Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Jump to bottom

Black Screen for Video #26

Open
GoogleCodeExporter opened this issue Aug 17, 2015 · 1 comment
Open

Black Screen for Video #26

GoogleCodeExporter opened this issue Aug 17, 2015 · 1 comment
Labels
auto-migrated Priority-Medium Type-Defect

Comments

@GoogleCodeExporter
Copy link 

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
@GoogleCodeExporter
Copy link
Author 

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

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment
Assignees
No one assigned
Labels
auto-migrated Priority-Medium Type-Defect
Projects
None yet
Milestone
No milestone
Development

No branches or pull requests
1 participant
@GoogleCodeExporter