[jitsi-users] Jitsi error: "call ended by remote side - Reason failed-application... (ICE failed)"


#1

Hi everyone,
I really like the idea of an alternative free and open software to make encrypted video calls.
Today I tried Jitsi with a friend (Win8.1 and Win8.0). We both created XMPP Accounts, using the german site jabber.ccc.de
We could chat - also encrypted - but we were not able to start a phone call.
Jitsi did not start any video connection and said:
"call ended by remote side. Reason: failed-application. Error: Could not establish connection (ICE failed)".
Unfortunately, I tried this half a year ago and it also did not work. But I found the following link:
where this problem was discussed one and two years ago without any solution.
So could you please help me in what might be going wrong?
Can i solve the problem and is it possible to improve the error-message to state more clearly to a normal user what is going on? I assume that more users install and uninstall this software after this annoying experience. But I would like to recommend more people using this software!
Thanks in advance
Markus
···

http://forum.ubuntuusers.de/topic/jitsi-baut-keine-sprechverbindung-auf/#post-3835057


#2

You're probably using a NAT router. Jitsi won't work well with that.


#3

See also
https://jitsi.org/GSOC/AdvancedNATTraversal

FC

···

On Thu, Apr 24, 2014 at 3:30 AM, Mitchell Langs <napole@gmail.com> wrote:

You're probably using a NAT router. Jitsi won't work well with that.

--
During times of Universal Deceit, telling the truth becomes a revolutionary
act
- George Orwell


#4

This provides more info. You might also want to check if your routers
(both users) have UPNP enabled?.

https://jitsi.org/Documentation/FAQ#stun
...
Does Jitsi support STUN? (and how about TURN, UPnP and Jingle Nodes?)

STUN, together with TURN, Jingle Nodes, IPv6 and UPnP, is one of the
techniques that Jitsi uses as part of the Interactive Connectivity
Establishment (ICE <http://tools.ietf.org/html/rfc5245>) protocol to handle
NAT traversal for calls made over XMPP.

For its SIP calls, Jitsi currently relies on servers to relay media (a
technique also known as Hosted NAT Traversal or
latching<http://tools.ietf.org/html/draft-ietf-mmusic-latching>,
which would be the case of the majority of the SIP servers used on the
Internet today. Note that in terms of reliability Hosted NAT Traversal
gives the same results as use of ICE. It even works better in some ways
because the connection is setup immediately and no time is waisted for
gathering candidates and making connectivity checks. The only downside of
HNT is that it may put a strain on SIP providers requiring more bandwidth.
This could become a problem especially in environments with a high number
of all IP high quality video calls.

*It is likely that ICE support for SIP calls would also be added to Jitsi
in 2014 especially since this would also help with WebRTC compatibility. *

....

The devs might be able to comment if this is already enabled in nightly
builds...

So, just in case, you might want to test the latest nightly builds and see
if it makes any difference. But make sure you're both running the same
build/version...

https://download.jitsi.org/jitsi/nightly/windows/

FC

···

On Thu, Apr 24, 2014 at 3:38 AM, Fernando Cassia <fcassia@gmail.com> wrote:

See also
https://jitsi.org/GSOC/AdvancedNATTraversal

--
During times of Universal Deceit, telling the truth becomes a revolutionary
act
Durante épocas de Engaño Universal, decir la verdad se convierte en un Acto
Revolucionario
- George Orwell