[webrtc2sip] - Githubissues

What steps will reproduce the problem?
1. Build webrtc2sip with latest trunk (doubango 1008 and webrtc2sip 121)
2. Run webrtc2sip
3. Try to connect/register to a trunk with sipml5

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

Expected: sipml5 connects to the trunk
Actual: webrtc2sip closes the websocket connection immediately and you see the 
error

***ERROR: function: "tsip_transport_layer_ws_cb()" 
file: "src/transports/tsip_transport_layer.c" 
line: "397" 
MSG: WS handshaking not done yet

What version of the product are you using? On what operating system?
doubango 1008 and webrtc2sip 121 on ubuntu 12.10

Please provide any additional information below.

root@stage:/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: 2.6.0
'quit' to quit the application.
*******************************************************************

SSL is enabled :)
DTLS supported: yes
DTLS-SRTP supported: yes
*INFO: transport = udp://*:10060
*INFO: transport = ws://*:10060
*INFO: transport = wss://*:10062
*INFO: transport = tcp://*:5060
*INFO: enable-rtp-symetric = yes
*INFO: enable-100rel = no
*INFO: enable-media-coder = no
*INFO: enable-videojb = no
*INFO: video-size-pref = vga
*INFO: rtp-buffsize = 65535
*INFO: avpf-tail-length = [100-400]
*INFO: srtp-mode = optional
*INFO: srtp-type = sdes;dtls
*INFO: dtmf-type = rfc4733
*INFO: codecs = opus;pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+
*INFO: 'opus' codec enabled but not supported
*INFO: UnRegister codec: PCMA, G.711a codec (native)
*INFO: UnRegister codec: PCMU, G.711u codec (native)
*INFO: UnRegister codec: GSM, GSM Full Rate (libgsm)
*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: stun-server = stun.l.google.com;19302;-;-
*INFO: enable-icestun = no
*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=9
*INFO: Socket added[SIP transport]: fd=9, tail.count=1
*INFO: master fd=5
*INFO: Socket added[SIP transport]: fd=5, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=11
*INFO: Socket added[SIP transport]: fd=11, tail.count=1
*INFO: master fd=6
*INFO: Socket added[SIP transport]: fd=6, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=13
*INFO: Socket added[SIP transport]: fd=13, tail.count=1
*INFO: master fd=7
*INFO: Socket added[SIP transport]: fd=7, tail.count=2
*INFO: tnet_transport_prepare()
*INFO: pipeR fd=15
*INFO: Socket added[SIP transport]: fd=15, tail.count=1
*INFO: master fd=8
*INFO: Socket added[SIP transport]: fd=8, tail.count=2
*INFO: SIP STACK -- START
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Transport::run() - enter
*INFO: Starting [SIP transport] server with IP {198.199.123.172} on port 
{10062} using fd {8} with type {128}...
*INFO: Starting [SIP transport] server with IP {198.199.123.172} on port 
{10060} using fd {7} with type {64}...
*INFO: Starting [SIP transport] server with IP {198.199.123.172} on port {5060} 
using fd {6} with type {8}...
*INFO: Starting [SIP transport] server with IP {198.199.123.172} on port 
{10060} using fd {5} with type {2}...

*INFO: ioctlt(7), len=0 returned zero or failed
*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- FD_ACCEPT(fd=17)
*INFO: Socket added[SIP transport]: fd=17, tail.count=3
*INFO: WebSocket Peer accepted/connected with fd = 17
*INFO: #1 peers in the 'SIP transport' transport
*INFO: WebSocket handshake message: GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: stage.laboosh.com:10060
Origin: http://sipml5.org
Sec-WebSocket-Protocol: sip
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: 5V1FthoySzcFv3D1X8O80g==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_8_5) AppleWebKit/537.36 
(KHTML, like Gecko) Chrome/30.0.1599.101 Safari/537.36
Cookie: fbm_532352173445102=base_domain=.laboosh.com

*INFO: NETWORK EVENT FOR SERVER [SIP transport] -- TNET_POLLOUT
*INFO: WebSocket Peer accepted/connected with fd = 17
*INFO: *** Stream Peer destroyed ***
*INFO: #0 peers in the 'SIP transport' transport
*INFO: #1 peers in the 'SIP transport' transport
***ERROR: function: "tsip_transport_layer_ws_cb()" 
file: "src/transports/tsip_transport_layer.c" 
line: "397" 
MSG: WS handshaking not done yet
*INFO: Removing socket 17
*INFO: Socket to remove: fd=17, index=2, tail.count=3
*INFO: CloseSocket(17)
*INFO: WebSocket Peer closed with fd = 17
*INFO: #0 peers in the 'SIP transport' transport
*INFO: *** Stream Peer destroyed ***
*INFO: PipeR event = 1
Original issue reported on code.google.com by andre.di...@gmail.com on 22 Oct 2013 at 4:57
tahirjilani / doubango

[webrtc2sip] #324
