Open GoogleCodeExporter opened 8 years ago
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;stun-user@doubango.org;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:10063@10.27.153.140:20060 SIP/2.0 Via: SIP/2.0/UDP 10.27.153.140:5060;branch=z9hG4bK2be33aa6;rport Max-Forwards: 70 From: "AAAAAA" <sip:100@10.27.153.140>;tag=as11210624 To: <sip:10063@10.27.153.140:20060> Contact: <sip:100@10.27.153.140:5060> Call-ID: 1ec62a15437e08c34b758cb764ca390b@10.27.153.140 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:100@10.27.153.140>;tag=as11210624 To: <sip:10063@10.27.153.140:20060> Call-ID: 1ec62a15437e08c34b758cb764ca390b@10.27.153.140 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:100@10.27.153.140>;tag=as11210624 To: <sip:10063@10.27.153.140:20060>;tag=998129733 Call-ID: 1ec62a15437e08c34b758cb764ca390b@10.27.153.140 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:10063@10.27.153.140:20060 SIP/2.0 Via: SIP/2.0/UDP 10.27.153.140:5060;branch=z9hG4bK2be33aa6;rport Max-Forwards: 70 From: "AAAAAA" <sip:100@10.27.153.140>;tag=as11210624 To: <sip:10063@10.27.153.140:20060>;tag=998129733 Contact: <sip:100@10.27.153.140:5060> Call-ID: 1ec62a15437e08c34b758cb764ca390b@10.27.153.140 CSeq: 102 ACK User-Agent: Asterisk PBX 12.0.0 Content-Length: 0 -----------------------------------
Original issue reported on code.google.com by lannt2...@gmail.com on 21 Feb 2014 at 12:04
lannt2...@gmail.com
Original issue reported on code.google.com by
lannt2...@gmail.comon 21 Feb 2014 at 12:04