Registration to Freeswitch ok, dialing not possible

[GoogleCodeExporter]

a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?

There is only one machine. Everything is installed on this machine in a local 
network.

Install sipml5, webrtc2sip, Doubango IMS framework and Freeswitch.

Run webrtc2sip, config.xml changed to set logging output to INFO.

In sipml5: Disable Video, Enable RTCWeb Breaker are both checked.
WebSocket Server URL[2]: ws:https://192.168.118.8:10060
SIP outbound Proxy URL[3]: udp:https://192.168.118.8:5060

Display Name: SipML5

Private Identity*: 1001

Public Identity*: sip:[email protected]

Password: ****

Realm*: 192.168.118.8

Registration to Freeswitch works ok.

Trying to place a call, I get the following message in the sipml5 window in 
Chrome: "Call in progress..."

JS console output:

State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:110
==stack event = m_permission_requested call.htm:632
==session event = connecting call.htm:720

tcpdump -i lo shows traffic on registration. But there is no output of tcpdump, 
even not on any other network interfaces, which indicates any traffic 
concerning the dialing process.


What is the expected output? What do you see instead?
There should be network traffic as the sipml5 is dialing out...


What version of the product are you using? On what operating system?
Fedora F17 linux with Chrome Version 24.0.1312.52 beta


Please provide any additional information below.
I use the sample config.xml in webrtc2sip. webrtc2sip and doubango IMS 
framework are latest svn as of today.

Original issue reported on code.google.com by [email protected] on 19 Jan 2013 at 2:58

[GoogleCodeExporter]

I don't know wether this is a defect, maybe I am just doing something the wrong 
way. Sorry if I am bothering you just because I am doing something simple in 
the wrong way, but I have spent days on this now and I would really like to get 
sipml5 going, because I think it's a very nice and elegant thing.

It's not only that tcpdump does not show any traffic, the INFO level logging 
output of webrtc2sip stays silent, too. Though logging shall be ok, because it 
shows logging messages at startup of webrtc2sip and on registering sipml5 to FS.

webrtc2sip-start.txt shows logging messages after webrtc2sip has been launched.
webrtc2sip-register.txt shows logging messages after registering sipml5 to FS.

On sipml5 dialing, I only get js console output as indicated above. There is no 
output on the INFO level webrtc2sip logs.

Original comment by [email protected] on 19 Jan 2013 at 10:16

Attachments:

• webrtc2sip-start.txt
• webrtc2sip-register.txt

[GoogleCodeExporter]

[deleted comment]

[GoogleCodeExporter]

i have the same issue when use mcu.

Original comment by [email protected] on 26 Sep 2013 at 7:03