Open Doeon opened 8 years ago
Le 2/10/2016 2:47 PM, Doeon a écrit :
All local. The goal is to use telepresence as an audio chat room.
The www.doubango.org/conf-call http://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:2003@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. You must use secure websocket (wss://) 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:2003@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 log (default |telepresence.cfg|, except the debug level / audio loopback):
|*** Copyright (C) 2013 Doubango Telecom http://www.doubango.org PRODUCT: telepresence - the open source TelePresence System HOME PAGE: http://conf-call.org CODE SOURCE: https://code.google.com/p/telepresence/ LICENCE: GPLv3 or commercial(contact us) VERSION: 2.1.0 'quit' to quit the application.
SSL is enabled :) DTLS supported: yes DTLS-SRTP supported: yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] debug-audio-loopback = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] accept-sip-reg = no [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = udp;;20060;* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = udp://:20060@* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = ws;;20060;* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = ws://:20060@* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = http;;20065;* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = http://:20065@* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = https;;20066;* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] transport = https://:20066@* [DOUBANGO INFO]: [TELEPRESENCE] [CFG] rtp-symmetric-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] ice-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] icestun-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] stun-server = stun.l.google.com;19302;stun-user@doubango.org;stun-password [DOUBANGO INFO]: [TELEPRESENCE] [CFG] stun-server = stun.l.google.com;19302;-;- [DOUBANGO INFO]: [TELEPRESENCE] [CFG] rtcp-mux-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] rtp-buffersize = 65535 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] avpf-tail-length = 200;500 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] codecs = pcma;pcmu;opus;vp8;h264-bp;h264-mp [DOUBANGO INFO]: UnRegister codec: PCMA, G.711a codec (native) [DOUBANGO INFO]: UnRegister codec: PCMU, G.711u codec (native) [DOUBANGO INFO]: UnRegister codec: opus, opus Codec [DOUBANGO INFO]: UnRegister codec: VP8, VP8 codec (libvpx) [DOUBANGO INFO]: UnRegister codec: H264, H264 Base Profile (FFmpeg, x264) [DOUBANGO INFO]: UnRegister codec: H264, H264 Main Profile (FFmpeg, x264) [DOUBANGO INFO]: [TELEPRESENCE] [CFG] codec-opus-maxrates = 48000;48000 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] congestion-ctrl-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-max-upload-bandwidth = -1 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-max-download-bandwidth = -1 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-motion-rank = 2 *[DOUBANGO INFO]:
[CFG] video-jb-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-zeroartifacts-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-mixed-size = vga [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-speaker-par = 0:0 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] video-listener-par = 1:1 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-channels = 1 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-bits-per-sample = 16 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-sample-rate = 8000 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-ptime = 20 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-volume = 1.0f [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-dim = 2d [DOUBANGO INFO]: [TELEPRESENCE] [CFG] audio-max-latency = 200 [DOUBANGO INFO]:
[CFG] record-file-ext = avi [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-fonts-folder-path = ./fonts/truetype/freefont [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-copyright-text = Doubango Telecom [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-copyright-fontsize = 12 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-copyright-fontfile = FreeSerif.ttf [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-speaker-name-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-speaker-name-fontsize = 16 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-speaker-name-fontfile = FreeMonoBold.ttf [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-speaker-jobtitle-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] overlay-watermark-image-path = ./images/logo35x34.jpg [DOUBANGO INFO]: [TELEPRESENCE] [CFG] srtp-mode = optional [DOUBANGO INFO]: [TELEPRESENCE] [CFG] srtp-type = sdes;dtls [DOUBANGO INFO]: [TELEPRESENCE] [CFG] presentation-sharing-enabled = yes [DOUBANGO INFO]: [TELEPRESENCE] [CFG] presentation-sharing-process-local-port = 2083 [DOUBANGO INFO]: [TELEPRESENCE] [CFG] presentation-sharing-base-folder = ./presentations [DOUBANGO INFO]: [TELEPRESENCE] [CFG] presentation-sharing-app = soffice [DOUBANGO INFO]: [TELEPRESENCE]
No doc streamer implementation [DOUBANGO INFO]: tnet_transport_prepare() [DOUBANGO INFO]: pipeR fd=6, pipeW=7 [DOUBANGO INFO]: Socket added[TCP/IPv4 transport]: fd=6, tail.count=1 [DOUBANGO INFO]: master fd=4 [DOUBANGO INFO]: Socket added[TCP/IPv4 transport]: fd=4, tail.count=2 [DOUBANGO INFO]: Transport::run(TCP/IPv4 transport) - enter [DOUBANGO INFO]: tnet_transport_prepare() [DOUBANGO INFO]: pipeR fd=8, pipeW=9 [DOUBANGO INFO]: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {20065} using master fd {4} with type {9} with max_fds {1024}... [DOUBANGO INFO]: Socket added[TLS/IPv4 transport]: fd=8, tail.count=1 [DOUBANGO INFO]: master fd=5 [DOUBANGO INFO]: Socket added[TLS/IPv4 transport]: fd=5, tail.count=2 [DOUBANGO INFO]: Stack running in SERVER mode [DOUBANGO INFO]: tsk_timer_manager_start [DOUBANGO INFO]: Transport::run(TLS/IPv4 transport) - enter [DOUBANGO INFO]: Starting [TLS/IPv4 transport] server with IP {0.0.0.0} on port {20066} using master fd {5} with type {17} with max_fds {1024}... [DOUBANGO INFO]: TIMER MANAGER -- START [DOUBANGO INFO]: Timer manager run()::enter [DOUBANGO INFO]: Best source at 0: 192.168.5.40 [DOUBANGO INFO]: Best source at 4: 192.168.5.40 [DOUBANGO INFO]: SIP STACK::run -- START [DOUBANGO INFO]: tnet_transport_prepare() [DOUBANGO INFO]: pipeR fd=12, pipeW=13 [DOUBANGO INFO]: Socket added[SIP transport]: fd=12, tail.count=1 [DOUBANGO INFO]: master fd=10 [DOUBANGO INFO]: Socket added[SIP transport]: fd=10, tail.count=2 [DOUBANGO INFO]: tnet_transport_prepare() [DOUBANGO INFO]: Transport::run(SIP transport) - enter [DOUBANGO INFO]: pipeR fd=14, pipeW=15 [DOUBANGO INFO]: Socket added[SIP transport]: fd=14, tail.count=1 [DOUBANGO INFO]: master fd=11 [DOUBANGO INFO]: Socket added[SIP transport]: fd=11, tail.count=2 [DOUBANGO INFO]: Transport::run(SIP transport) - enter [DOUBANGO INFO]: Starting [SIP transport] server with IP {192.168.5.40} on port {20060} using master fd {10} with type {2} with max_fds {1024}... [DOUBANGO INFO]: SIP STACK -- START [DOUBANGO INFO]: Starting [SIP transport] server with IP {192.168.5.40} on port {20060} using master fd {11} with type {64} with max_fds {1024}... |
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|
— Reply to this email directly or view it on GitHub https://github.com/DoubangoTelecom/doubango/issues/467.
Hi @Doeon
Did you get any solution for your problem ? I have been facing the same issue.
Hi @Doeon
I didn't get any error but when I click
"join"
button status will be
"connecting to mcu..."
I got same log as you pasted. If you got any solution please share.
Thanks in advance.
All local. The goal is to use telepresence as an audio chat room.
The www.doubango.org/conf-call test application gives:
The sipml5 test application gives:
Server log (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
What's missing?