav-ffmpeg, srtp from source, everything else from Ubuntu repos.

Doubango: `./configure --prefix=/usr --with-ssl --with-srtp --with-speexdsp --with-ffmpeg --with-opus`
Telepresence: `./configure --prefix=/usr`

Installed. Cfg installed too.

What's missing?

(sorry, maybe I've posted this multiple times; I don't see the post appear)