A call with a SIP endpoint over TCP results in a call disconnection when the TCP SIP connection closes.

What steps will reproduce the problem?
1. Setup a trunk/PBX that only does SIP over TCP
2. Start up a WebRTC call (I tested this with an inbound call to the browser)
3. Call will drop in roughly 40-50 seconds which is the result of what I assume 
is one of the sides closing up the SIP TCP connection from inactivity. If 
either side sends a message it needs to reopen the TCP connection.

My "PBX" that is connecting to webrtc2sip is actually a node.js process using 
https://github.com/kirm/sip.js to interact and proxy requests from outside SIP 
trunks. Not sure if this information is relevant.

If I don't specify transport=tcp in the URI when connecting to webrtc2sip 
(resulting in a UDP connection) the call remains stable. This is my current 
workaround for this issue.

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

Call should not drop.

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

webrtc2sip@rev.114
doubango@rev.998

on Ubuntu 12.04 64bit.

NOTE: I am not using sipml5 for this call. I assume the behaviour will be the 
same, but my test environment runs with a custom tool connected to webrtc2sip 
via WebSockets.

Please provide server logs with DEBUG level equal to INFO

root@ubuntu:/usr/local/etc/webrtc2sip# webrtc2sip --config=./config.xml 
*******************************************************************
Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>
PRODUCT: webrtc2sip
HOME PAGE: http://webrtc2sip.org
LICENCE: GPLv3 or proprietary
VERSION: 2.5.1
'quit' to quit the application.
*******************************************************************

SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
*INFO: transport = tcp://*:5060
*INFO: transport = ws://*:10060
*INFO: transport = udp://*:5060
*INFO: enable-rtp-symetric = yes
*INFO: enable-100rel = no
*INFO: enable-media-coder = yes
*INFO: enable-videojb = no
*INFO: video-size-pref = vga
*INFO: rtp-buffsize = 65535
*INFO: avpf-tail-length = [100-400]
*INFO: srtp-mode = none
*INFO: srtp-type = sdes;dtls
*INFO: dtmf-type = rfc4733
*INFO: codecs = pcma;pcmu;vp8;h264-bp;h264-mp;h263;h263+
*INFO: UnRegister codec: PCMA, G.711a codec (native)
*INFO: UnRegister codec: PCMU, G.711u codec (native)
*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: UnRegister codec: H263, H263-1996 codec (FFmpeg)
*INFO: UnRegister codec: H263-1998, H263-1998 codec (FFmpeg)
*INFO: codec-opus-maxrates = 48000;48000
*INFO: enable-icestun = no
*INFO: nameserver = 8.8.8.8
*INFO: Stack running in SERVER mode
*INFO: tsk_timer_manager_start
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: SIP STACK::run -- START
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=6
*INFO: Socket added[SIP transport]: fd=6, tail.count=1
*INFO: master fd=3
*INFO: Socket added[SIP transport]: fd=3, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: Transport::run() - enter
*INFO: pipeR fd=8
*INFO: Socket added[SIP transport]: fd=8, tail.count=1
*INFO: master fd=4
*INFO: Socket added[SIP transport]: fd=4, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=10
*INFO: Socket added[SIP transport]: fd=10, tail.count=1
*INFO: master fd=5
*INFO: Socket added[SIP transport]: fd=5, tail.count=2
*INFO: Transport::run() - enter
*INFO: SIP STACK -- START
*INFO: Starting [SIP transport] server with IP {10.0.1.17} on port {5060} using 
fd {3} with type {2}...
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {10.0.1.17} on port {5060} using 
fd {4} with type {8}...
*INFO: Starting [SIP transport] server with IP {10.0.1.17} on port {10060} 
using fd {5} with type {64}...
*INFO: ioctlt(5), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=12)
*INFO: Socket added[SIP transport]: fd=12, tail.count=3
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 12
*INFO: WebSocket handshake message: GET / HTTP/1.1
Connection: Upgrade
Upgrade: websocket
Host: 10.0.1.17:10060
Origin: http://www.laboosh.com
Sec-WebSocket-Version: 13
Sec-WebSocket-Key: MTMtMTM4MDY3MDM0NzE1Mw==
Sec-WebSocket-Protocol: sip

*INFO: WebSocket Peer accepted/connected with fd = 12
*INFO: *** Stream Peer destroyed ***
*INFO: Receiving SIP o/ WebSocket message: REGISTER sip:signal.laboosh.com 
SIP/2.0
Call-ID: f8b994f5-b9d9-4899-9e28-9939c15ca5cb
CSeq: 47257 REGISTER
Contact:  
<sip:signal-1@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;click2call=n
o;+g.oma.sip-im=true;+audio=true;language=en
Via: SIP/2.0/WS df7jal23ls0d.invalid
To:  <sip:signal-1@signal.dev.laboosh.com>
From:  <sip:signal-1@signal.dev.laboosh.com>
Route:  <sip:signal.dev.laboosh.com;lr;sipml5-outbound;transport=tcp>
content-length: 0

*INFO: State machine: tsip_transac_nict_Started_2_Trying_X_send
*INFO: 

SEND: REGISTER sip:signal.laboosh.com SIP/2.0
Via: SIP/2.0/TCP 10.0.1.17:5060;branch=z9hG4bK-1277818648;rport
From: <sip:signal-1@signal.dev.laboosh.com>
To: <sip:signal-1@signal.dev.laboosh.com>
Contact: 
<sip:signal-1@10.0.1.17:5060;rtcweb-breaker=yes;transport=tcp;ws-src-ip=10.0.1.5
;ws-src-port=49870;ws-src-proto=ws>;click2call=no;+g.oma.sip-im=true;+audio=true
;language=en
Call-ID: f8b994f5-b9d9-4899-9e28-9939c15ca5cb
CSeq: 47257 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 10.0.1.5:49870;ws-hacked=WS

*INFO: Cannot find peer with remote IP/Port=10.0.1.5/5060, connecting to the 
destination...
**WARN: function: "tnet_sockfd_connectto()" 
file: "src/tnet_utils.c" 
line: "1612" 
MSG: 
TNET_ERROR_WOULDBLOCK/TNET_ERROR_ISCONN/TNET_ERROR_INPROGRESS/TNET_ERROR_EAGAIN 
 ==> use tnet_sockfd_waitUntilWritable.
*INFO: Socket added[SIP transport]: fd=13, tail.count=3
*INFO: Socket added (external call) 13
*INFO: Data send requested but peer not connected yet...saving data
*INFO: PipeR event = 1
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: Stream Peer accepted/connected - 13
*INFO: 

RECV:SIP/2.0 200 OK
Via: SIP/2.0/TCP 
10.0.1.17:5060;branch=z9hG4bK-1277818648;rport;received=10.0.1.17
Via: SIP/2.0/TCP 10.0.1.5:49870;ws-hacked=WS
To:  <sip:signal-1@signal.dev.laboosh.com>
From:  <sip:signal-1@signal.dev.laboosh.com>
call-id: f8b994f5-b9d9-4899-9e28-9939c15ca5cb
CSeq: 47257 REGISTER
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: ioctlt(4), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=14)
*INFO: Socket added[SIP transport]: fd=14, tail.count=4
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: Stream Peer accepted/connected - 14
*INFO: Stream Peer accepted/connected - 14
*INFO: 

RECV:INVITE 
sip:signal-1@10.0.1.17:5060;rtcweb-breaker=yes;transport=tcp;ws-src-ip=10.0.1.5;
ws-src-port=49870;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/TCP 10.0.1.5:5060;branch=z9hG4bK620704
Via: SIP/2.0/UDP 
10.0.1.5:52222;rport=52222;branch=z9hG4bKPjIgTV8mV7rpiwvXTlTP046eMiv.1MvAqh;rece
ived=10.0.1.5
max-forwards: 70
From: "Andre Di Genova" <sip:123@localhost>;tag=XLa076AHzMmUTUaOdan.8KQnYMdX3OHo
To:  <sip:123@10.0.1.5>
Contact:  <sip:14807623@10.0.1.5:52222>
call-id: JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G
CSeq: 23807 INVITE
allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, 
REFER
supported: 100rel, replaces, norefersub, gruu
user-agent: Blink Lite 3.0.0 (MacOSX)
content-type: application/sdp
content-length: 423

v=0
o=- 3589659155 3589659155 IN IP4 10.0.1.5
s=Blink Lite 3.0.0 (MacOSX)
c=IN IP4 10.0.1.5
t=0 0
m=audio 50022 RTP/AVP 99 98 9 0 8 104 3 96
c=IN IP4 10.0.1.5
a=rtcp:50023
a=rtpmap:99 speex/32000
a=rtpmap:98 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=20
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
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/TCP 10.0.1.5:5060;branch=z9hG4bK620704
From: "Andre Di Genova"<sip:123@localhost>;tag=XLa076AHzMmUTUaOdan.8KQnYMdX3OHo
To: <sip:123@10.0.1.5>
Call-ID: JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G
CSeq: 23807 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 
10.0.1.5:52222;rport=52222;received=10.0.1.5;branch=z9hG4bKPjIgTV8mV7rpiwvXTlTP0
46eMiv.1MvAqh

*INFO: Add call-id = 'JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G' to peer with local fd = 
14
*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
**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: State machine: s0000_Started_2_Ringing_X_iINVITE
*INFO: State machine: tsip_transac_ist_Proceeding_2_Proceeding_X_1xx
*INFO: 

SEND: SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.0.1.5:5060;branch=z9hG4bK620704
From: "Andre Di Genova"<sip:123@localhost>;tag=XLa076AHzMmUTUaOdan.8KQnYMdX3OHo
To: <sip:123@10.0.1.5>;tag=217872317
Contact: <sip:123@10.0.1.17:5060;transport=tcp>
Call-ID: JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G
CSeq: 23807 INVITE
Content-Length: 0
Via: SIP/2.0/UDP 
10.0.1.5:52222;rport=52222;received=10.0.1.5;branch=z9hG4bKPjIgTV8mV7rpiwvXTlTP0
46eMiv.1MvAqh
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

***ERROR: function: "tsk_params_get_param_value()" 
file: "src/tsk_params.c" 
line: "219" 
MSG: Invalid parameter
***ERROR: function: "tsk_params_get_param_value()" 
file: "src/tsk_params.c" 
line: "219" 
MSG: Invalid parameter
*INFO: State machine: x0500_Current_2_Current_X_oINVITE
*INFO: tsk_timer_manager_start
*INFO: ICE CTX::run -- START
*INFO: Timer manager run()::enter
*INFO: TIMER MANAGER -- START
*INFO: State machine: 
ICE_Started_2_GatheringHostCandidates_X_GatherHostCandidates
*INFO: RTP/RTCP manager[Begin]: Trying to bind to random ports [10.0.1.17:57048]
*INFO: RTP/RTCP manager[End]: Trying to bind to random ports
*INFO: local ip address = 10.0.1.17
*INFO: State machine: 
ICE_GatheringHostCandidates_2_GatheringHostCandidatesDone_X_Success
*INFO: Do not gather reflexive candidates because ICE-STUN is disabled
*INFO: ICE callback: Gathering host candidates succeed
*INFO: State machine: ICE_Any_2_GatheringCompleted_X_GatheringComplet
*INFO: ICE callback: Gathering candidates completed
*INFO: State machine: c0000_Started_2_Outgoing_X_oINVITE
*INFO: tdav_consumer_audio_init()
*INFO: Create SpeexDSP jitter buffer
**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: ICE enabled on RTP manager
*INFO: Add call-id = '12020746-5571-98ad-1773-2b6a6edae9b8' to peer with local 
fd = 12
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 100 TRYING
Via: SIP/2.0/WS 10.0.1.17:10060;branch=z9hG4bK-1652783461;rport
To:  <sip:123@10.0.1.5>
From:  <sip:123@localhost>;tag=2069426841
call-id: 12020746-5571-98ad-1773-2b6a6edae9b8
CSeq: 1847593054 INVITE
content-length: 0

*INFO: State machine: x0000_Any_2_Any_X_i1xx
*INFO: Receiving SIP o/ WebSocket message: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.1.17:10060;branch=z9hG4bK-1652783461;rport
To:  <sip:123@10.0.1.5>
From:  <sip:123@localhost>;tag=2069426841
call-id: 12020746-5571-98ad-1773-2b6a6edae9b8
CSeq: 1847593054 INVITE
content-type: application/sdp
Contact:  <sip:signal-1@df7jal23ls0d.invalid;transport=ws>
content-length: 760

v=0
o=- 2222682702064054662 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 12020746-5571-98ad-1773-2b6a6edae9b8
m=audio 1 RTP/AVPF 0 8
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:7Ul358dV62BlhxXC
a=ice-pwd:Yu4DiriQCMulRXLEOjLYhNIR
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ssrc:3859301221 cname:tsqZIec4MPhJDg+I
a=ssrc:3859301221 msid:12020746-5571-98ad-1773-2b6a6edae9b8 
12020746-5571-98ad-1773-2b6a6edae9b8
a=ssrc:3859301221 mslabel:12020746-5571-98ad-1773-2b6a6edae9b8
a=ssrc:3859301221 label:12020746-5571-98ad-1773-2b6a6edae9b8
a=candidate:2891281911 1 udp 2113937151 10.0.1.210 54429 typ host generation 0
a=candidate:3805710599 1 tcp 1509957375 10.0.1.210 51443 typ host generation 0

*INFO: State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
*INFO: tnet_ice_ctx_set_remote_candidates
*INFO: State machine: ICE_GatheringComplet_2_ConnChecking_X_ConnCheck
*INFO: is_ice_active=1,
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: ICE Pair: [VXrJzWpaO 1 10.0.1.17 57048] -> [2891281911 1 10.0.1.210 
54429]
*INFO: Remote SSRC = 3859301221
*INFO: State machine: s0000_Ringing_2_Connected_X_Accept
*INFO: State machine: tsip_transac_ist_Proceeding_2_Accepted_X_2xx
*INFO: 

SEND: SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.1.5:5060;branch=z9hG4bK620704
From: "Andre Di Genova"<sip:123@localhost>;tag=XLa076AHzMmUTUaOdan.8KQnYMdX3OHo
To: <sip:123@10.0.1.5>;tag=217872317
Contact: <sip:123@10.0.1.17:5060;transport=tcp>
Call-ID: JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G
CSeq: 23807 INVITE
Content-Type: application/sdp
Content-Length: 396
Via: SIP/2.0/UDP 
10.0.1.5:52222;rport=52222;received=10.0.1.5;branch=z9hG4bKPjIgTV8mV7rpiwvXTlTP0
46eMiv.1MvAqh
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=doubango 1983 678901 IN IP4 10.0.1.17
s=-
c=IN IP4 10.0.1.17
t=0 0
m=audio 51786 RTP/AVP 0 8
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=sendrecv
a=ssrc:1283834951 cname:66abcb681f20f6b655d5ea7d5364e670
a=ssrc:1283834951 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1283834951 label:doubango@audio

*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=10.0.1.5, rtcp.remote_port=50023, rtcp.local_fd=17
*INFO: rtcp.local_ip=10.0.1.17, rtcp.local_port=51787, rtcp.local_fd=17
*INFO: Socket added[RTP/RTCP Manager]: fd=17, tail.count=1
*INFO: pipeW (write site) not initialized yet.
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=20
*INFO: Socket added[RTP/RTCP Manager]: fd=20, tail.count=2
*INFO: master fd=16
*INFO: Socket added[RTP/RTCP Manager]: fd=16, tail.count=3
*INFO: Audio denoiser to be opened(record_frame_size_samples=160, 
record_sampling_rate=8000, playback_frame_size_samples=160, 
playback_sampling_rate=8000)
*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: Transport::run() - enter
*INFO: SO_RCVBUF = 65535, SO_SNDBUF = 65535
*INFO: rtcp.remote_ip=10.0.1.210, rtcp.remote_port=54429, rtcp.local_fd=19
*INFO: tsk_timer_manager_start
*INFO: Timer manager already running
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=22
*INFO: Socket added[RTP/RTCP Manager]: fd=22, tail.count=1
*INFO: master fd=19
*INFO: Socket added[RTP/RTCP Manager]: fd=19, tail.count=2
warning: The VAD has been replaced by a hack pending a complete rewrite
*INFO: 

RECV:ACK sip:123@10.0.1.17:5060;transport=tcp SIP/2.0
To:  <sip:123@10.0.1.5>;tag=217872317
From: "Andre Di Genova" <sip:123@localhost>;tag=XLa076AHzMmUTUaOdan.8KQnYMdX3OHo
call-id: JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G
CSeq: 23807 ACK
Via: SIP/2.0/TCP 10.0.1.5:5060;branch=z9hG4bK76125
content-length: 0

*INFO: Starting [RTP/RTCP Manager] server with IP {10.0.1.17} on port {51786} 
using fd {16} with type {3}...
*INFO: State machine: tsip_transac_ist_Accepted_2_Accepted_iACK
*INFO: Transport::run() - enter
*INFO: Audio denoiser to be opened(record_frame_size_samples=160, 
record_sampling_rate=8000, playback_frame_size_samples=160, 
playback_sampling_rate=8000)
*INFO: State machine: x0000_Connected_2_Connected_X_iACK
warning: The VAD has been replaced by a hack pending a complete rewrite
*INFO: Starting [RTP/RTCP Manager] server with IP {10.0.1.17} on port {57048} 
using fd {19} with type {3}...
*INFO: Using symetric RTCP for [10.0.1.5]:50023
*INFO: Using symetric RTP for [10.0.1.5]:50022

*INFO: TMMBN
*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: ioctlt(13), len=0 returned zero or failed
*INFO: Closing socket with fd = 13 because ioctlt() returned zero or failed
*INFO: Removing socket 13
*INFO: Socket to remove: fd=13, index=2, tail.count=4
*INFO: CloseSocket(13)
*INFO: PipeR event = 1
*INFO: Stream Peer closed - 13
*INFO: SIP socket closed
*INFO: State machine: x9998_Any_2_Terminated_X_transportError
*INFO: === INVITE Dialog terminated ===
*INFO: tmedia_session_mgr_stop()
*INFO: Transport::run(RTP/RTCP Manager) - exit
*INFO: Stopping [RTP/RTCP Manager] server with IP {10.0.1.17} on port {51786} 
with type {3}...
*INFO: Stopped [RTP/RTCP Manager] server with IP {10.0.1.17} on port {51786}
*INFO: Socket to remove: fd=17, index=0, tail.count=3
*INFO: Socket to remove: fd=20, index=0, tail.count=2
*INFO: CloseSocket(20)
*INFO: Socket to remove: fd=16, index=0, tail.count=1
*INFO: CloseSocket(16)
*INFO: CloseSocket(17)
*INFO: [Stream] Removed call-id = 'JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G' from peer 
with local fd = 14
*INFO: [Transport] Removed call-id = 'JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G' from 
transport with type = 8
*INFO: [Transport Layer] Removed call-id = 'JmIXPjtsN0uoc55cELbjqkKJGMMqfS3G' 
from transport layer
*INFO: *** INVITE Dialog destroyed ***
*INFO: State machine: x9998_Any_2_Terminated_X_transportError
*INFO: === INVITE Dialog terminated ===
*INFO: tmedia_session_mgr_stop()
*INFO: Transport::run(RTP/RTCP Manager) - exit
*INFO: Stopping [RTP/RTCP Manager] server with IP {10.0.1.17} on port {57048} 
with type {3}...
*INFO: Stopped [RTP/RTCP Manager] server with IP {10.0.1.17} on port {57048}
*INFO: Socket to remove: fd=22, index=0, tail.count=2
*INFO: CloseSocket(22)
*INFO: Socket to remove: fd=19, index=0, tail.count=1
*INFO: Timer manager run()::exit
*INFO: TIMER MANAGER -- STOP
*INFO: ICE CTX::run -- STOP
*INFO: CloseSocket(18)
*INFO: CloseSocket(19)
*INFO: [Stream] Removed call-id = '12020746-5571-98ad-1773-2b6a6edae9b8' from 
peer with local fd = 12
*INFO: [Transport] Removed call-id = '12020746-5571-98ad-1773-2b6a6edae9b8' 
from transport with type = 64
*INFO: [Transport Layer] Removed call-id = 
'12020746-5571-98ad-1773-2b6a6edae9b8' from transport layer
*INFO: *** INVITE Dialog destroyed ***
*INFO: *** Stream Peer destroyed ***
*INFO: Stream Peer closed - 13
*INFO: SIP socket closed
***ERROR: function: "tsip_ssession_handle()" 
file: "src/tsip_ssession.c" 
line: "607" 
MSG: Failed to find dialog with this opid [2]
***ERROR: function: "tsip_api_invite_send_bye()" 
file: "src/api/tsip_api_invite.c" 
line: "311" 
MSG: Bye() failed.
*INFO: MPPeer object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** ICE context destroyed ***
*INFO: MPSipSession object destroyed
*INFO: MPSipSessionAV object destroyed
*INFO: *** tdav_session_audio_t destroyed ***
*INFO: *** SpeexDSP denoiser destroyed ***
*INFO: *** SpeexDSP jb destroyed ***
*INFO: MPProxyPluginConsumerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: MPProxyPluginProducerAudio object destroyed
*INFO: *** RTP manager destroyed ***
*INFO: *** Audio session destroyed ***
*INFO: *** SIP Session destroyed ***
*INFO: MPSipSession object destroyed
*INFO: *** SIP Session destroyed ***
*INFO: ioctlt(14), len=0 returned zero or failed
*INFO: Closing socket with fd = 14 because ioctlt() returned zero or failed
*INFO: Removing socket 14
*INFO: Socket to remove: fd=14, index=2, tail.count=3
*INFO: CloseSocket(14)
*INFO: PipeR event = 1
*INFO: Stream Peer closed - 14
*INFO: *** Stream Peer destroyed ***
*INFO: Stream Peer closed - 14
Original issue reported on code.google.com by an...@symbicode.com on 1 Oct 2013 at 11:41
edgar017 / webrtc2sip

A call with a SIP endpoint over TCP results in a call disconnection when the TCP SIP connection closes. #132
