Comments (9)
[deleted comment]
from webrtc2sip.
Have you edited the wireshark trace? Because there is no server reflexive
candidate i the 200 OK from webrtc2sip to chrome.
Original comment by [email protected]
on 16 Jan 2013 at 12:08
from webrtc2sip.
The webrtc2sip is not supposed to have reflexive candidates, since it is not
behind a NAT. More precisely, the webrtc2sip and Chrome are on a same private
network without access to internet.
Original comment by [email protected]
on 16 Jan 2013 at 9:32
from webrtc2sip.
New network trace
Original comment by [email protected]
on 16 Jan 2013 at 10:50
Attachments:
from webrtc2sip.
Very strange because you said that it was working with the previous version.
Looks at the code source it looks like the only conditions to stop gathering
reflexive candidates are "network error" or "timeout". In the current state,
you should always receive "timeout" error if you don't have access to internet.
As this issue is not related to chrome, I've opened ticket 197 on Doubango
(http://code.google.com/p/doubango/issues/detail?id=197)
Original comment by [email protected]
on 16 Jan 2013 at 11:18
- Changed state: Accepted
from webrtc2sip.
Some news about that issue :
I have deactivated STUN server on the browser side (i.e. in the SIPML5), and
changed the STUN server IP on the webrtc2sip side to set a reachable (thus
internal) STUN server.
-> Result : the issue is solved
Note that using the reachable STUN server on the browser side generates still
an issue leading to a "SetRemoteDescription Failed" on the browser side. This
SetRemoteDescription failure occurs after the SIPML5 tries to initiate a second
offer/answer exchange.
Original comment by [email protected]
on 21 Feb 2013 at 12:40
from webrtc2sip.
you should try latest code, some ICE issues have been fixed
Original comment by [email protected]
on 27 Feb 2013 at 7:44
from webrtc2sip.
I tried to install stund (from http://sourceforge.net/projects/stun/) locally
and use it with webrtc2sip but I get this error messages bellow from stun
server console. Stéphane, could you tell me which stun server did you use on
your local network to make it work?
Thanks,
Minh
-------------------------
Received stun message: 104 bytes
Bad length strign 9
problem parsing Username
Request did not parse
Failed to parse message
Original comment by [email protected]
on 12 Mar 2013 at 2:58
from webrtc2sip.
Nevermind, I manage to make it work by using user name and password with the
right length. STUND requires the lengh of username and password to be a
divisible by 4
Original comment by [email protected]
on 12 Mar 2013 at 3:16
from webrtc2sip.
Related Issues (20)
- Webrtc2sip Installing but net/if_dl.h / net/if_types.h not found in doubango compiling
- Problem in BYE send from Webrtc2sip (Version 2.6.0) HOT 1
- asterisk 11.2.1+sipml5+webrtc2sip - No audio in chrome, audio available in firefox HOT 1
- Unable to start webrtc2sip using init.d script
- 183/early media not forwarded to browser by webrtc2sip
- Webrtc2sip crash when media coder is enabled.
- how to make sip client direct to gateway ? HOT 1
- Webrtc2sip crashing randomly in SIPML5 scenario HOT 1
- The git directory miss the tinywrap folder and it's contents and can therefore not compile HOT 4
- No Audio if call is accepted after 30 seconds HOT 2
- tinywrap folder should be external from doubango project
- Installed webrtc2sip but can't use the command
- Using WSS with WebRTC and Letsencrypt (Let's Encrypt) via sipML = "no shared cipher" HOT 1
- High CPU during calls also with same codecs
- Not respecting original User-Agent on INVITE and REGISTER
- Got a problem with calls HOT 1
- How to compile webrtc2sip correctly? HOT 1
- Redirect call from webapp to sip phone with 302 Moved Temporarily message
- Integration with freepbx or fusionpbx HOT 1
- libvpx.so.1 not found
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