The delay from allow mic and start call to send invite delayed aoubt 10 secondes

[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?
1.use sipml5 connect to webrtc2sip server and make a call
2.after click call , chrome ask to allow mic
3.click allow mic the sipml start to show "call in process"
4.after about 10 seconds status changed to remote ring
5. from the wireshark and check the exact time. The chrome ask to send invite 
and the webrtc2sip send invite after 10 second start to call . 

What is the expected output? What do you see instead?
call flow ok . but I have to wait so long time.

What version of the product are you using? On what operating system?
dont now how to get it .checked out on 2013/10/29

Please provide any additional information below.

Original issue reported on code.google.com by zhujian...@gmail.com on 7 Nov 2013 at 1:39

[GoogleCodeExporter]

disable ICE reflexive candidates gathering to speed up the process

Original comment by boss...@yahoo.fr on 7 Nov 2013 at 8:05

[GoogleCodeExporter]

I would like to check with you where to disable?
webrtc2sip / jsp page / or softswitch?

I use FS to test  and I found asterisk has a patch for disable ICE
reflexive candidates gathering.

what I found is that the local js in chrome speed 10 seconds to waiting but
I can not find ICE reflexive candidates gathering configurations in js.

Thanks.

Original comment by zhujian...@gmail.com on 12 Nov 2013 at 5:33

[GoogleCodeExporter]

I found something in SIPml.js

To disable TURN/STUN to speedup ICE candidates gathering you can use an
empty array. e.g. <i>[]</i>. <br />
Example: <i>[{ url: 'stun:stun.l.google.com:19302'}, {
url:'turn:user@numb.viagenie.ca', credential:'myPassword'}]</i>

but after doing this . Still get 10 seconds delay before remote ring.

2013/11/12 David Zhu <zhujiankai@gmail.com>

Original comment by zhujian...@gmail.com on 12 Nov 2013 at 5:52

[GoogleCodeExporter]

*     ice_servers: ("[]"),*

I found it . When it goes to SIP .no ice need.

But another question is that if I want to make video call and need top do
p2p traffic without occupy the server's bandwidth .How can I do it? I think
in the scenario ice is needed.  Thanks

2013/11/12 David Zhu <zhujiankai@gmail.com>

Original comment by zhujian...@gmail.com on 12 Nov 2013 at 7:03