Hi all, I’m trying these days to connect two clients using the demo offered by sipml5 using asterisk and webrtc2sip getaway. As for the configurations I have not encountered problems because I followed various guides on the web. I am able to authenticate the users with sipml5, I can also make the call but once instantiated this video does not start (you only see a black screen and the video bar below with my working web). Asterisk reports that: Ignoring video stream offer Because port number is zero. Now I think it is a codec problem, but I can not find a solution … can someone help me?
Show the logs and the sdp transaction between Asterisk & webrtc2sip
This is my sip.conf:
[general]
realm=doubango.org
udpbinaddr=0.0.0.0:5060
videosupport=yes
[1060]
type=friend
username=1060
host=dynamic
secret=1060
context=default
transport=udp
disallow=all
allow=ulaw
allow=alaw
allow=h264
allow=h263
allow=h263p
allow=g729
allow=vp8
nat=yes
[1061]
type=friend
username=1061
host=dynamic
secret=1061
transport=udp
disallow=all
allow=ulaw
allow=alaw
allow=h264
allow=263
allow=263p
allow=g729
allow=vp8
nat=yes
this is my extensions.conf:
[default]
exten => 1060,1,Dial(SIP/1060)
exten => 1060,2,Hangup()
exten => 1061,1,Dial(SIP/1061)
exten => 1061,2,Hangup()
Asterisk give me this:
– Registered SIP ‘1061’ at 192.168.1.102:10060
– Registered SIP ‘1060’ at 192.168.1.102:10060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Executing [1061@default:1] Dial(“SIP/1060-00000002”, “SIP/1061”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/1061
– SIP/1061-00000003 is ringing
[Jul 9 18:55:47] WARNING[3648][C-00000001]: chan_sip.c:10140 process_sdp: Ignoring video stream offer because port number is zero
– SIP/1061-00000003 answered SIP/1060-00000002
– Remotely bridging SIP/1060-00000002 and SIP/1061-00000003
[Jul 9 18:55:47] WARNING[3648][C-00000001]: chan_sip.c:10140 process_sdp: Ignoring video stream offer because port number is zero
webrtc2sip give me this:
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
INFO: === NICT terminated ===
INFO: *** NICT destroyed ***
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2150773640 cname:247bfda31ac34345367c62246cd6af1f
a=ssrc:2150773640 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2150773640 label:doubango@audio
m=video 0 RTP/AVP 99
a=rtcp-fb: nack pli
a=rtcp-fb: ccm fir
a=rtcp-fb: goog-remb
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.1.102} on port {55586} using fd {32} with type {3}…
*INFO: ICE context not active
*INFO:
RECV:INVITE sip:1060@192.168.1.102:10060;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK5c564c0e;rport
Max-Forwards: 70
From: sip:1061@doubango.org;tag=as1d6f6ce1
To: sip:1060@doubango.org;tag=35600799
Contact: sip:1061@192.168.1.102:5060
Call-ID: 67648292-6667-e8e2-9a9e-03039fc86e6f
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 2096295206 2096295207 IN IP4 192.168.1.102
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.1.102
t=0 0
m=audio 2188 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*INFO: Transport::run() - enter
*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 192.168.1.102:5060;rport=5060;received=192.168.1.102;branch=z9hG4bK5c564c0e
From: sip:1061@doubango.org;tag=as1d6f6ce1
To: sip:1060@doubango.org;tag=35600799
Call-ID: 67648292-6667-e8e2-9a9e-03039fc86e6f
CSeq: 102 INVITE
Content-Length: 0
*INFO: State machine: x0000_Connected_2_Connected_X_iINVITE
***ERROR: function: "tsdp_header_M_get_holdresume_att()"
file: "src/headers/tsdp_header_M.c"
line: "759"
MSG: Invalid parameter
*INFO: media type has changed
*INFO: is_ro_provisional_final_matching=0,
is_ro_media_lines_changed=1,
is_ro_network_info_changed=1,
is_ro_loopback_address=0,
is_media_type_changed=1
*INFO: stopped_to_reconf=true
*INFO: tmedia_session_mgr_stop()
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER – STOP
*INFO: ICE CTX::run – STOP
*INFO: Transport::run(RTP/RTCP Manager) - exit
*INFO: Stopping [RTP/RTCP Manager] server with IP {192.168.1.102} on port {39868} with type {3}…
*INFO: Stopped [RTP/RTCP Manager] server with IP {192.168.1.102} on port {39868}
*INFO: Socket to remove: fd=51, index=0, tail.count=2
*INFO: CloseSocket(fd=51)
*INFO: Socket to remove: fd=37, index=0, tail.count=1
*INFO: tdav_session_video_stop
*INFO: tdav_video_jb_stop()
*INFO: Video jitter buffer thread - EXIT
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER – STOP
*INFO: ICE CTX::run – STOP
*INFO: Transport::run(RTP/RTCP Manager) - exit
*INFO: Stopping [RTP/RTCP Manager] server with IP {192.168.1.102} on port {55586} with type {3}…
*INFO: Stopped [RTP/RTCP Manager] server with IP {192.168.1.102} on port {55586}
*INFO: Socket to remove: fd=55, index=0, tail.count=2
*INFO: CloseSocket(fd=55)
*INFO: Socket to remove: fd=32, index=0, tail.count=1
*INFO: *** tdav_session_video_t destroyed ***
*INFO: tdav_session_video_stop
*INFO: MPProxyPluginConsumerVideo object destroyed
*INFO: twrap_producer_proxy_video_dtor()
*INFO: ~ProxyVideoProducer
*INFO: *** RTP manager destroyed ***
*INFO: *** tdav_codec_vp8_dtor destroyed ***
*INFO: *** Video session destroyed ***
*INFO: max_bw_up=2147483647 kpbs, max_bw_down=2147483647 kpbs, congestion_ctrl_enabled=0, media_type=2
***ERROR: function: "_trtp_manager_ice_init()"
file: "src/trtp_manager.c"
line: "675"
MSG: ICE context not ready
***ERROR: function: "trtp_manager_start()"
file: "src/trtp_manager.c"
line: "1203"
MSG: _trtp_manager_ice_init() failed
*INFO: Audio denoiser to be opened(record_frame_size_samples=160, record_sampling_rate=8000, playback_frame_size_samples=160, playback_sampling_rate=8000)
warning: The VAD has been replaced by a hack pending a complete rewrite
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.1.102} on port {2188} using fd {38} with type {3}…
*INFO: State machine: tsip_transac_ist_Proceeding_2_Accepted_X_2xx
*INFO:
SEND: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5060;received=192.168.1.102;branch=z9hG4bK5c564c0e
From: sip:1061@doubango.org;tag=as1d6f6ce1
To: sip:1060@doubango.org;tag=35600799
Contact: sip:1060@192.168.1.102:10060;transport=udp
Call-ID: 67648292-6667-e8e2-9a9e-03039fc86e6f
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 486
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.1.102
s=-
c=IN IP4 192.168.1.102
t=0 0
m=audio 39868 RTP/AVP 0 8 101
c=IN IP4 192.168.1.102
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2174808041 cname:5d60c3d1b6d95db1ee0fb4c0b3924539
a=ssrc:2174808041 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2174808041 label:doubango@audio
*INFO:
RECV:ACK sip:1061@192.168.1.102:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1e7a85a2;rport
Max-Forwards: 70
From: sip:1060@192.168.1.102;tag=as20c3db1a
To: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;tag=1580385888
Contact: sip:1060@192.168.1.102:5060
Call-ID: 50897e295f1102831cc074d71b908f42@192.168.1.102:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
*INFO:
RECV:ACK sip:1060@192.168.1.102:10060;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK3bef0ba4;rport
Max-Forwards: 70
From: sip:1061@doubango.org;tag=as1d6f6ce1
To: sip:1060@doubango.org;tag=35600799
Contact: sip:1061@192.168.1.102:5060
Call-ID: 67648292-6667-e8e2-9a9e-03039fc86e6f
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
*INFO:
RECV:INVITE sip:1061@192.168.1.102:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK60b0a92a;rport
Max-Forwards: 70
From: sip:1060@192.168.1.102;tag=as20c3db1a
To: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;tag=1580385888
Contact: sip:1060@192.168.1.102:5060
Call-ID: 50897e295f1102831cc074d71b908f42@192.168.1.102:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 168710685 168710687 IN IP4 192.168.1.102
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.1.102
t=0 0
m=audio 39868 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
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 192.168.1.102:5060;rport=5060;received=192.168.1.102;branch=z9hG4bK60b0a92a
From: sip:1060@192.168.1.102;tag=as20c3db1a
To: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;tag=1580385888
Call-ID: 50897e295f1102831cc074d71b908f42@192.168.1.102:5060
CSeq: 104 INVITE
Content-Length: 0
*INFO: State machine: x0000_Connected_2_Connected_X_iINVITE
*INFO: 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
*INFO: State machine: tsip_transac_ist_Proceeding_2_Accepted_X_2xx
*INFO:
SEND: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;rport=5060;received=192.168.1.102;branch=z9hG4bK60b0a92a
From: sip:1060@192.168.1.102;tag=as20c3db1a
To: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;tag=1580385888
Contact: sip:1061@192.168.1.102:10060;transport=udp
Call-ID: 50897e295f1102831cc074d71b908f42@192.168.1.102:5060
CSeq: 104 INVITE
Content-Type: application/sdp
Content-Length: 549
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.1.102
s=-
c=IN IP4 192.168.1.102
t=0 0
m=audio 2188 RTP/AVP 0 8 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2150773640 cname:247bfda31ac34345367c62246cd6af1f
a=ssrc:2150773640 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2150773640 label:doubango@audio
m=video 0 RTP/AVP 99
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
*INFO:
RECV:ACK sip:1061@192.168.1.102:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK42180a82;rport
Max-Forwards: 70
From: sip:1060@192.168.1.102;tag=as20c3db1a
To: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;tag=1580385888
Contact: sip:1060@192.168.1.102:5060
Call-ID: 50897e295f1102831cc074d71b908f42@192.168.1.102:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
*INFO: Receiving SIP o/ WebSocket message: ACK sip:1061@192.168.1.102:10060;transport=ws;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKoIWBXzTBGux7g6pIRezQ;rport
From: "1060"sip:1060@doubango.org;tag=5zrzOp14VV6o8QyUPBk4
To: sip:1061@doubango.org;tag=1791194543
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2c271d8a-cf9b-2182-cf45-28249e396403
CSeq: 11580 ACK
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
*INFO: Mapped address different than local connection address…probably symetric NAT: 151.73.36.106#192.168.1.102 or 12270#45452
*INFO: Mapped address different than local connection address…probably symetric NAT: 151.73.36.106#192.168.1.102 or 12271#45453
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 45452#12270
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 45453#12271
*INFO: Mapped address different than local connection address…probably symetric NAT: 151.73.36.106#192.168.1.102 or 12268#52738
*INFO: Mapped address different than local connection address…probably symetric NAT: 151.73.36.106#192.168.1.102 or 12269#52739
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 52738#12268
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 52739#12269
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 45452#12270
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 45453#12271
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 52738#12268
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 52739#12269
*INFO: Mapped address different than local connection address…probably symetric NAT: 192.168.1.102#151.73.36.106 or 52739#12269
*INFO: State machine: No matching state found.
*INFO: State machine: ICE_ConnChecking_2_ConnCheckingCompleted_X_Success
*INFO: ICE callback: ConnCheck succeed
*INFO: State machine: No matching state found.
*INFO: State machine: ICE_ConnChecking_2_ConnCheckingCompleted_X_Success
*INFO: ICE callback: ConnCheck succeed
*INFO: max_bw_up=2147483647 kpbs, max_bw_down=2147483647 kpbs, congestion_ctrl_enabled=0, media_type=2
*INFO: SO_RCVBUF = 65535, SO_SNDBUF = 65535
*INFO: rtcp.remote_ip=192.168.1.101, rtcp.remote_port=51310, rtcp.local_fd=25
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: srtp_use_different_keys=false
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=30
*INFO: Socket added[RTP/RTCP Manager]: fd=30, tail.count=1
*INFO: master fd=25
*INFO: Socket added[RTP/RTCP Manager]: fd=25, tail.count=2
*INFO: Audio denoiser to be opened(record_frame_size_samples=960, record_sampling_rate=48000, playback_frame_size_samples=960, playback_sampling_rate=48000)
*INFO: Transport::run() - enter
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.1.102} on port {45452} using fd {25} with type {3}…
warning: The VAD has been replaced by a hack pending a complete rewrite
*INFO: [VP8] target_bitrate=630 kbps
*INFO: Video jitter buffer thread - ENTER
*INFO: max_bw_up=2147483647 kpbs, max_bw_down=2147483647 kpbs, congestion_ctrl_enabled=0, media_type=4
*INFO: SO_RCVBUF = 65535, SO_SNDBUF = 65535
*INFO: rtcp.remote_ip=192.168.1.101, rtcp.remote_port=51310, rtcp.local_fd=31
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: srtp_use_different_keys=false
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=35
*INFO: Socket added[RTP/RTCP Manager]: fd=35, tail.count=1
*INFO: master fd=31
*INFO: Socket added[RTP/RTCP Manager]: fd=31, tail.count=2
*INFO: Transport::run() - enter
*INFO: Starting [RTP/RTCP Manager] server with IP {192.168.1.102} on port {52738} using fd {31} with type {3}…
*INFO: Decoded VP8 IDR
*INFO: IDR frame decoded
*INFO: State machine: No matching state found.
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: === ICT terminated ===
*INFO: *** ICT destroyed ***
**WARN: function: "tsip_transac_fsm_act()"
file: "src/transactions/tsip_transac.c"
line: "265"
MSG: Invalid parameter.
*INFO: State machine: tsip_transac_ist_Accepted_2_Terminated_timerL
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: === ICT terminated ===
*INFO: *** ICT destroyed ***
**WARN: function: "tsip_transac_fsm_act()"
file: "src/transactions/tsip_transac.c"
line: "265"
MSG: Invalid parameter.
*INFO: State machine: tsip_transac_ist_Accepted_2_Terminated_timerL
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: State machine: tsip_transac_ist_Accepted_2_Terminated_timerL
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: State machine: tsip_transac_ist_Accepted_2_Terminated_timerL
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: State machine: tsip_transac_ist_Accepted_2_Terminated_timerL
*INFO: === IST terminated ===
*INFO: *** IST destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: === ICT terminated ===
*INFO: *** ICT destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKF3fapL5cDq3kF8dxfPx9vuJfnCnd9Dxu;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25390 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“29380b1b”,uri=“sip:doubango.org”,response=“2dfbc73942ec72d5ebe318c403ab4319”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKF3fapL5cDq3kF8dxfPx9vuJfnCnd9Dxu;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25390 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“29380b1b”,uri=“sip:doubango.org”,response=“2dfbc73942ec72d5ebe318c403ab4319”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKF3fapL5cDq3kF8dxfPx9vuJfnCnd9Dxu;ws-hacked=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKF3fapL5cDq3kF8dxfPx9vuJfnCnd9Dxu;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKF3fapL5cDq3kF8dxfPx9vuJfnCnd9Dxu;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org;tag=as450f42e9
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25390 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="3ce4d0cd"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKithzIoG57XG9TTAEIndkEITt9IkFMRRO;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25391 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3ce4d0cd”,uri=“sip:doubango.org”,response=“9857cafb8ef9de19d0d31cb2abc991d9”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKithzIoG57XG9TTAEIndkEITt9IkFMRRO;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25391 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3ce4d0cd”,uri=“sip:doubango.org”,response=“9857cafb8ef9de19d0d31cb2abc991d9”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKithzIoG57XG9TTAEIndkEITt9IkFMRRO;ws-hacked=WS
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKithzIoG57XG9TTAEIndkEITt9IkFMRRO;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKithzIoG57XG9TTAEIndkEITt9IkFMRRO;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org;tag=as450f42e9
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25391 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200
Date: Tue, 09 Jul 2013 16:57:02 GMT
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb5LlPGrTRieKxnIKVTrvd0whw2xq8pXH;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54685 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“0e117cb7”,uri=“sip:doubango.org”,response=“a9cf589cc2b743694ea4b1a57bea4251”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKb5LlPGrTRieKxnIKVTrvd0whw2xq8pXH;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54685 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“0e117cb7”,uri=“sip:doubango.org”,response=“a9cf589cc2b743694ea4b1a57bea4251”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKb5LlPGrTRieKxnIKVTrvd0whw2xq8pXH;ws-hacked=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKb5LlPGrTRieKxnIKVTrvd0whw2xq8pXH;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKb5LlPGrTRieKxnIKVTrvd0whw2xq8pXH;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org;tag=as4326354c
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54685 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="7c2e7780"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTmh1RiAGxWbqTVbWq9WkM2ImENd5zdxL;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54686 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“7c2e7780”,uri=“sip:doubango.org”,response=“c8c181213ebde526644ca25127eb7a1c”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKTmh1RiAGxWbqTVbWq9WkM2ImENd5zdxL;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54686 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“7c2e7780”,uri=“sip:doubango.org”,response=“c8c181213ebde526644ca25127eb7a1c”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKTmh1RiAGxWbqTVbWq9WkM2ImENd5zdxL;ws-hacked=WS
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKTmh1RiAGxWbqTVbWq9WkM2ImENd5zdxL;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKTmh1RiAGxWbqTVbWq9WkM2ImENd5zdxL;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org;tag=as4326354c
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54686 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200
Date: Tue, 09 Jul 2013 16:57:27 GMT
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKH9KcTIfZVqgSLMPKoIpANHXLdefxtcKs;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25392 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3ce4d0cd”,uri=“sip:doubango.org”,response=“9857cafb8ef9de19d0d31cb2abc991d9”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKH9KcTIfZVqgSLMPKoIpANHXLdefxtcKs;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25392 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3ce4d0cd”,uri=“sip:doubango.org”,response=“9857cafb8ef9de19d0d31cb2abc991d9”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKH9KcTIfZVqgSLMPKoIpANHXLdefxtcKs;ws-hacked=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKH9KcTIfZVqgSLMPKoIpANHXLdefxtcKs;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKH9KcTIfZVqgSLMPKoIpANHXLdefxtcKs;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org;tag=as128793ff
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25392 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="3601abb5"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKskFT9OzMGg3Mlaus1cB8t3xwfIZOkO4g;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25393 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3601abb5”,uri=“sip:doubango.org”,response=“7d87e2bc4d66234315718f8ce6066633”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKskFT9OzMGg3Mlaus1cB8t3xwfIZOkO4g;rport
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25393 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“3601abb5”,uri=“sip:doubango.org”,response=“7d87e2bc4d66234315718f8ce6066633”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKskFT9OzMGg3Mlaus1cB8t3xwfIZOkO4g;ws-hacked=WS
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKskFT9OzMGg3Mlaus1cB8t3xwfIZOkO4g;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64479;rport;branch=z9hG4bKskFT9OzMGg3Mlaus1cB8t3xwfIZOkO4g;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=Ls20FpChRCDvyKQThkFg
To: "1061"sip:1061@doubango.org;tag=as128793ff
Call-ID: 8f385023-68f1-ddd6-6e3c-d5ebfff6784f
CSeq: 25393 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1061@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64479;ws-src-proto=ws;expires=200
Date: Tue, 09 Jul 2013 16:58:42 GMT
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: State machine: tsip_transac_nict_Completed_2_Terminated_X_timerK
*INFO: === NICT terminated ===
*INFO: *** NICT destroyed ***
*INFO: According to rtp-timestamps …FPS = 6 (clipped to 10) tail_max=20, latency_max=10
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKoX9KKdJBto8b76d5bzb5oWESADR7Fipb;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54687 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“7c2e7780”,uri=“sip:doubango.org”,response=“c8c181213ebde526644ca25127eb7a1c”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKoX9KKdJBto8b76d5bzb5oWESADR7Fipb;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54687 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“7c2e7780”,uri=“sip:doubango.org”,response=“c8c181213ebde526644ca25127eb7a1c”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKoX9KKdJBto8b76d5bzb5oWESADR7Fipb;ws-hacked=WS
*INFO:
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKoX9KKdJBto8b76d5bzb5oWESADR7Fipb;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKoX9KKdJBto8b76d5bzb5oWESADR7Fipb;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org;tag=as57e69538
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54687 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="527f673d"
Content-Length: 0
*INFO: State machine: tsip_transac_nict_Trying_2_Completed_X_200_to_699
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGrJUagEePVFaeQ1o6Tg2Ti7WmX9TWhIU;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54688 REGISTER
Content-Length: 0
Route: sip:192.168.1.102:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“527f673d”,uri=“sip:doubango.org”,response=“7c3990796d70bb98e9580f56a10eaf56”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO:
SEND: REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKGrJUagEePVFaeQ1o6Tg2Ti7WmX9TWhIU;rport
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54688 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“527f673d”,uri=“sip:doubango.org”,response=“7c3990796d70bb98e9580f56a10eaf56”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKGrJUagEePVFaeQ1o6Tg2Ti7WmX9TWhIU;ws-hacked=WS
*INFO:
RECV:SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:10060;branch=z9hG4bKGrJUagEePVFaeQ1o6Tg2Ti7WmX9TWhIU;received=192.168.1.102;rport=10060
Via: SIP/2.0/TCP 192.168.1.101:64481;rport;branch=z9hG4bKGrJUagEePVFaeQ1o6Tg2Ti7WmX9TWhIU;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=pCmum6eHrAzsdF9IPXt5
To: "1060"sip:1060@doubango.org;tag=as57e69538
Call-ID: 2e62b0e6-d343-e67e-4b0c-00b1d4e08499
CSeq: 54688 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1060@192.168.1.102:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.1.101;ws-src-port=64481;ws-src-proto=ws;expires=200
Date: Tue, 09 Jul 2013 16:59:07 GMT
Content-Length: 0
Try adding the videosupport to the peer too, and enable the sip debug on asterisk side. The webrtc2sip log shown port 0 for video so attach the js log drone chrome too.
<------------>
Scheduling destruction of SIP dialog ‘979b9832-3497-bdf8-a89f-da2c1244e4c4’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:192.168.100.208:10060 —>
INVITE sip:1060@doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bK-1578546353;rport
From: sip:1061@doubango.org;tag=75890588
To: sip:1060@doubango.org
Contact: sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 2116950504 INVITE
Content-Type: application/sdp
Content-Length: 2111
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“424e7dfe”,uri="sip:1060@doubango.org",response=“316e7f7aa2033d2d17dfc12ae5e27309”,algorithm=MD5
User-Agent: webrtc2sip Media Server 2.5.1
v=0
o=doubango 1983 678901 IN IP4 192.168.100.208
s=-
c=IN IP4 192.168.100.208
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 56316 RTP/AVP 111 8 0 101
c=IN IP4 192.168.100.208
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:0AGC34HYxPLCiqEbVQOODtXxoQ/nVLzh2qivSHtW
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:lzBHIsaPW+JReuT+IHtG8pI3+qXsWrS4sT10WWLu
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2999846880 cname:33c5055ae4eeac9017efa65cc800052c
a=ssrc:2999846880 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2999846880 label:doubango@audio
a=ice-ufrag:epUOsK4ccPXlF1v
a=ice-pwd:sdk0mGozxmKxOSXFUzu6h
a=candidate:sbb9ALyH5 1 udp 2130706431 192.168.100.208 56316 typ host
a=candidate:sbb9ALyH5 2 udp 2130706430 192.168.100.208 56317 typ host
m=video 48390 RTP/AVP 100 34 103
c=IN IP4 192.168.100.208
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtpmap:100 VP8/90000
a=imageattr:100 recv [x=[128:16:640],y=[96:16:480]] send [x=[128:16:640],y=[96:16:480]]
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:103 H263-1998/90000
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UUoSjRDuSVJhAn3sDhV6X/e0hHdm0qDiFlQvQi9o
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:42jkYWX95ozoGnEeF5tRtuKGCgUgD+nda0NoyjhE
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:2475278685 cname:a8f24c7df244f42fa5da4868f932a898
a=ssrc:2475278685 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2475278685 label:doubango@video
a=ice-ufrag:9OkP3TRncY09Fx8
a=ice-pwd:AMbW75azCMflno8TbzXdN
a=candidate:nnyMDgrjB 1 udp 2130706431 192.168.100.208 48390 typ host
a=candidate:nnyMDgrjB 2 udp 2130706430 192.168.100.208 48391 typ host
<------------->
— (12 headers 57 lines) —
Sending to 192.168.100.208:10060 (NAT)
Using INVITE request as basis request - 979b9832-3497-bdf8-a89f-da2c1244e4c4
Found peer ‘1061’ for ‘1061’ from 192.168.100.208:10060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found unknown media description format opus for ID 111
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 100
Found RTP video format 34
Found RTP video format 103
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 103
Capabilities: us - (ulaw|alaw|g729|h263|h263p|h264), peer - audio=(ulaw|alaw)/video=(h263|h263p|silk12)/text=(nothing), combined - (ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.208:56316
Peer video RTP is at port 192.168.100.208:48390
Looking for 1060 in default (domain doubango.org)
list_route: hop: sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bK-1578546353;received=192.168.100.208;rport=10060
From: sip:1061@doubango.org;tag=75890588
To: sip:1060@doubango.org
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 2116950504 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1060@192.168.100.208:5060
Content-Length: 0
<------------>
– Executing [1060@default:1] Dial(“SIP/1061-00000006”, “SIP/1060”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 13096
Video is at 192.168.100.208:18470
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.100.208:10060:
INVITE sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK0419ebf8;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Date: Wed, 10 Jul 2013 10:10:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 698
v=0
o=root 104413716 104413716 IN IP4 192.168.100.208
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.100.208
b=CT:384
t=0 0
m=audio 13096 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18470 RTP/AVP 99 34 98
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK0419ebf8
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
– Called SIP/1060
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK0419ebf8
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1060@192.168.100.208:10060;transport=udp
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
— (9 headers 0 lines) —
list_route: hop: sip:1060@192.168.100.208:10060;transport=udp
– SIP/1060-00000007 is ringing
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bK-1578546353;received=192.168.100.208;rport=10060
From: sip:1061@doubango.org;tag=75890588
To: sip:1060@doubango.org;tag=as7f6248fb
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 2116950504 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1060@192.168.100.208:5060
Content-Length: 0
<------------>
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK0419ebf8
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1060@192.168.100.208:10060;transport=udp
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 851
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678901 IN IP4 192.168.100.208
s=-
c=IN IP4 192.168.100.208
t=0 0
m=audio 7604 RTP/AVP 0 8 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=sendrecv
a=ssrc:2979485199 cname:abb0d39c9a5646a7cbcd559a216818bf
a=ssrc:2979485199 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2979485199 label:doubango@audio
m=video 39180 RTP/AVP 34 98
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=2;QCIF=2;SQCIF=2
a=sendrecv
a=ssrc:3023293270 cname:380b7e0f911f597df86184deeaeef27a
a=ssrc:3023293270 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3023293270 label:doubango@video
<------------->
— (10 headers 30 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found RTP video format 98
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 98
Capabilities: us - (ulaw|alaw|g729|h263|h263p|h264), peer - audio=(ulaw|alaw)/video=(h263|h263p)/text=(nothing), combined - (ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.208:7604
Peer video RTP is at port 192.168.100.208:39180
list_route: hop: sip:1060@192.168.100.208:10060;transport=udp
set_destination: Parsing sip:1060@192.168.100.208:10060;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Transmitting (NAT) to 192.168.100.208:10060:
ACK sip:1060@192.168.100.208:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK2d4931cc;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
-- SIP/1060-00000007 answered SIP/1061-00000006
Audio is at 18822
Video is at 192.168.100.208:15074
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bK-1578546353;received=192.168.100.208;rport=10060
From: sip:1061@doubango.org;tag=75890588
To: sip:1060@doubango.org;tag=as7f6248fb
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 2116950504 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1060@192.168.100.208:5060
Content-Type: application/sdp
Content-Length: 535
v=0
o=root 1116477755 1116477755 IN IP4 192.168.100.208
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.100.208
b=CT:384
t=0 0
m=audio 18822 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 15074 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:103 h263-1998/90000
a=fmtp:103 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<------------>
– Remotely bridging SIP/1061-00000006 and SIP/1060-00000007
set_destination: Parsing sip:1060@192.168.100.208:10060;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Audio is at 13096
Video is at 192.168.100.208:48390
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.100.208:10060:
INVITE sip:1060@192.168.100.208:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK27a9ee6a;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 530
v=0
o=root 104413716 104413717 IN IP4 192.168.100.208
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.100.208
b=CT:384
t=0 0
m=audio 56316 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 48390 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<— SIP read from UDP:192.168.100.208:10060 —>
ACK sip:1060@192.168.100.208:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bK-1316707973;rport
From: sip:1061@doubango.org;tag=75890588
To: sip:1060@doubango.org;tag=as7f6248fb
Contact: sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 2116950504 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“424e7dfe”,uri=“sip:1060@192.168.100.208:5060”,response=“a229d9311eab872ab91af2ea6a7418c2”,algorithm=MD5
User-Agent: webrtc2sip Media Server 2.5.1
<------------->
— (11 headers 0 lines) —
set_destination: Parsing sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Audio is at 18822
Video is at 192.168.100.208:39180
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.100.208:10060:
INVITE sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK4bb69a91;rport
Max-Forwards: 70
From: sip:1060@doubango.org;tag=as7f6248fb
To: sip:1061@doubango.org;tag=75890588
Contact: sip:1060@192.168.100.208:5060
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 534
v=0
o=root 1116477755 1116477756 IN IP4 192.168.100.208
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.100.208
b=CT:384
t=0 0
m=audio 7604 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 39180 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:103 h263-1998/90000
a=fmtp:103 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK27a9ee6a
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 103 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK27a9ee6a
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1060@192.168.100.208:10060;transport=udp
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 103 INVITE
Content-Type: application/sdp
Content-Length: 851
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.100.208
s=-
c=IN IP4 192.168.100.208
t=0 0
m=audio 7604 RTP/AVP 0 8 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2979485199 cname:abb0d39c9a5646a7cbcd559a216818bf
a=ssrc:2979485199 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2979485199 label:doubango@audio
m=video 39180 RTP/AVP 34 98
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=sendrecv
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=2;QCIF=2;SQCIF=2
a=ssrc:3023293270 cname:380b7e0f911f597df86184deeaeef27a
a=ssrc:3023293270 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3023293270 label:doubango@video
<------------->
— (10 headers 30 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found RTP video format 98
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 98
Capabilities: us - (ulaw|alaw|g729|h263|h263p|h264), peer - audio=(ulaw|alaw)/video=(h263|h263p)/text=(nothing), combined - (ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.208:7604
Peer video RTP is at port 192.168.100.208:39180
set_destination: Parsing sip:1060@192.168.100.208:10060;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Transmitting (NAT) to 192.168.100.208:10060:
ACK sip:1060@192.168.100.208:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK3f5af620;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK4bb69a91
From: sip:1060@doubango.org;tag=as7f6248fb
To: sip:1061@doubango.org;tag=75890588
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK4bb69a91
From: sip:1060@doubango.org;tag=as7f6248fb
To: sip:1061@doubango.org;tag=75890588
Contact: sip:1061@192.168.100.208:10060;transport=udp
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 907
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.100.208
s=-
c=IN IP4 192.168.100.208
t=0 0
m=audio 56316 RTP/AVP 0 8 101
c=IN IP4 192.168.100.208
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2999846880 cname:33c5055ae4eeac9017efa65cc800052c
a=ssrc:2999846880 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2999846880 label:doubango@audio
m=video 48390 RTP/AVP 34 103
c=IN IP4 192.168.100.208
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=sendrecv
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:103 H263-1998/90000
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2
a=ssrc:2475278685 cname:a8f24c7df244f42fa5da4868f932a898
a=ssrc:2475278685 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2475278685 label:doubango@video
<------------->
— (10 headers 32 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found RTP video format 103
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 103
Capabilities: us - (ulaw|alaw|g729|h263|h263p|h264), peer - audio=(ulaw|alaw)/video=(h263|h263p)/text=(nothing), combined - (ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.208:56316
Peer video RTP is at port 192.168.100.208:48390
set_destination: Parsing sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Transmitting (NAT) to 192.168.100.208:10060:
ACK sip:1061@192.168.100.208:10060;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK6510a71f;rport
Max-Forwards: 70
From: sip:1060@doubango.org;tag=as7f6248fb
To: sip:1061@doubango.org;tag=75890588
Contact: sip:1060@192.168.100.208:5060
Call-ID: 979b9832-3497-bdf8-a89f-da2c1244e4c4
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
set_destination: Parsing sip:1060@192.168.100.208:10060;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Audio is at 13096
Video is at 192.168.100.208:48390
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.100.208:10060:
INVITE sip:1060@192.168.100.208:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK4b7c159b;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 530
v=0
o=root 104413716 104413718 IN IP4 192.168.100.208
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.100.208
b=CT:384
t=0 0
m=audio 56316 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 48390 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 SQCIF=2;QCIF=2;CIF=2;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK4b7c159b
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 104 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:5060;rport=5060;received=192.168.100.208;branch=z9hG4bK4b7c159b
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1060@192.168.100.208:10060;transport=udp
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 104 INVITE
Content-Type: application/sdp
Content-Length: 851
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
v=0
o=doubango 1983 678902 IN IP4 192.168.100.208
s=-
c=IN IP4 192.168.100.208
t=0 0
m=audio 7604 RTP/AVP 0 8 101
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ssrc:2979485199 cname:abb0d39c9a5646a7cbcd559a216818bf
a=ssrc:2979485199 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2979485199 label:doubango@audio
m=video 39180 RTP/AVP 34 98
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=sendrecv
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2;QCIF=2;SQCIF=2
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=2;QCIF=2;SQCIF=2
a=ssrc:3023293270 cname:380b7e0f911f597df86184deeaeef27a
a=ssrc:3023293270 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:3023293270 label:doubango@video
<------------->
— (10 headers 30 lines) —
set_destination: Parsing sip:1060@192.168.100.208:10060;transport=udp for address/port to send to
set_destination: set destination to 192.168.100.208:10060
Transmitting (NAT) to 192.168.100.208:10060:
ACK sip:1060@192.168.100.208:10060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:5060;branch=z9hG4bK193ef39d;rport
Max-Forwards: 70
From: sip:1061@192.168.100.208;tag=as5452c956
To: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;tag=747254416
Contact: sip:1061@192.168.100.208:5060
Call-ID: 24a1393a678e8abc0824ffa95fb31ffb@192.168.100.208:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.4.0
Content-Length: 0
<— SIP read from UDP:192.168.100.208:10060 —>
REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKk8SqOso1KB33G6L8Ja0l9JTL4SiS5WAH;rport
From: "1061"sip:1061@doubango.org;tag=RkalTrUKrROK8N9uOHiS
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 87b72d3b-8b74-cbf8-1c96-42491a1f912d
CSeq: 5705 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“185b011d”,uri=“sip:doubango.org”,response=“c6466a67736803e2542ccdd44b6394a9”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.100.205:49463;rport;branch=z9hG4bKk8SqOso1KB33G6L8Ja0l9JTL4SiS5WAH;ws-hacked=WS
<------------->
— (13 headers 0 lines) —
Sending to 192.168.100.208:10060 (no NAT)
Sending to 192.168.100.208:10060 (no NAT)
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKk8SqOso1KB33G6L8Ja0l9JTL4SiS5WAH;received=192.168.100.208;rport=10060
Via: SIP/2.0/TCP 192.168.100.205:49463;rport;branch=z9hG4bKk8SqOso1KB33G6L8Ja0l9JTL4SiS5WAH;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=RkalTrUKrROK8N9uOHiS
To: "1061"sip:1061@doubango.org;tag=as6e66236f
Call-ID: 87b72d3b-8b74-cbf8-1c96-42491a1f912d
CSeq: 5705 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="7ab40168"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘87b72d3b-8b74-cbf8-1c96-42491a1f912d’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:192.168.100.208:10060 —>
REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKN3ODno36fBa6Tthu537dmq3VcTKJgfy0;rport
From: "1061"sip:1061@doubango.org;tag=RkalTrUKrROK8N9uOHiS
To: "1061"sip:1061@doubango.org
Contact: "1061"sip:1061@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 87b72d3b-8b74-cbf8-1c96-42491a1f912d
CSeq: 5706 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1061”,realm=“doubango.org”,nonce=“7ab40168”,uri=“sip:doubango.org”,response=“5de49bdcf7b837318d005c6bc768c73b”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.100.205:49463;rport;branch=z9hG4bKN3ODno36fBa6Tthu537dmq3VcTKJgfy0;ws-hacked=WS
<------------->
— (13 headers 0 lines) —
Sending to 192.168.100.208:10060 (no NAT)
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKN3ODno36fBa6Tthu537dmq3VcTKJgfy0;received=192.168.100.208;rport=10060
Via: SIP/2.0/TCP 192.168.100.205:49463;rport;branch=z9hG4bKN3ODno36fBa6Tthu537dmq3VcTKJgfy0;ws-hacked=WS
From: "1061"sip:1061@doubango.org;tag=RkalTrUKrROK8N9uOHiS
To: "1061"sip:1061@doubango.org;tag=as6e66236f
Call-ID: 87b72d3b-8b74-cbf8-1c96-42491a1f912d
CSeq: 5706 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1061@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.205;ws-src-port=49463;ws-src-proto=ws;expires=200
Date: Wed, 10 Jul 2013 10:10:37 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘87b72d3b-8b74-cbf8-1c96-42491a1f912d’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:192.168.100.208:10060 —>
REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKpiZ0GwaBImbO9ihpCTSuOB1rbUG1RT6e;rport
From: "1060"sip:1060@doubango.org;tag=7IedLwahwv2zogwaJr40
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e301ad69-c937-f4c2-cfa8-6bf17e14f535
CSeq: 27360 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“289026a7”,uri=“sip:doubango.org”,response=“2ccc2df6f937285fe01e55298d51c61a”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.100.207:50848;rport;branch=z9hG4bKpiZ0GwaBImbO9ihpCTSuOB1rbUG1RT6e;ws-hacked=WS
<------------->
— (13 headers 0 lines) —
Sending to 192.168.100.208:10060 (no NAT)
Sending to 192.168.100.208:10060 (no NAT)
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKpiZ0GwaBImbO9ihpCTSuOB1rbUG1RT6e;received=192.168.100.208;rport=10060
Via: SIP/2.0/TCP 192.168.100.207:50848;rport;branch=z9hG4bKpiZ0GwaBImbO9ihpCTSuOB1rbUG1RT6e;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=7IedLwahwv2zogwaJr40
To: "1060"sip:1060@doubango.org;tag=as1a2d1227
Call-ID: e301ad69-c937-f4c2-cfa8-6bf17e14f535
CSeq: 27360 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“doubango.org”, nonce="3086e5cb"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘e301ad69-c937-f4c2-cfa8-6bf17e14f535’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:192.168.100.208:10060 —>
REGISTER sip:doubango.org SIP/2.0
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKOrLkH7yBTTlmXeHU4O079cOe0yqQdNk2;rport
From: "1060"sip:1060@doubango.org;tag=7IedLwahwv2zogwaJr40
To: "1060"sip:1060@doubango.org
Contact: "1060"sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: e301ad69-c937-f4c2-cfa8-6bf17e14f535
CSeq: 27361 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“doubango.org”,nonce=“3086e5cb”,uri=“sip:doubango.org”,response=“0c9ca49341bd324fffce623f6b5dfd45”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24
Organization: Doubango Telecom
Via: SIP/2.0/TCP 192.168.100.207:50848;rport;branch=z9hG4bKOrLkH7yBTTlmXeHU4O079cOe0yqQdNk2;ws-hacked=WS
<------------->
— (13 headers 0 lines) —
Sending to 192.168.100.208:10060 (no NAT)
<— Transmitting (NAT) to 192.168.100.208:10060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.208:10060;branch=z9hG4bKOrLkH7yBTTlmXeHU4O079cOe0yqQdNk2;received=192.168.100.208;rport=10060
Via: SIP/2.0/TCP 192.168.100.207:50848;rport;branch=z9hG4bKOrLkH7yBTTlmXeHU4O079cOe0yqQdNk2;ws-hacked=WS
From: "1060"sip:1060@doubango.org;tag=7IedLwahwv2zogwaJr40
To: "1060"sip:1060@doubango.org;tag=as1a2d1227
Call-ID: e301ad69-c937-f4c2-cfa8-6bf17e14f535
CSeq: 27361 REGISTER
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:1060@192.168.100.208:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=192.168.100.207;ws-src-port=50848;ws-src-proto=ws;expires=200
Date: Wed, 10 Jul 2013 10:10:47 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘e301ad69-c937-f4c2-cfa8-6bf17e14f535’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘87b72d3b-8b74-cbf8-1c96-42491a1f912d’ Method: REGISTER
In your last debug I can see the port of the video negotiation 48390, so you cannot see it? Ehat about Chrome, does it detect your camera? Which OS are you using. alos share the JS log.