gqadonis / telepresence Goto Github PK

What steps will reproduce the problem?

Use sipml5 to join a group video call with 8 participants

What is the expected output? What do you see instead?

Telepresence was crashed after about 40 minutes without no output

What version of the product are you using? On what operating system?

Ubuntu 12.04 LTS 64 bit

Please provide any additional information below.

[deleted issue]

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?


Please provide any additional information below.

What steps will reproduce the problem?
1. two imsdroid register telepresence
2. call the same bridge for audio conference
3. after a while, no voice can be heard via imsdroid

What is the expected output? What do you see instead?
Both imsdroids always talk with and listen to each other.

What version of the product are you using? On what operating system?
centos 6.5-64bit
doubango ver 1144
telepresence ver 142

Please provide any additional information below.
telepresence configure file and log file are uploaded as attachment.

What steps will reproduce the problem?
1. Build against new ffmpeg
2. ./configure && make

What is the expected output? What do you see instead?

Expected to build, here is the error:

---
source/filters/OTFilter.cc:171:68: error: too many arguments to function ‘int 
av_buffersrc_add_frame(AVFilterContext*, AVFrame*)’
make[1]: *** [telepresence-OTFilter.o] Error 1

---

What version of the product are you using? On what operating system?

ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
gcc version 4.8.2

Please provide any additional information below.

If I remove the third parameter from the call to av_buffersrc_add_frame on line 
171 of OTFilter.cc it seems to compile OK.

I would like Telepresence to support ffmpeg 2.x series.

I install tp server on centos-64bit ver6.5. webrtc is used to join the default 
tp server, where setting 'WebSocket URL' and 'Private Identity'.

Then press join button, and 'transport error' is reported. why webrtc client 
can't join the default bridge?

The problem lies in the fact that button mute does not turn off the sound of 
the caller.
Instead mutes the called user.

1.connect to test 10060 conf two different browser chrome(different hosts)
2.press mute user1 
3.now user2 CAN hear user1, and user1 can't hear user2

I guess that's a bug. Fix please

my test station: centos64,telepresence revision: 128, browser chrome 
29.0.1547.57m/chromium 31.0.1614.0

What steps will reproduce the problem?
1. Sign in into the boghe clients. Registration successful (can be seen in the 
telepresence logs)
2. Add the other client as a contact using the SIP URI in the register request
3. Open the chat box, type in some text and press the button 'send text'

What is the expected output? What do you see instead?
expected: The INVITE reaching the other client.
actual: the INVITE not reaching the other client. Instead status 600 internal 
error occuring.

What version of the product are you using? On what operating system?
2.1.0

Please provide any additional information below.
Attaching the logs as well as the configuration file.

Pasting the uname -a output for your reference
root@computeNode150:/usr/local/sbin# uname -a
Linux computeNode150 3.2.0-52-generic #78-Ubuntu SMP Fri Jul 26 16:21:44 UTC 
2013 x86_64 x86_64 x86_64 GNU/Linux

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


Please use labels and text to provide additional information.
https://groups.google.com/forum/#!topic/opentelepresence/YLeDe-DPv-Q

In my conference system, telepresence is integrated with asterisk in AS mode 
Now i got an error, when i call from Tandberg to MCU as below.
Tandberg <--> Asterisk <--> Telepresence

Telepresence version: 2.1.0
Server OS: CentOS 6.5
Telepresence and Asterisk are installed in the same PC (10.27.153.140)


#Asterisk
----------------------------------
localhost*CLI> 
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
[Feb 21 02:01:05] NOTICE[7513][C-0000000f]: chan_sip.c:10689 process_sdp: No 
compatible codecs, not accepting this offer!
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
[Feb 21 02:01:52] WARNING[7513][C-00000010]: chan_sip.c:11245 
process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps 
supported; ignoring.
    -- Executing [10063@testtest:1] Dial("SIP/AAAAAA-00000018", "SIP/10063@to_telepresence") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/10063@to_telepresence
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [10063@testtest:2] Hangup("SIP/AAAAAA-00000018", "") in new stack
  == Spawn extension (testtest, 10063, 2) exited non-zero on 'SIP/AAAAAA-00000018'
----------------------------------

#Telepresence
-----------------------------------
[root@localhost 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 = no
*INFO: [TELEPRESENCE] [CFG] transport = udp;*;20060;*
*INFO: [TELEPRESENCE] [CFG] transport = udp://*:20060@*
*INFO: [TELEPRESENCE] [CFG] transport = ws;*;20061;*
*INFO: [TELEPRESENCE] [CFG] transport = ws://*:20061@*
*INFO: [TELEPRESENCE] [CFG] transport = wss;*;20062;*
*INFO: [TELEPRESENCE] [CFG] transport = wss://*:20062@*
*INFO: [TELEPRESENCE] [CFG] transport = tcp;*;20063;*
*INFO: [TELEPRESENCE] [CFG] transport = tcp://*:20063@*
*INFO: [TELEPRESENCE] [CFG] transport = tls;*;20064;*
*INFO: [TELEPRESENCE] [CFG] transport = tls://*:20064@*
*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 = yes
*INFO: [TELEPRESENCE] [CFG] icestun-enabled = yes
*INFO: [TELEPRESENCE] [CFG] stun-server = 
111.111.111.111;19302;[email protected];stun-password
*INFO: [TELEPRESENCE] [CFG] stun-server = 111.111.111.111;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 = vga
*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] 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 = soffice
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10060' added
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10061' added
*INFO: [TELEPRESENCE] No doc streamer implementation
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=5
*INFO: Socket added[TCP/IPv4 transport]: fd=5, 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=7
*INFO: Socket added[TLS/IPv4 transport]: fd=7, 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: tnet_transport_prepare()
*INFO: pipeR fd=14
*INFO: Socket added[SIP transport]: fd=14, tail.count=1
*INFO: master fd=9
*INFO: Socket added[SIP transport]: fd=9, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=16
*INFO: Socket added[SIP transport]: fd=16, tail.count=1
*INFO: master fd=10
*INFO: Socket added[SIP transport]: fd=10, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=18
*INFO: Socket added[SIP transport]: fd=18, tail.count=1
*INFO: master fd=11
*INFO: Socket added[SIP transport]: fd=11, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=20
*INFO: Socket added[SIP transport]: fd=20, tail.count=1
*INFO: master fd=12
*INFO: Socket added[SIP transport]: fd=12, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=22
*INFO: Socket added[SIP transport]: fd=22, tail.count=1
*INFO: master fd=13
*INFO: Socket added[SIP transport]: fd=13, tail.count=2
*INFO: SIP STACK -- START
*INFO: Timer manager run()::enter
*INFO: SIP STACK::run -- START
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: TIMER MANAGER -- START
*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: Starting [SIP transport] server with IP {10.27.153.140} on port {20060} 
using fd {9} with type {2}...
*INFO: Starting [SIP transport] server with IP {10.27.153.140} on port {20063} 
using fd {10} with type {8}...
*INFO: Starting [SIP transport] server with IP {10.27.153.140} on port {20064} 
using fd {11} with type {16}...
*INFO: Starting [SIP transport] server with IP {10.27.153.140} on port {20061} 
using fd {12} with type {64}...
*INFO: Starting [SIP transport] server with IP {10.27.153.140} on port {20062} 
using fd {13} with type {128}...
*INFO: Starting [TCP/IPv4 transport] server with IP {0.0.0.0} on port {20065} 
using fd {3} with type {9}...
*INFO: 
RECV:INVITE sip:[email protected]:20060 SIP/2.0
Via: SIP/2.0/UDP 10.27.153.140:5060;branch=z9hG4bK2be33aa6;rport
Max-Forwards: 70
From: "AAAAAA" <sip:[email protected]>;tag=as11210624
To: <sip:[email protected]:20060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Thu, 20 Feb 2014 18:28:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 371

v=0
o=root 1160631484 1160631484 IN IP4 10.27.153.140
s=Asterisk PBX 12.0.0
c=IN IP4 10.27.153.140
b=CT:384
t=0 0
m=audio 18170 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17314 RTP/AVP 105
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4280D;max-fs=3840;max-br=768
a=sendrecv



*INFO: State machine: tsip_transac_ist_Started_2_Proceeding_X_INVITE
*INFO: 

SEND: SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 
10.27.153.140:5060;rport=5060;received=10.27.153.140;branch=z9hG4bK2be33aa6
From: "AAAAAA"<sip:[email protected]>;tag=as11210624
To: <sip:[email protected]:20060>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

*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
**WARN: function: "tdav_session_av_prepare()" 
file: "src/tdav_session_av.c" 
line: "422" 
MSG: DTLS-SRTP requested but no SSL certificates provided, disabling this 
option :(
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
**WARN: function: "tdav_session_av_prepare()" 
file: "src/tdav_session_av.c" 
line: "422" 
MSG: DTLS-SRTP requested but no SSL certificates provided, disabling this 
option :(
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: No codec matching for media type = 2
*INFO: Media session with media type = 'audio' is a zombie
*INFO: [H.264] Trying to match 
[fmtp:profile-level-id=4280D;max-fs=3840;max-br=768]
***ERROR: function: "tdav_codec_h264_parse_profile()" 
file: "src/codecs/h264/tdav_codec_h264_rtp.c" 
line: "63" 
MSG: I say [4280D] is an invalid profile-level-id
***ERROR: function: "tdav_codec_h264_common_sdp_att_match()" 
file: "include/tinydav/codecs/h264/tdav_codec_h264_common.h" 
line: "223" 
MSG: Not valid profile-level: profile-level-id=4280D;max-fs=3840;max-br=768
*INFO: [H.264] Trying to match 
[fmtp:profile-level-id=4280D;max-fs=3840;max-br=768]
***ERROR: function: "tdav_codec_h264_parse_profile()" 
file: "src/codecs/h264/tdav_codec_h264_rtp.c" 
line: "63" 
MSG: I say [4280D] is an invalid profile-level-id
***ERROR: function: "tdav_codec_h264_common_sdp_att_match()" 
file: "include/tinydav/codecs/h264/tdav_codec_h264_common.h" 
line: "223" 
MSG: Not valid profile-level: profile-level-id=4280D;max-fs=3840;max-br=768
*INFO: No codec matching for media type = 4
*INFO: Media session with media type = 'video' is a zombie
*INFO: State machine: tsip_transac_ist_Proceeding_2_Completed_X_300_to_699

SEND: SIP/2.0 488 Not Acceptable
Via: SIP/2.0/UDP 
10.27.153.140:5060;rport=5060;received=10.27.153.140;branch=z9hG4bK2be33aa6
From: "AAAAAA"<sip:[email protected]>;tag=as11210624
To: <sip:[email protected]:20060>;tag=998129733
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
Reason: SIP; cause=488; text="No common codecs"

*INFO: State machine: s0000_Started_2_Terminated_X_iINVITE
*INFO: === INVITE Dialog terminated ===
*INFO: State machine: tsip_transac_ist_Any_2_Terminated_X_cancel
*INFO: === IST terminated ===
*INFO: === IST terminated ===
*INFO: *** SIP Session destroyed ***
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: CloseSocket(25)
*INFO: CloseSocket(26)
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginConsumerAudio destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginConsumer destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPlugin destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginProducerAudio destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginProducer destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPlugin destroyed ***
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** tdav_session_video_t destroyed ***
*INFO: tdav_session_video_stop
*INFO: CloseSocket(27)
*INFO: CloseSocket(28)
*INFO: [TELEPRESENCE] *** OTProxyPluginConsumerVideo destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginConsumer destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPlugin destroyed ***
*INFO: twrap_producer_proxy_video_dtor()
*INFO: [TELEPRESENCE] *** OTProxyPluginProducerVideo destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPluginProducer destroyed ***
*INFO: [TELEPRESENCE] *** OTProxyPlugin destroyed ***
*INFO: ~ProxyVideoProducer
*INFO: *** RTP manager destroyed ***
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: tdav_codec_h264_common_deinit
*INFO: tdav_codec_h264_common_deinit
*INFO: *** Video session destroyed ***
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** IST destroyed ***

RECV:ACK sip:[email protected]:20060 SIP/2.0
Via: SIP/2.0/UDP 10.27.153.140:5060;branch=z9hG4bK2be33aa6;rport
Max-Forwards: 70
From: "AAAAAA" <sip:[email protected]>;tag=as11210624
To: <sip:[email protected]:20060>;tag=998129733
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.0.0
Content-Length: 0

-----------------------------------

What steps will reproduce the problem?
1. Build telepresence
2. Run telepresence
3. Make conference call from http://conf-call.org/

What is the expected output? 

Video and audio work

What do you see instead?

Black screen both side and a lot of warning in log file:
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
*INFO: [Opus] Packet loss, seq_num=31943
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"
MSG: 320 not expected as frame size. 20<>3
**WARN: function: "tdav_speex_jitterbuffer_get()"
file: "src/audio/tdav_speex_jitterbuffer.c"
line: "172"

What version of the product are you using? On what operating system?

Ubuntu 12.04 LTS 64 bit.

Please provide any additional information below.

What steps will reproduce the problem?
1. Initiate a Call.
2. Call 'hold()' on the Call Session object. -> Video becomes black, remote end 
video pauses.
3. Call 'resume()' Segementation Fault occurs for server.

What is the expected output? What do you see instead?
Call should resume normally, perhaps some indication should occur on the remote 
end that a particular user has put their video on hold.

What version of the product are you using? On what operating system?
Doubango 2.0.1089
https://telepresence.googlecode.com/svn/trunk@140

Please provide any additional information below.

What steps will reproduce the problem?
1. Connect to bridge using conf-call.org using DTLS-SRTP via Chrome.
2. Observe chrome://webrtc-internals and see that process stalls on 
iceGatheringStateChange: Checking
3. Sniff with WireShark, and filter for DTLS type packets, and observe.

- Telepresence->Client: Client Hello
- Client->Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
- Client->Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
- Client->Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
- Client->Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
- Client->Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
(Repeated with no response)

What is the expected output? What do you see instead?

When the DTLS/SRTP connection succeeds I see instead:
- Telepresence -> Client: Client Hello
- Client -> Telepresence: Server Hello, Certificate, Server Key Exchange, 
Certificate Request, Server Hello Done
- Telepresence -> Client: Certificate (Fragment), Certificate (Reassembled), 
Client Key Exchange, Certificate Verify, Change Cipher Spec, Encrypted 
Handshake Message
- Client -> Telepresence: Change Cipher Spec, Encrypted Handshake Message

What version of the product are you using? On what operating system?
Ubuntu Linux 12.04
Doubango 2.0.1089
https://telepresence.googlecode.com/svn/trunk@140

Please provide any additional information below.
I've attached the wireshark captures for both a good (Video established) and a 
bad (Black video) session.

What steps will reproduce the problem?
1.make the telepresence system
2.
3.

What is the expected output? What do you see instead?
The expected output is that make should be successful. But I see the following 
error - 

/usr/local/include/libavfilter/buffersrc.h:127:5: note: declared here
source/filters/OTFilter.cc:180:15: warning: âint 
av_buffersink_get_buffer_ref(AVFilterContext*, AVFilterBufferRef**, int)â is 
deprecated (declared at /usr/local/include/libavfilter/buffersink.h:41) 
[-Wdeprecated-declarations]
source/filters/OTFilter.cc:180:73: warning: âint 
av_buffersink_get_buffer_ref(AVFilterContext*, AVFilterBufferRef**, int)â is 
deprecated (declared at /usr/local/include/libavfilter/buffersink.h:41) 
[-Wdeprecated-declarations]
source/filters/OTFilter.cc:195:10: warning: âint 
avfilter_copy_buf_props(AVFrame*, const AVFilterBufferRef*)â is deprecated 
(declared at /usr/local/include/libavfilter/avfilter.h:1117) 
[-Wdeprecated-declarations]
source/filters/OTFilter.cc:195:53: warning: âint 
avfilter_copy_buf_props(AVFrame*, const AVFilterBufferRef*)â is deprecated 
(declared at /usr/local/include/libavfilter/avfilter.h:1117) 
[-Wdeprecated-declarations]
source/filters/OTFilter.cc:196:4: warning: âvoid 
avfilter_unref_bufferp(AVFilterBufferRef**)â is deprecated (declared at 
/usr/local/include/libavfilter/avfilter.h:236) [-Wdeprecated-declarations]
source/filters/OTFilter.cc:196:35: warning: âvoid 
avfilter_unref_bufferp(AVFilterBufferRef**)â is deprecated (declared at 
/usr/local/include/libavfilter/avfilter.h:236) [-Wdeprecated-declarations]
source/filters/OTFilter.cc:202:2: warning: âvoid 
avfilter_unref_bufferp(AVFilterBufferRef**)â is deprecated (declared at 
/usr/local/include/libavfilter/avfilter.h:236) [-Wdeprecated-declarations]
source/filters/OTFilter.cc:202:33: warning: âvoid 
avfilter_unref_bufferp(AVFilterBufferRef**)â is deprecated (declared at 
/usr/local/include/libavfilter/avfilter.h:236) [-Wdeprecated-declarations]


What version of the product are you using? On what operating system?
SIP telepresence latest svn downloaded code (2.0), Ububtu

Please provide any additional information below.

I use telepresence rev.143 and doubango 1150. 

Imsdroids join video conference. If some imsdroids leave the conference 
abnormally, there is some mechanism to delete those UA' connections in 
telepresence. After session expire time, UAS should delete UAC's connections 
and think those UACs leave the conference.

Does telepresence support session timers and refresher? And how to configure 
telepresence to support detect UA that leave conference abnormally?

Look forward to reply. Thanks!

What steps will reproduce the problem?
1. i failed to configure 

What is the expected output? What do you see instead?

./configure 
--------------this error message -------------------------
checking tinysak/tinysak_config.h usability... yes
checking tinysak/tinysak_config.h presence... yes
checking for tinysak/tinysak_config.h... yes
checking for tsk_object_new in -ltinySAK... yes
checking tinynet/tinynet_config.h usability... yes
checking tinynet/tinynet_config.h presence... yes
checking for tinynet/tinynet_config.h... yes
checking for tnet_startup in -ltinyNET... yes
checking tinyhttp/tinyhttp_config.h usability... yes
checking tinyhttp/tinyhttp_config.h presence... yes
checking for tinyhttp/tinyhttp_config.h... yes
checking for thttp_stack_create in -ltinyHTTP... yes
checking tinysip/tinysip_config.h usability... yes
checking tinysip/tinysip_config.h presence... yes
checking for tinysip/tinysip_config.h... yes
checking for tsip_stack_create in -ltinySIP... yes
checking tinydav/tinydav_config.h usability... yes
checking tinydav/tinydav_config.h presence... yes
checking for tinydav/tinydav_config.h... yes
checking for tdav_init in -ltinyDAV... no
configure: error: Failed to find libtinyDAV



What version of the product are you using? On what operating system?

  svn revision 140

Please provide any additional information below.

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


Please use labels and text to provide additional information.
https://groups.google.com/forum/#!topic/opentelepresence/X8oM7vOC1I4

I want to know the capability of telepresence. How many participants can a 
bridge handle? and how many bridges can telepresence run simultaneously?

Thanks!

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

What steps will reproduce the problem?
1.svn checkout http://telepresence.googlecode.com/svn/trunk/ telepresence
2.cd telepresence
3.svn checkout 
http://doubango.googlecode.com/svn/branches/2.0/doubango/bindings/_common
tinywrap
 Till here everything is fine.
4. ./autogen.sh && ./configure
But As I am running ./autogen.sh && ./configure then I am getting belwo error.

 ./autogen.sh && ./configure
-bash: ./autogen.sh: No such file or directory
[root@localhost tinywrap]#

I have checked both directory tinywrap and telepresence both doesn't have 
autogent.sh && configure script.


What is the expected output? What do you see instead?
As I have described above about error message, I was expecting some 
compilation, but compilation scripts are't in directory. 


What version of the product are you using? On what operating system?
I am using TelePresence System version 2.0, Operating system Cent OS 6.4

Please provide any additional information below.
Everything went fine. but only When I came to telepresence and tinywrap section 
then there I faced issue due to autogent.sh and configure script, I think it is 
little things, Which is missing, if you guys will help me then I can fix it,

Sir Waiting for your response. 

hi every one ,

i m IFFI , to try implement AS mode of Presence . mean behind asterisk,

i have successfully configure and installed Presence , but i dnt understand how 
to configure this one with asterisk , 


Can any one know how to configure this one with asterisk ,

Thanks 

I install telepresence behind NAT, and install turnserver and telepresence in 
the same server.

how to set the parameter of 'stun-server'. stun-server should be set to 
local-ip(i.e. LAN server ip), or public ip(i.e. NAT ip)?

What steps will reproduce the problem?
1.build doubango
2.build telepresence
3.

What is the expected output? What do you see instead?

From doubango config.log, required video/audio codecs has been seleceted, which 
should be expected be supported and registered when running telelpresence.

telepresence report all required codec are 'unregister codec'.

What version of the product are you using? On what operating system?

telepresence revision : 140
doubango revision : 1116
operating system : CentOS 64bit ver6.5

Please provide any additional information below.

telepresence server log, and cfg file are uploaded as attachment.

What steps will reproduce the problem?
1.Have followed all points in technical guide till 4.3.2
2.How to use WebRTC, what Bridge ID should i put in web page 
3.Need guidance to configure Telepresence after followed guide till 4.3.2

What is the expected output? What do you see instead?

I expect Configuration window using WebRTC but it is givng error "failed to 
connect" in conf-call.org

What version of the product are you using? On what operating system?

followed latest technical guide

Please provide any additional information below.

What steps will reproduce the problem
have the account sip password??
in where... i put the password on telepresence.cfg.. ???

I'm involved in build telepresence system. I'm stuck in calling the default 
bridge.

I follow the step as shown in Support_Testing_the_system 
(https://code.google.com/p/telepresence/wiki/Support_Testing_the_system)
,but fail to join the default bridge via webrtc client.

who can show me the typical call flow? 

looking forward to your help. Thanks!

What version of the product are you using? On what operating system?

Ubuntu 12.04 LTS 64 bit

Please provide any additional information below.

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


Please use labels and text to provide additional information.

how could i integrate telepresence with asterisk in AS mode? is there some 
documents about that. thanks very much!

[deleted issue]

I check 'technical-guide.pdf', where '5.5 Selecting the speaker and listeners' 
is marked as '--This section intentionally left blank--'.

Does telepresence support the feature?

What steps will reproduce the problem?
1.Using FFmpeg 1.2.3
2.Using FFmpeg 1.1.6


What is the expected output? What do you see instead?
- Expected : Working Telepresence video flow

- Instead : Green/Black screen (depending the client), and full of 
"***ERROR: function: "tdav_codec_h263_encode()" 
file: "src/codecs/h263/tdav_codec_h263.c" 
line: "297" 
MSG: Invalid parameter"

in the Telepresence log.

What version of the product are you using? On what operating system?
Telepresence rev998 hosted on a 64b Ubuntu 12.04.

Please provide any additional information below.
If I use h264 on my Linphone windows client, I have a video flow, but with h263 
(my Linphone linux doesn't have h264) I get only errors and no video.

What steps will reproduce the problem?
1.when running make command inside telepresence folder below error is generated.
/usr/bin/ld: cannot find -lopenal
collect2: ld returned 1 exit status
make[1]: *** [telepresence] Error 1
make[1]: Leaving directory `/usr/local/sft/telepresence'
make: *** [all] Error 2
2.
3.

What is the expected output? What do you see instead?
Below error  is seen when run make command inside telepresence folder
/usr/bin/ld: cannot find -lopenal
collect2: ld returned 1 exit status
make[1]: *** [telepresence] Error 1
make[1]: Leaving directory `/usr/local/sft/telepresence'
make: *** [all] Error 2


What version of the product are you using? On what operating system?

trying to install telepresence on Centos 5.3

Please provide any additional information below.
Make file does not have the path to openal. giving a symbolic link is also not 
working.
Please advice. 

What steps will reproduce the problem?
1. telepresence server is installed on LAN, follow the steps as shown in the 
section of 'Support_Testing_the_system'
2. telepresence needs ssl configuration, so I follow below steps to generate 
ssl certificates : 
https://groups.google.com/forum/#!topic/doubango/asAfP5ZCgdI
4. to extract public key from private key : openssl rsa -in key.ca.cg.pem 
-pubout > key.ca.cg.pub
5  I get ca, public key, and private key, which are crt.ca.cg.pem, 
key.ca.cg.pub, key.ca.cg.pem respectively

What is the expected output? What do you see instead?

webrtc client is expected to join the default bridge without error.

However, telepresence log show something wrong with ssl certificates. what's 
wrong with my method of key/ca generation. In addition, two much latency are 
reported. how to reduce transport latency via telepresence.cfg?

For details please reference attachment.

What version of the product are you using? On what operating system?
telepresence revision : 140
doubango revision : 1089
operating system : CentOS 64bit ver6.5

Please provide any additional information below.

telepresence/doubango config.log, wireshare file, and ssl ca/key are uploaded 
as attachment.

 I am trying to compile in centos , it getting an error 
Failed to find libtinySAK . I have libtinySAK.so.0 in the lib directory

Can any one advise me , how to resolve it?

What steps will reproduce the problem?

Clients: SIPML5 with group audio call

What is the expected output? What do you see instead?

it should working properly but telepresence crashed.

What version of the product are you using? On what operating system?

Ubuntu 12.04 LTS 64 bit

Please provide any additional information below.

Crash log.

What steps will reproduce the problem?
1. the latest linphone-android, and vp8 is turned on
2. install latest telepresence on CentOS 6.5
3. one linphone-android calls telepresence

What is the expected output? What do you see instead?
video is expected. But there is no video on linphone.

What version of the product are you using? On what operating system?
CentOS 6.5

Please provide any additional information below.
telepresence log is uploaded as attachment

What steps will reproduce the problem?
1. one imsdroid calls one conference
2. force to stop imsdroid unconditionally without leaving the conference 
normally
3. the imsdroid joins the same conference

What is the expected output? What do you see instead?
there is still one person in the conference, which is shown on the video. But 
there are two clients image, where one image is the previous one which left 
abnormally.

What version of the product are you using? On what operating system?
doubango revision is 1157, and telepresence revision is 143. Centos 6.5 64bit 
is used.

Please provide any additional information below.
none

What steps will reproduce the problem?
1.start the telepresence system
2.modify the settings of WebRTC SIP telepresence client, enter bridge ID and 
pincode and press the join button
3.

What is the expected output? What do you see instead?
expected output: the call to be successful
actual output: core dump

What version of the product are you using? On what operating system?
2.1.0

Please provide any additional information below.

I have attached the logs, gdb back trace as well as the telpresence.cfg. Please 
let me know if  you need any more information

I'm try to use telepresence rev #130.
I call test room from linphone and got an coredump:

Console and gdb log:

RECV:INFO sip:[email protected]:5070;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.53.4:5071;rport;branch=z9hG4bK13872
From: <sip:[email protected]:5071>;tag=17900
To: "test" <sip:[email protected]:5070>;tag=1578128240
Call-ID: 9750
CSeq: 21 INFO
Contact: <sip:[email protected]:5071>
Content-Type: application/media_control+xml
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/3.6.0)
Content-Length:   185

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    
<to_encoder>      <picture_fast_update></picture_fast_update>    </to_encoder>  
</vc_primitive></media_control>

*INFO: State machine: tsip_transac_nist_Started_2_Trying_X_request
*INFO: State machine: x0000_Any_2_Any_X_iINFO
*INFO: State machine: tsip_transac_nist_Trying_2_Completed_X_send_200_to_699
*INFO:

SEND: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 
192.168.53.4:5071;rport=49348;received=xx.xx.xx.xx;branch=z9hG4bK13872
From: <sip:[email protected]:5071>;tag=17900
To: "test"<sip:[email protected]:5070>;tag=1578128240
Contact: <sip:[email protected]:5070;transport=udp>
Call-ID: 9750
CSeq: 21 INFO
Content-Length: 0


Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 80460a740 (LWP 100225/telepresence)]
0x0000000803108607 in strlen () from /lib/libc.so.7
(gdb)bt
#0  0x0000000803108607 in strlen () from /lib/libc.so.7
#1  0x0000000803101262 in open () from /lib/libc.so.7
#2  0x00000008031022c9 in open () from /lib/libc.so.7
#3  0x00000008031023fc in vfprintf () from /lib/libc.so.7
#4  0x00000008030f3eb8 in fprintf () from /lib/libc.so.7
#5  0x000000000043cca9 in OTSipSessionAV::handleIncomingData (
    this=0x804be5d30, pcContentType=Variable "pcContentType" is not available.
) at source/OTWrap.cc:172
#6  0x000000000043ff66 in OTSipCallback::OnInviteEvent (this=Variable "this" is 
not available.
)
    at source/OTWrap.cc:526
#7  0x000000000047cc10 in stack_callback (sipevent=0x8055ff2e0)
    at tinywrap/SipStack.cxx:504
#8  0x0000000800abc04c in run () from /usr/local/lib/libtinySIP.so.0
#9  0x000000080409f511 in pthread_getprio () from /lib/libthr.so.3
#10 0x0000000000000000 in ?? ()
Error accessing memory address 0x7ffffeff9000: Bad address.

(gdb) bt full
#0  0x0000000803108607 in strlen () from /lib/libc.so.7
No symbol table info available.
#1  0x0000000803101262 in open () from /lib/libc.so.7
No symbol table info available.
#2  0x00000008031022c9 in open () from /lib/libc.so.7
No symbol table info available.
#3  0x00000008031023fc in vfprintf () from /lib/libc.so.7
No symbol table info available.
#4  0x00000008030f3eb8 in fprintf () from /lib/libc.so.7
No symbol table info available.
#5  0x000000000043cca9 in OTSipSessionAV::handleIncomingData (
    this=0x804be5d30, pcContentType=Variable "pcContentType" is not available.
) at source/OTWrap.cc:172
        root = {static null = {
    static null = <same as static member of an already seen type>,
    static minLargestInt = -9223372036854775808,
    static maxLargestInt = 9223372036854775807,
    static maxLargestUInt = 18446744073709551615, static minInt = -2147483648,
    static maxInt = 2147483647, static maxUInt = 4294967295,
    static minInt64 = -9223372036854775808,
    static maxInt64 = 9223372036854775807,
    static maxUInt64 = 18446744073709551615, value_ = {int_ = 0, uint_ = 0,
      real_ = 0, bool_ = false, string_ = 0x0, map_ = 0x0},
    type_ = Json::nullValue, allocated_ = 0, comments_ = 0x0},
  static minLargestInt = -9223372036854775808,
  static maxLargestInt = 9223372036854775807,
  static maxLargestUInt = 18446744073709551615, static minInt = -2147483648,
  static maxInt = 2147483647, static maxUInt = 4294967295,
  static minInt64 = -9223372036854775808,
  static maxInt64 = 9223372036854775807,
  static maxUInt64 = 18446744073709551615, value_ = {int_ = 140737471548352,
    uint_ = 140737471548352, real_ = 6.9533549774600591e-310, bool_ = 192,
    string_ = 0x7ffffeff8bc0 "\020▒▒\004\b", map_ = 0x7ffffeff8bc0},
  type_ = 48, allocated_ = -1, comments_ = 0x7ffffeff8db0}
        reader = {nodes_ = {
    c = {<std::_Deque_base<Json::Value*,std::allocator<Json::Value*> >> = {
        _M_impl = {<std::allocator<Json::Value*>> = {<__gnu_cxx::new_allocator<Json::Value*>> = {<No data fields>}, <No data fields>}, _M_map = 0x8005d0d88,
          _M_map_size = 16, _M_start = {_M_cur = 0x804bf8940,
            _M_first = 0x803096f69, _M_last = 0x8005d80b0,
            _M_node = 0x200000}, _M_finish = {_M_cur = 0x1000,
            _M_first = 0x80460a740, _M_last = 0x804bf89c0,
            _M_node = 0x7ffffd3eb000}}}, <No data fields>}},
  errors_ = {<std::_Deque_base<Json::Reader::ErrorInfo,std::allocator<Json::Reader::ErrorInfo> >> = {
      _M_impl = {<std::allocator<Json::Reader::ErrorInfo>> = {<__gnu_cxx::new_allocator<Json::Reader::ErrorInfo>> = {<No data fields>}, <No data fields>},
        _M_map = 0x201000, _M_map_size = 34427510548, _M_start = {
          _M_cur = 0x804a24b60, _M_first = 0x0, _M_last = 0x804bf8940,
          _M_node = 0x8040a89d1}, _M_finish = {_M_cur = 0x42d020,
          _M_first = 0x0, _M_last = 0x0, _M_node = 0x0}}}, <No data fields>},
  document_ = {static npos = 18446744073709551615,
    _M_dataplus = {<std::allocator<char>> = {<__gnu_cxx::new_allocator<char>> = {<No data fields>}, <No data fields>}, _M_p = 0x7ffffeff8ac0 "\220▒Z"}},
  begin_ = 0x0, end_ = 0x804a00558 "\0010▒\004\b",
  current_ = 0x38 <Error reading address 0x38: Bad address>,
  lastValueEnd_ = 0x5ae590 "\220▒Z", lastValue_ = 0x38, commentsBefore_ = {
    static npos = 18446744073709551615,
    _M_dataplus = {<std::allocator<char>> = {<__gnu_cxx::new_allocator<char>> = {<No data fields>}, <No data fields>},
      _M_p = 0x1 <Error reading address 0x1: Bad address>}}, features_ = {
    allowComments_ = false, strictRoot_ = false}, collectComments_ = false}
        parsingSuccessful = Variable "parsingSuccessful" is not available.
(gdb)

What steps will reproduce the problem?

1. Got an error "***ERROR: function: "tsip_transport_layer_ws_cb()" in console 
then try to connect..

What is the expected output? What do you see instead?

# ./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 = no
*INFO: [TELEPRESENCE] [CFG] transport = udp;*;20060;*
*INFO: [TELEPRESENCE] [CFG] transport = udp://*:20060@*
*INFO: [TELEPRESENCE] [CFG] transport = ws;*;20060;4
*INFO: [TELEPRESENCE] [CFG] transport = ws://*:20060@4
*INFO: [TELEPRESENCE] [CFG] transport = http;*;20065
*INFO: [TELEPRESENCE] [CFG] transport = https;*;20066
*INFO: [TELEPRESENCE] [CFG] rtp-symmetric-enabled = no
*INFO: [TELEPRESENCE] [CFG] ice-enabled = no
*INFO: [TELEPRESENCE] [CFG] icestun-enabled = no
*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 = g729;vp8
*INFO: 'g729' codec enabled but not supported
*INFO: 'vp8' codec enabled but not supported
*INFO: [TELEPRESENCE] [CFG] codec-opus-maxrates = 48000;48000
*INFO: [TELEPRESENCE] [CFG] congestion-ctrl-enabled = yes
*INFO: [TELEPRESENCE] [CFG] video-max-upload-bandwidth = 1890
*INFO: [TELEPRESENCE] [CFG] video-max-download-bandwidth = 1890
*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 = vga
*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/teleconf-xxx-key.pem
*INFO: [TELEPRESENCE] [CFG] ssl-public-key = /tmp/teleconf-xxx-key.pem
*INFO: [TELEPRESENCE] [CFG] ssl-ca = /tmp/teleconf-xxx-key.pem
*INFO: [TELEPRESENCE] [CFG] ssl-mutual-auth = no
*INFO: [TELEPRESENCE] [CFG] srtp-mode = optional
*INFO: [TELEPRESENCE] [CFG] srtp-type = sdes;dtls
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-enabled = no
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-process-local-port = 2083
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-base-folder = ./presentations
*INFO: [TELEPRESENCE] [CFG] presentation-sharing-app = soffice
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10060' added
*INFO: [TELEPRESENCE] [CFG] Bridge with id ='10061' added
*INFO: [TELEPRESENCE] Presentation sharing not enabled
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: SIP STACK::run -- START
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=5
*INFO: Socket added[SIP transport]: fd=5, tail.count=1
*INFO: master fd=3
*INFO: Socket added[SIP transport]: fd=3, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: Transport::run() - enter
*INFO: pipeR fd=7
*INFO: Socket added[SIP transport]: fd=7, tail.count=1
*INFO: master fd=4
*INFO: Socket added[SIP transport]: fd=4, tail.count=2
*INFO: Starting [SIP transport] server with IP {192.168.99.99} on port {20060} 
using fd {3} with type {2}...
*INFO: Transport::run() - enter
*INFO: SIP STACK -- START
*INFO: Starting [SIP transport] server with IP {192.168.99.99} on port {20060} 
using fd {4} with type {64}...
*INFO: ioctlt(4), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=9)
*INFO: Socket added[SIP transport]: fd=9, tail.count=3
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 9
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Host: 192.168.0.21:20060
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64; rv:26.0) Gecko/20100101 
Firefox/26.0
Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8
Accept-Language: ru-RU,ru;q=0.8,en-US;q=0.5,en;q=0.3
Accept-Encoding: gzip, deflate
Sec-WebSocket-Version: 13
Origin: http://xxx.yyy.zzz.com
Sec-WebSocket-Protocol: sip
Sec-WebSocket-Key: wmkDEUSqxVD7QuQSh+saXw==
Connection: keep-alive, Upgrade
Pragma: no-cache
Cache-Control: no-cache
Upgrade: websocket


*INFO: WebSocket Peer accepted/connected with fd = 9
*INFO: *** Stream Peer destroyed ***
*INFO: #0 peers in the 'SIP transport' transport
*INFO: #1 peers in the 'SIP transport' transport
***ERROR: function: "tsip_transport_layer_ws_cb()"
file: "src/transports/tsip_transport_layer.c"
line: "397"
MSG: WS handshaking not done yet
*INFO: Removing socket 9
*INFO: Socket to remove: fd=9, index=2, tail.count=3
*INFO: RemoveSocket(fd=9) has been requested but we are poll()ing the socket. 
ShutdownSocket(fd) called on the socket and we deferred the request.
*INFO: ShutdownSocket(fd=9)
*INFO: WebSocket Peer closed with fd = 9
*INFO: #0 peers in the 'SIP transport' transport
*INFO: PipeR event = 1
*INFO: *** Stream Peer destroyed ***
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLHUP(9)
*INFO: Removing socket 9
*INFO: Socket to remove: fd=9, index=2, tail.count=3
*INFO: CloseSocket(9)
*INFO: PipeR event = 1
*INFO: WebSocket Peer closed with fd = 9
*INFO: WebSocket Peer closed with fd = 9


What version of the product are you using? On what operating system?

Centos 6.5 64-bit, Mozilla Firefox 26.0

Please provide any additional information below.

How can i figure this out? 

the latest revision of telepresence is 143, and that of doubango is 1173. Does 
telepresence support session timers now?

Error compiling telepresence. The problem seems to be in file 
source/OTResampler.cc. See attached file for full error.

OS: CentOS 7.0 (x86_64), Ubuntu 12.04 (x86_64), Ubuntu 14.04 (x86_64)

Hi,

I tried to build telepresence from svn trunk, the result was:

In file included from include/opentelepresence/OTEngine.h:8:0,
                 from include/OpenTelepresenceAPI.h:14,
                 from source/main.cc:7:
include/opentelepresence/OTWrap.h:17:24: fatal error: SipSession.h: No such 
file or directory

I found file SipSession.h only in doubango/bindings/_common directory. Path 
bindings/_common is used only in *.vcproj files, but I can't see it in 
configure.ac or Makefile.am.

So, doubango make install step must be fixed to copy SipSession.h and others to 
system-wide directory like /usr/local/include and telepresence must be fixed to 
use SipSession.h from system-wide directory.

My OS is centos 64bit ver6.5.

telepresence and doubango are updated day after day.

please suggest the revision number of telepresnece and doubango which can work 
together, and pass the basic call flow of 'Support_Testing_the_system'.

Thanks!

What steps will reproduce the problem?
1. follow the steps as shown in the section of 'Support_Testing_the_system'
https://code.google.com/p/telepresence/wiki/Support_Testing_the_system
2. telepresence server is installed on LAN

What is the expected output? What do you see instead?
webrtc client is expected to join the default bridge.

webrtc client can connect telepresence server, but fails to join the default 
bridge, and reports 'transport error'

What version of the product are you using? On what operating system?
telepresence revision : 140
doubango revision : 1116
operating system : CentOS 64bit ver6.5

Please provide any additional information below.
telepresence/doubango config.log, and wireshare file are uploaded as attachment.

telepresence revision : 140
doubango revision : 1116
operating system : CentOS 64bit ver6.5

I build doubango with ffmpeg, and build telepresence. Then telepresence is run, 
but video/audio codec are in the state of unregister.

I review the source code of telepresence. 

In the file of OTEngine.cc, av_register_all() and avfilter_register_all() are 
called in the function of OTEngine::OTEngine(). And there is one log, 
"'av_register_all()' not called by Doubango as there is no dependncy with 
libavformat". What does that mean? are the two functions useless?

where to register codec in the source code of telepresence? 

Looking forward to reply! :)

What steps will reproduce the problem?
1. follow the steps as shown in the section of 'Support_Testing_the_system'
2. telepresence server is installed on LAN

What is the expected output? What do you see instead?

webrtc client is expected to join the default bridge.

webrtc client can connect telepresence server, but fails to join the default 
bridge, and reports 'transport error'

What version of the product are you using? On what operating system?

telepresence revision : 140
doubango revision : 1116
operating system : CentOS 64bit ver6.5

Please provide any additional information below.

telepresence server log, and cfg file are uploaded as attachment.

What steps will reproduce the problem?
1. Use conf-call.org to make video call with 2 participants

What do you see instead?

Telepresence was crashed.

What version of the product are you using? On what operating system?

Ubuntu 12.04 LTS 64 bit.


Please provide any additional information below.

include/opentelepresence/OTProxyPluginProducerVideo.h:48: error: expected â;â 
before âsenderThreadâ
include/opentelepresence/OTProxyPluginProducerVideo.h:55: error: ISO C++ 
forbids declaration of âtsk_semaphore_handle_tâ with no type
include/opentelepresence/OTProxyPluginProducerVideo.h:55: error: expected â;â 
before â*â token
include/opentelepresence/OTProxyPluginProducerVideo.h:56: error: ISO C++ 
forbids declaration of âtsk_mutex_handle_tâ with no type
include/opentelepresence/OTProxyPluginProducerVideo.h:56: error: expected â;â 
before â*â token
include/opentelepresence/OTProxyPluginProducerVideo.h:57: error: ISO C++ 
forbids declaration of âtsk_buffer_tâ with no type
include/opentelepresence/OTProxyPluginProducerVideo.h:57: error: expected â;â 
before â*â token
include/opentelepresence/OTProxyPluginProducerVideo.h:58: error: âtsk_size_tâ 
does not name a type
In file included from include/OpenTelepresenceAPI.h:18,
                 from source/main.cc:7:
include/opentelepresence/OTMixerMgrVideo.h:40: error: use of enum 
âtmedia_rtcp_event_type_eâ without previous declaration
include/opentelepresence/OTMixerMgrVideo.h:41: error: expected â;â before 
âpullThreadFuncâ
include/opentelepresence/OTMixerMgrVideo.h:67: error: ISO C++ forbids 
declaration of âtsk_mutex_handle_tâ with no type
include/opentelepresence/OTMixerMgrVideo.h:67: error: expected â;â before 
â*â token
include/opentelepresence/OTMixerMgrVideo.h:68: error: ISO C++ forbids 
declaration of âtsk_mutex_handle_tâ with no type
include/opentelepresence/OTMixerMgrVideo.h:68: error: expected â;â before 
â*â token
include/opentelepresence/OTMixerMgrVideo.h:75: error: ISO C++ forbids 
declaration of âtsk_thread_handle_tâ with no type
include/opentelepresence/OTMixerMgrVideo.h:75: error: expected â;â before 
â*â token
include/opentelepresence/OTMixerMgrVideo.h:76: error: ISO C++ forbids 
declaration of âtsk_condwait_handle_tâ with no type
include/opentelepresence/OTMixerMgrVideo.h:76: error: expected â;â before 
â*â token
In file included from include/OpenTelepresenceAPI.h:24,
                 from source/main.cc:7:
include/opentelepresence/OTProxyPluginMgr.h:29: error: ISO C++ forbids 
declaration of âtsk_mutex_handle_tâ with no type
include/opentelepresence/OTProxyPluginMgr.h:29: error: expected â;â before 
â*â token
include/opentelepresence/OTProxyPluginMgr.h:63: error: ISO C++ forbids 
declaration of âtsk_mutex_handle_tâ with no type
include/opentelepresence/OTProxyPluginMgr.h:63: error: âtsk_mutex_handle_tâ 
declared as an âinlineâ field
include/opentelepresence/OTProxyPluginMgr.h:63: error: expected â;â before 
â*â token
include/opentelepresence/OTProxyPluginMgr.h:68: error: expected â;â before 
âstaticâ
source/main.cc: In function âint printUsage()â:
source/main.cc:25: error: âstdoutâ was not declared in this scope
source/main.cc:30: error: âfprintfâ was not declared in this scope
source/main.cc: At global scope:
source/main.cc:35: error: âtsk_size_tâ has not been declared
source/main.cc:35: error: âtsk_size_tâ has not been declared
source/main.cc: In function âint parseArgument(const char*, const char**, 
int*, const char**, int*)â:
source/main.cc:38: error: âtsk_strnullORemptyâ was not declared in this scope
source/main.cc:40: error: âTSK_DEBUG_ERRORâ was not declared in this scope
source/main.cc:43: error: âtsk_nullâ was not declared in this scope
source/main.cc:45: error: âtsk_strlenâ was not declared in this scope
source/main.cc:48: error: âtsk_strindexOfâ was not declared in this scope
source/main.cc: In function âint parseArguments(int, char**)â:
source/main.cc:64: error: âtsk_size_tâ was not declared in this scope
source/main.cc:64: error: expected â;â before âname_sizeâ
source/main.cc:73: error: âname_sizeâ was not declared in this scope
source/main.cc:73: error: âvalue_sizeâ was not declared in this scope
source/main.cc:78: error: âname_sizeâ was not declared in this scope
source/main.cc:78: error: âtsk_strniequalsâ was not declared in this scope
source/main.cc:80: error: âvalue_sizeâ was not declared in this scope
source/main.cc:82: error: âstderrâ was not declared in this scope
source/main.cc:82: error: âfprintfâ was not declared in this scope
source/main.cc:86: error: âvalue_sizeâ was not declared in this scope
source/main.cc:86: error: âtsk_strndupâ was not declared in this scope
source/main.cc:95: error: âstdoutâ was not declared in this scope
source/main.cc:95: error: âfprintfâ was not declared in this scope
source/main.cc:100: error: âstderrâ was not declared in this scope
source/main.cc:100: error: âfprintfâ was not declared in this scope
source/main.cc: In function âint main(int, char**)â:
source/main.cc:122: error: âprintfâ was not declared in this scope
source/main.cc:132: error: âtsk_strnullORemptyâ was not declared in this scope
source/main.cc:143: error: âgetcharâ was not declared in this scope
source/main.cc:151: error: âtsk_thread_sleepâ was not declared in this scope
In file included from include/opentelepresence/OTCodec.h:11,
                 from include/OpenTelepresenceAPI.h:11,
                 from source/main.cc:7:
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTCodecEncodingResult*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTCodecEncodingResult*]â
include/opentelepresence/OTCodec.h:134:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTBridgeInfo*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTBridgeInfo*]â
include/opentelepresence/OTSessionInfo.h:34:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTRecorder*]â:
include/opentelepresence/OTObject.h:154:   instantiated from 
âOTObjectWrapper<OTObjectType>& OTObjectWrapper<OTObjectType>::operator=(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTRecorder*]â
include/opentelepresence/OTMixerMgr.h:30:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTSessionInfo*]â:
include/opentelepresence/OTObject.h:90:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(OTObjectType) [with 
OTObjectType = OTSessionInfo*]â
include/opentelepresence/OTWrap.h:64:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTDocStreamer*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTDocStreamer*]â
include/opentelepresence/OTWrap.h:65:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTEngine*]â:
include/opentelepresence/OTObject.h:154:   instantiated from 
âOTObjectWrapper<OTObjectType>& OTObjectWrapper<OTObjectType>::operator=(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTEngine*]â
include/opentelepresence/OTWrap.h:92:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTNetTransportCallback*]â:
include/opentelepresence/OTObject.h:154:   instantiated from 
âOTObjectWrapper<OTObjectType>& OTObjectWrapper<OTObjectType>::operator=(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTNetTransportCallback*]â
include/opentelepresence/nettransport/OTNetTransport.h:127:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTEngineInfo*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTEngineInfo*]â
include/opentelepresence/OTEngine.h:108:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTProxyPluginConsumerAudio*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = 
OTProxyPluginConsumerAudio*]â
include/opentelepresence/OTProxyPluginConsumerAudio.h:78:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTProxyPluginProducerAudio*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = 
OTProxyPluginProducerAudio*]â
include/opentelepresence/OTProxyPluginProducerAudio.h:61:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTFrameVideo*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = OTFrameVideo*]â
include/opentelepresence/OTProxyPluginConsumerVideo.h:40:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTProxyPluginConsumerVideo*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = 
OTProxyPluginConsumerVideo*]â
include/opentelepresence/OTProxyPluginConsumerVideo.h:71:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTProxyPluginProducerVideo*]â:
include/opentelepresence/OTObject.h:95:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(const 
OTObjectWrapper<OTObjectType>&) [with OTObjectType = 
OTProxyPluginProducerVideo*]â
include/opentelepresence/OTProxyPluginProducerVideo.h:68:   instantiated from 
here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
include/opentelepresence/OTObject.h: In member function âvoid 
OTObjectWrapper<OTObjectType>::wrapObject(OTObjectType) [with OTObjectType = 
OTProxyPlugin*]â:
include/opentelepresence/OTObject.h:90:   instantiated from 
âOTObjectWrapper<OTObjectType>::OTObjectWrapper(OTObjectType) [with 
OTObjectType = OTProxyPlugin*]â
include/opentelepresence/OTProxyPluginMgr.h:39:   instantiated from here
include/opentelepresence/OTObject.h:136: error: âTSK_DEBUG_ERRORâ was not 
declared in this scope
make[1]: *** [telepresence-main.o] Error 1
make[1]: Leaving directory `/telepresence'
make: *** [all] Error 2