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Black Screen for Video #26

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GoogleCodeExporter opened this issue Sep 6, 2015 · 1 comment
Open

Black Screen for Video #26

GoogleCodeExporter opened this issue Sep 6, 2015 · 1 comment
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auto-migrated Priority-Medium Type-Defect

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@GoogleCodeExporter
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i have installed telepresence following techinical guide i am able to start and 
join using http://conf-call.org/ but on startup i see no video instead only 
black screen pls help me to resolve this issue here is my log 

my OS:fedora 13
browser :chrome
[root@igstdev009 telepresence]# /usr/local/sbin/telepresence 
*******************************************************************
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
*INFO: [TELEPRESENCE] [CFG] debug-audio-loopback = no
*INFO: [TELEPRESENCE] [CFG] accept-sip-reg = yes
*INFO: [TELEPRESENCE] [CFG] transport = udp;*;20060;*
*INFO: [TELEPRESENCE] [CFG] transport = udp://*:20060@*
*INFO: [TELEPRESENCE] [CFG] transport = ws;*;20060;*
*INFO: [TELEPRESENCE] [CFG] transport = ws://*:20060@*
*INFO: [TELEPRESENCE] [CFG] transport = http;*;20065;*
*INFO: [TELEPRESENCE] [CFG] transport = http://*:20065@*
*INFO: [TELEPRESENCE] [CFG] transport = https;*;20066;*
*INFO: [TELEPRESENCE] [CFG] transport = https://*:20066@*
*INFO: [TELEPRESENCE] [CFG] rtp-symmetric-enabled = yes
*INFO: [TELEPRESENCE] [CFG] ice-enabled = no
*INFO: [TELEPRESENCE] [CFG] icestun-enabled = yes
*INFO: [TELEPRESENCE] [CFG] stun-server = 
stun.l.google.com;19302;[email protected];stun-password
*INFO: [TELEPRESENCE] [CFG] stun-server = stun.l.google.com;19302;-;-
*INFO: [TELEPRESENCE] [CFG] rtcp-mux-enabled = yes
*INFO: [TELEPRESENCE] [CFG] rtp-buffersize = 65535
*INFO: [TELEPRESENCE] [CFG] avpf-tail-length = 200;500
*INFO: [TELEPRESENCE] [CFG] codecs = pcma;pcmu;opus;vp8;h264-bp;h264-mp
*INFO: UnRegister codec: PCMA, G.711a codec (native)
*INFO: UnRegister codec: PCMU, G.711u codec (native)
*INFO: UnRegister codec: opus, opus Codec
*INFO: UnRegister codec: VP8, VP8 codec (libvpx)
*INFO: UnRegister codec: H264, H264 Base Profile (FFmpeg, x264)
*INFO: UnRegister codec: H264, H264 Main Profile (FFmpeg, x264)
*INFO: [TELEPRESENCE] [CFG] codec-opus-maxrates = 48000;48000
*INFO: [TELEPRESENCE] [CFG] congestion-ctrl-enabled = yes
*INFO: [TELEPRESENCE] [CFG] video-max-upload-bandwidth = -1
*INFO: [TELEPRESENCE] [CFG] video-max-download-bandwidth = -1
*INFO: [TELEPRESENCE] [CFG] video-motion-rank = 2
*INFO: [TELEPRESENCE] [CFG] video-fps = 15
*INFO: [TELEPRESENCE] [CFG] video-jb-enabled = yes
*INFO: [TELEPRESENCE] [CFG] video-zeroartifacts-enabled = yes
*INFO: [TELEPRESENCE] [CFG] video-mixed-size = qvga
*INFO: [TELEPRESENCE] [CFG] video-speaker-par = 0:0
*INFO: [TELEPRESENCE] [CFG] video-listener-par = 1:1
*INFO: [TELEPRESENCE] [CFG] audio-channels = 1
*INFO: [TELEPRESENCE] [CFG] audio-bits-per-sample = 16
*INFO: [TELEPRESENCE] [CFG] audio-sample-rate = 8000
*INFO: [TELEPRESENCE] [CFG] audio-ptime = 20
*INFO: [TELEPRESENCE] [CFG] audio-volume = 1.0f
*INFO: [TELEPRESENCE] [CFG] audio-dim = 2d
*INFO: [TELEPRESENCE] [CFG] audio-max-latency = 200
*INFO: [TELEPRESENCE] [CFG] record = no
*INFO: [TELEPRESENCE] [CFG] record-file-ext = avi
*INFO: [TELEPRESENCE] [CFG] overlay-fonts-folder-path = 
./fonts/truetype/freefont
*INFO: [TELEPRESENCE] [CFG] overlay-copyright-text = Doubango Telecom
*INFO: [TELEPRESENCE] [CFG] overlay-copyright-fontsize = 12
*INFO: [TELEPRESENCE] [CFG] overlay-copyright-fontfile = FreeSerif.ttf
*INFO: [TELEPRESENCE] [CFG] overlay-speaker-name-enabled = yes
*INFO: [TELEPRESENCE] [CFG] overlay-speaker-name-fontsize = 16
*INFO: [TELEPRESENCE] [CFG] overlay-speaker-name-fontfile = FreeMonoBold.ttf
*INFO: [TELEPRESENCE] [CFG] overlay-speaker-jobtitle-enabled = yes
*INFO: [TELEPRESENCE] [CFG] overlay-watermark-image-path = 
./images/logo35x34.jpg
*INFO: [TELEPRESENCE] [CFG] ssl-private-key = /tmp/ssl.pem
*INFO: [TELEPRESENCE] [CFG] ssl-public-key = /tmp/ssl.pem
*INFO: [TELEPRESENCE] [CFG] ssl-ca = /tmp/ssl.pem
*INFO: [TELEPRESENCE] [CFG] srtp-mode = optional
*INFO: [TELEPRESENCE] [CFG] srtp-type = sdes;dtls
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-enabled = yes
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-process-local-port = 2083
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-base-folder = ./presentations
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-app = 
/opt/openoffice4/program/soffice
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10060' added
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10061' added
*INFO: [TELEPRESENCE] popen(/opt/openoffice4/program/soffice -norestore 
-headless -nofirststartwizard -invisible 
"-accept=socket,host=localhost,port=2083;urp;StarOffice.ServiceManager")
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=6
*INFO: Socket added[TCP/IPv4 transport]: fd=6, tail.count=1
*INFO: master fd=3
*INFO: Socket added[TCP/IPv4 transport]: fd=3, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=8
*INFO: Socket added[TLS/IPv4 transport]: fd=8, tail.count=1
*INFO: master fd=4
*INFO: Socket added[TLS/IPv4 transport]: fd=4, tail.count=2
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Timer manager run()::enter
*INFO: Transport::run() - enter
*INFO: TIMER MANAGER -- START
*INFO: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {20065} 
using fd {3} with type {9}...
*INFO: Transport::run() - enter
*INFO: Starting [TLS/IPv4 transport] server with IP {0.0.0.0} on port {20066} 
using fd {4} with type {17}...
*INFO: SIP STACK::run -- START
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=12
*INFO: Socket added[SIP transport]: fd=12, tail.count=1
*INFO: master fd=10
*INFO: Socket added[SIP transport]: fd=10, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=14
*INFO: Transport::run() - enter
*INFO: Socket added[SIP transport]: fd=14, tail.count=1
*INFO: master fd=11
*INFO: Socket added[SIP transport]: fd=11, tail.count=2
*INFO: Starting [SIP transport] server with IP {192.168.2.59} on port {20060} 
using fd {10} with type {2}...
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {192.168.2.59} on port {20060} 
using fd {11} with type {64}...
*INFO: SIP STACK -- START
*INFO: ioctlt(11), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=16)
*INFO: Socket added[SIP transport]: fd=16, tail.count=3
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 16
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket Peer accepted/connected with fd = 16
*INFO: *** Stream Peer destroyed ***
*INFO: #0 peers in the 'SIP transport' transport
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: 192.168.2.59:20060
Origin: http://www.conf-call.org
Sec-WebSocket-Protocol: sip
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: wpyY4AqmQtKPwc4alVVYZA==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like Gecko) 
Chrome/29.0.1547.57 Safari/537.36


*INFO: Receiving SIP o/ WebSocket message: (null)
***ERROR: function: "tsip_message_parser_execute()" 
file: "src/parsers/tsip_parser_message.c" 
line: "466" 
MSG: Failed to parse header - TP-BridgePin: 
TP-AudioPosition: [0.0f, 0.0f, 0.0f]
TP-AudioVelocity: [0.0f, 0.0f, 0.0f]
Organization: Doubango Telecom

v=0
o=- 2614386048560681500 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea
m=audio 52195 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 192.168.2.52
a=rtcp:52195 IN IP4 192.168.2.52
a=candidate:2641087038 1 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:2641087038 2 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:3555210958 1 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=candidate:3555210958 2 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=ice-ufrag:TU7yNBl9hjunRnXd
a=ice-pwd:WrlY76vAW6bROI4GzF5IUS3L
a=ice-options:google-ice
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:kr4nV1EObcc2ZcGaBLpHzzjXkxcDqJubtoe1LldV
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:484786 cname:XUeRBxyMm29nk9t+
a=ssrc:484786 msid:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea 
WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1eaa0
a=ssrc:484786 mslabel:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea
a=ssrc:484786 label:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1eaa0
m=video 52195 RTP/SAVPF 100 116 117
c=IN IP4 192.168.2.52
a=rtcp:52195 IN IP4 192.168.2.52
a=candidate:2641087038 1 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:2641087038 2 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:3555210958 1 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=candidate:3555210958 2 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=ice-ufrag:TU7yNBl9hjunRnXd
a=ice-pwd:WrlY76vAW6bROI4GzF5IUS3L
a=ice-options:google-ice
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:kr4nV1EObcc2ZcGaBLpHzzjXkxcDqJubtoe1LldV
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000

*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO: Add call-id = 'de75fa0c-e85d-31e6-096a-d6cee2d5f245' to peer with local 
fd = 16
*INFO: is_ice_active=0,
is_ro_hold_resume_changed=0,
is_ro_provisional_final_matching=0,
is_ro_media_lines_changed=0,
is_ro_network_info_changed=0,
is_ro_loopback_address=0,
is_media_type_changed=0,
is_ro_codecs_changed=0

*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
*INFO: Video 'zero-artifacts' option = yes
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: tdav_codec_h264_common_deinit
*INFO: tdav_codec_h264_common_deinit
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: Remote SSRC = 484786
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: [OPUS] Trying to match [fmtp:minptime=10]
*INFO: dtls.remote.setup=passive
*INFO: dtls.remote.setup=passive
*INFO: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO: [TELEPRESENCE] No bridge with id = 10000...create new one
*INFO: [TELEPRESENCE] Create new bridge with id = '10000'
*INFO: [TELEPRESENCE] Engine contains 1 bridges(insert)
*INFO: [TELEPRESENCE] Bridge(10000).avcalls.count = 1
*INFO: State machine: s0000_Ringing_2_Connected_X_Accept
*INFO: State machine: tsip_transac_ist_Proceeding_2_Accepted_X_2xx
*INFO: max_bw_up=2147483647 kpbs, max_bw_down=2147483647 kpbs, 
congestion_ctrl_enabled=1, media_type=2
*INFO: SO_RCVBUF = 65535, SO_SNDBUF = 65535
*INFO: rtcp.remote_ip=192.168.2.52, rtcp.remote_port=52195, rtcp.local_fd=18
*INFO: rtcp.local_ip=192.168.2.59, rtcp.local_port=57627, rtcp.local_fd=19
*INFO: Socket added[RTP/RTCP Manager]: fd=19, tail.count=1
*INFO: pipeW (write site) not initialized yet.
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: srtp_use_different_keys=false
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=22
*INFO: Socket added[RTP/RTCP Manager]: fd=22, tail.count=2
*INFO: master fd=18
*INFO: Socket added[RTP/RTCP Manager]: fd=18, tail.count=3
*INFO: setActualSndCardRecordParams(ptime=20, rate=8000, channels=1)
*INFO: ProxyAudioConsumer::setActualSndCardRecordParams(ptime=20, rate=8000, 
channels=1)
*INFO: Audio denoiser to be opened(record_frame_size_samples=960, 
record_sampling_rate=48000, playback_frame_size_samples=160, 
playback_sampling_rate=8000)
*INFO: Transport::run() - enter
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.2.59} on port 
{57626} using fd {18} with type {3}...
warning: The VAD has been replaced by a hack pending a complete rewrite
*INFO: [VP8] target_bitrate=157 kbps
*INFO: max_bw_up=157 kpbs, max_bw_down=157 kpbs, congestion_ctrl_enabled=1, 
media_type=4
*INFO: SO_RCVBUF = 65535, SO_SNDBUF = 65535
*INFO: Video jitter buffer thread - ENTER
*INFO: rtcp.remote_ip=192.168.2.52, rtcp.remote_port=52195, rtcp.local_fd=20
*INFO: rtcp.local_ip=192.168.2.59, rtcp.local_port=43069, rtcp.local_fd=21
*INFO: Socket added[RTP/RTCP Manager]: fd=21, tail.count=1
*INFO: pipeW (write site) not initialized yet.
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: srtp_use_different_keys=false
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=24
*INFO: Socket added[RTP/RTCP Manager]: fd=24, tail.count=2
*INFO: master fd=20
*INFO: Socket added[RTP/RTCP Manager]: fd=20, tail.count=3
*INFO: Transport::run() - enter
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.2.59} on port 
{43068} using fd {20} with type {3}...
*INFO: [TELEPRESENCE] Bride(10000) start
*INFO: [TELEPRESENCE] Audio Mixer Start - Consumers.Count=1, Producers.Count=1
*INFO: [TELEPRESENCE]  audio pullThreadFunc ENTER 
*INFO: [TELEPRESENCE] Video Mixer Start - Consumers.Count=1, Producers.Count=1
*INFO: [TELEPRESENCE]  video pullThreadFunc ENTER (ptime = 66)
*INFO: Open speex jb (ptime=20, rate=8000)
*INFO: Default Jitter buffer margin=0
*INFO: Default Jitter max late rate=4
*INFO: New Jitter buffer margin=100
*INFO: New Jitter buffer max late rate=1
*INFO: Receiving SIP o/ WebSocket message: (null)
***ERROR: function: "tsip_message_parser_execute()" 
file: "src/parsers/tsip_parser_message.c" 
line: "466" 
MSG: Failed to parse header - TP-BridgePin: 
TP-AudioPosition: [0.0f, 0.0f, 0.0f]
TP-AudioVelocity: [0.0f, 0.0f, 0.0f]
Organization: Doubango Telecom

27.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea
m=audio 52195 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 192.168.2.52
a=rtcp:52195 IN IP4 192.168.2.52
a=candidate:2641087038 1 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:2641087038 2 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:3555210958 1 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=candidate:3555210958 2 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=ice-ufrag:TU7yNBl9hjunRnXd
a=ice-pwd:WrlY76vAW6bROI4GzF5IUS3L
a=ice-options:google-ice
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:kr4nV1EObcc2ZcGaBLpHzzjXkxcDqJubtoe1LldV
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:484786 cname:XUeRBxyMm29nk9t+
a=ssrc:484786 msid:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea 
WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1eaa0
a=ssrc:484786 mslabel:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1ea
a=ssrc:484786 label:WxuOCFUOlGOsEZv8E2C0sDxs6s8PmTxeJ1eaa0
m=video 52195 RTP/SAVPF 100 116 117
c=IN IP4 192.168.2.52
a=rtcp:52195 IN IP4 192.168.2.52
a=candidate:2641087038 1 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:2641087038 2 udp 2113937151 192.168.2.52 52195 typ host generation 0
a=candidate:3555210958 1 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=candidate:3555210958 2 tcp 1509957375 192.168.2.52 0 typ host generation 0
a=ice-ufrag:TU7yNBl9hjunRnXd
a=ice-pwd:WrlY76vAW6bROI4GzF5IUS3L
a=ice-options:google-ice
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:kr4nV1EObcc2ZcGaBLpHzzjXkxcDqJubtoe1LldV
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000

*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
*INFO: [TELEPRESENCE] [FFmpegOverlay] Create filter (text)
*INFO: [TELEPRESENCE] [FFmpegOverlay] Create filter (text)
*INFO: [TELEPRESENCE] No codec associated to video producer with id = 4 yet
*INFO: [TELEPRESENCE] Create new codec with type = 67108864
*INFO: [VP8] target_bitrate=157 kbps

Original issue reported on code.google.com by [email protected] on 27 Mar 2014 at 6:46
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I faced the same erros.  Have you fixed this ?

*INFO: Receiving SIP o/ WebSocket message: (null)
***ERROR: function: "tsip_message_parser_execute()" 
file: "src/parsers/tsip_parser_message.c" 
line: "466" 
MSG: Failed to parse header - TP-BridgePin: 
TP-AudioPosition: [0.0f, 0.0f, 0.0f]
TP-AudioVelocity: [0.0f, 0.0f, 0.0f]
Organization: Doubango Telecom

Original comment by [email protected] on 29 Jun 2015 at 2:53

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