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Error when call from tandberg 990 mxp in AS mode #25

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GoogleCodeExporter opened this issue Sep 6, 2015 · 0 comments
Open

Error when call from tandberg 990 mxp in AS mode #25

GoogleCodeExporter opened this issue Sep 6, 2015 · 0 comments
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auto-migrated Priority-Medium Type-Defect

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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

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

Original issue reported on code.google.com by [email protected] on 21 Feb 2014 at 12:04
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