[jitsi-users] Where can I set a STUN server ?


#1

Can somebody tell me where I can set a STUN server in jitsi ?
thanks for any help, Mark


#2

I just found out that jitsi is not supporting STUN with SIP accounts. It is in the FAQ, but "hidden" in the explanation why ekiga.net is not working. I would suggest to update the FAQ and give STUN a headline and make ekiga a different topic with a reference to STUN.
As I do have constant trouble with SIP and NAT traversal I would like to use an XMPP VoIP (telephone) provider. Can anyone recommend a provider that uses XMPP, that works well with jitsi and that is not bound to the US like Google Voice - I live in the UK.

thanks a lot, Mark

···

On 06/06/11 18:04, Mark Dammer wrote:

Can somebody tell me where I can set a STUN server in jitsi ?
thanks for any help, Mark


#3

I didn't test it with jitsi, but did you try jabber? AFAIR linphone
offers voice via jabber, so it could work with jitsi too.

Did you try iptel via sip? It works without stun here. Then there are
some German freemailers, like gmx and web.de. It worked, but I see no
reason to use it.

Did you try sipgate.co.uk without stun-server? Sipgate.at works fine
for me including video.

Al

···

Am Mo, 06 Jun 2011 20:21:22 CEST schrieb Mark Dammer:

I just found out that jitsi is not supporting STUN with SIP accounts.
It is in the FAQ, but "hidden" in the explanation why ekiga.net is
not working. I would suggest to update the FAQ and give STUN a
headline and make ekiga a different topic with a reference to STUN.
As I do have constant trouble with SIP and NAT traversal I would like
to use an XMPP VoIP (telephone) provider. Can anyone recommend a
provider that uses XMPP, that works well with jitsi and that is not
bound to the US like Google Voice - I live in the UK.

thanks a lot, Mark


#4

На 06.06.11 21:21, Mark Dammer написа:

I just found out that jitsi is not supporting STUN with SIP accounts. It
is in the FAQ, but "hidden" in the explanation why ekiga.net is not
working. I would suggest to update the FAQ and give STUN a headline and
make ekiga a different topic with a reference to STUN.

Indeed! A lot of folks seem to be interested in STUN support so I just
added this:

http://jitsi.org/faq/stun

Thanks for the suggestion!

As I do have constant trouble with SIP and NAT traversal I would like to
use an XMPP VoIP (telephone) provider.

OK, this is probably a good time to remind that lack of STUN support,
does absolutely not imply lack of NAT support with SIP. The majority of
the SIP servers out there (like iptel.org, ippi.com and many others),
would put themselves on the media path and hence allow media to flow
regardless of where the clients are.

Granted, this would not be a direct connection and it would require the
provider to maintain a certain amount of bandwidth, but in terms of
reliability, NAT traversal doesn't get any better than that.

ICE is just a mechanism, which helps make sure that such relaying is
only used when a direct connection is not possible. In other words, as
long as both clients can freely access the SIP server/media relay, ICE
would be merely an optimisation in terms of performance, and not in
terms of reliability.

Can anyone recommend a provider
that uses XMPP, that works well with jitsi and that is not bound to the
US like Google Voice - I live in the UK.

Since you are mentioning Google Voice, I suppose PSTN access has to be
part of the service. Nimbuzz and Google Talk are hence the only two that
come to my mind right now and they both require the Google dialect of
Jingle. We we are still working on it though.

Emil

···

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31


#5

Emil - and all the other developers and supporters,
you guys are doing a REALLY good job. The speed of development of this project is outstanding.

You are mentioning Nimbuzz. I had a look at their website, but it looks to me as they only support their own proprietary software (although I read talk about Nimbuzz accounts being just "ordinary" jabber/xmpp accounts. Has anyone used Nimbuzz with Jitsi ?

all the best, Mark

···

On 06/06/11 22:09, Emil Ivov wrote:

Can anyone recommend a provider
that uses XMPP, that works well with jitsi and that is not bound to the
US like Google Voice - I live in the UK.

Since you are mentioning Google Voice, I suppose PSTN access has to be
part of the service. Nimbuzz and Google Talk are hence the only two that
come to my mind right now and they both require the Google dialect of
Jingle. We we are still working on it though.

Emil


#6

I have a sipgate account, but it does not seem to work with jitsi - that is why I am asking. It registers and when I try to call a phone number or their 10000 test number jitsi tells me that the call is being initiated and thats it. Maybe there is something wrong with my account.
Mark

···

On 06/06/11 20:45, Al Bogner wrote:

Am Mo, 06 Jun 2011 20:21:22 CEST schrieb Mark Dammer:

I just found out that jitsi is not supporting STUN with SIP accounts.
It is in the FAQ, but "hidden" in the explanation why ekiga.net is
not working. I would suggest to update the FAQ and give STUN a
headline and make ekiga a different topic with a reference to STUN.
As I do have constant trouble with SIP and NAT traversal I would like
to use an XMPP VoIP (telephone) provider. Can anyone recommend a
provider that uses XMPP, that works well with jitsi and that is not
bound to the US like Google Voice - I live in the UK.

thanks a lot, Mark

I didn't test it with jitsi, but did you try jabber? AFAIR linphone
offers voice via jabber, so it could work with jitsi too.

Did you try iptel via sip? It works without stun here. Then there are
some German freemailers, like gmx and web.de. It worked, but I see no
reason to use it.

Did you try sipgate.co.uk without stun-server? Sipgate.at works fine
for me including video.

Al


#7

Hi Emil,

First of all: you guys rock! Awesome software AND awesome way to run
this project! My highest respect!

I had a browse around the FAQ regarding XMPP/Gmail chat/GoogleTalk but
couldn't find the answer to the problem I have:

If I log in to a Gmail account via Jitsi I'm able to chat to people
signed in to Gmail or GoogleTalk application. But I'm not able to make
calls/video calls :frowning: Did I miss something to set up properly or does
it have to do with the Google dialect Jingle you mentioned below? I
can successfully make voice/video calls between two Gmail accounts
logged in via Jitsi though.

Thanks in advance!

John

···

On Tue, Jun 7, 2011 at 9:09 AM, Emil Ivov <emcho@jitsi.org> wrote:

На 06.06.11 21:21, Mark Dammer написа:

I just found out that jitsi is not supporting STUN with SIP accounts. It
is in the FAQ, but "hidden" in the explanation why ekiga.net is not
working. I would suggest to update the FAQ and give STUN a headline and
make ekiga a different topic with a reference to STUN.

Indeed! A lot of folks seem to be interested in STUN support so I just
added this:

http://jitsi.org/faq/stun

Thanks for the suggestion!

As I do have constant trouble with SIP and NAT traversal I would like to
use an XMPP VoIP (telephone) provider.

OK, this is probably a good time to remind that lack of STUN support,
does absolutely not imply lack of NAT support with SIP. The majority of
the SIP servers out there (like iptel.org, ippi.com and many others),
would put themselves on the media path and hence allow media to flow
regardless of where the clients are.

Granted, this would not be a direct connection and it would require the
provider to maintain a certain amount of bandwidth, but in terms of
reliability, NAT traversal doesn't get any better than that.

ICE is just a mechanism, which helps make sure that such relaying is
only used when a direct connection is not possible. In other words, as
long as both clients can freely access the SIP server/media relay, ICE
would be merely an optimisation in terms of performance, and not in
terms of reliability.

Can anyone recommend a provider
that uses XMPP, that works well with jitsi and that is not bound to the
US like Google Voice - I live in the UK.

Since you are mentioning Google Voice, I suppose PSTN access has to be
part of the service. Nimbuzz and Google Talk are hence the only two that
come to my mind right now and they both require the Google dialect of
Jingle. We we are still working on it though.

Emil

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31


#8

I don't have a sipgate account but I've seen this behavior with other providers. I don't know if you've tried this already but disable all audio and video codecs in Jitsi and enable only 1 or 2 (provider-supported, of course).

Vieri

···

--- On Tue, 6/7/11, MarkD <clunymark@yahoo.co.uk> wrote:

I have a sipgate account, but it does not seem to work with
jitsi - that
is why I am asking. It registers and when I try to call a
phone number
or their 10000 test number jitsi tells me that the call is
being
initiated and thats it. Maybe there is something wrong with
my account.


#9

hi john!

sorry to pop in to this conversation here, but that's exactly it. A
Gmail/Gtalk client doesn't exacly use jingle. You have to differentiate
with audio/video.
but that brings me to a question that came to my mind: I thought
audio/video functionality over XMPP is _entirely_ dependent on the
client you use, and only the client. So I don't really get the
discussion about "jingle works / doesn't work on this or that
server/provider". It's _just_ the client, isn't it?

thanks!

  martin

···

Am 07.06.2011 02:10, schrieb rsnewsbox:

Hi Emil,

First of all: you guys rock! Awesome software AND awesome way to run
this project! My highest respect!

I had a browse around the FAQ regarding XMPP/Gmail chat/GoogleTalk but
couldn't find the answer to the problem I have:

If I log in to a Gmail account via Jitsi I'm able to chat to people
signed in to Gmail or GoogleTalk application. But I'm not able to make
calls/video calls :frowning: Did I miss something to set up properly or does
it have to do with the Google dialect Jingle you mentioned below? I
can successfully make voice/video calls between two Gmail accounts
logged in via Jitsi though.

Thanks in advance!

John

--
                   Martin Kepplinger

E-Mail & Jabber martinkepplinger AT eml.cc
GPG Key C100 D7B5 7F2A 1E26
diaspora handle martinkepplinger AT joindiaspora.com

On Tue, Jun 7, 2011 at 9:09 AM, Emil Ivov <emcho@jitsi.org> wrote:

На 06.06.11 21:21, Mark Dammer написа:

I just found out that jitsi is not supporting STUN with SIP accounts. It
is in the FAQ, but "hidden" in the explanation why ekiga.net is not
working. I would suggest to update the FAQ and give STUN a headline and
make ekiga a different topic with a reference to STUN.

Indeed! A lot of folks seem to be interested in STUN support so I just
added this:

http://jitsi.org/faq/stun

Thanks for the suggestion!

As I do have constant trouble with SIP and NAT traversal I would like to
use an XMPP VoIP (telephone) provider.

OK, this is probably a good time to remind that lack of STUN support,
does absolutely not imply lack of NAT support with SIP. The majority of
the SIP servers out there (like iptel.org, ippi.com and many others),
would put themselves on the media path and hence allow media to flow
regardless of where the clients are.

Granted, this would not be a direct connection and it would require the
provider to maintain a certain amount of bandwidth, but in terms of
reliability, NAT traversal doesn't get any better than that.

ICE is just a mechanism, which helps make sure that such relaying is
only used when a direct connection is not possible. In other words, as
long as both clients can freely access the SIP server/media relay, ICE
would be merely an optimisation in terms of performance, and not in
terms of reliability.

Can anyone recommend a provider
that uses XMPP, that works well with jitsi and that is not bound to the
US like Google Voice - I live in the UK.

Since you are mentioning Google Voice, I suppose PSTN access has to be
part of the service. Nimbuzz and Google Talk are hence the only two that
come to my mind right now and they both require the Google dialect of
Jingle. We we are still working on it though.

Emil

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31


#10

Hey folks,

На 07.06.11 04:02, Martin Kepplinger написа:

Hi Emil,

First of all: you guys rock! Awesome software AND awesome way to run
this project! My highest respect!

Thanks a lot for the kind words!

I had a browse around the FAQ regarding XMPP/Gmail chat/GoogleTalk but
couldn't find the answer to the problem I have:

If I log in to a Gmail account via Jitsi I'm able to chat to people
signed in to Gmail or GoogleTalk application. But I'm not able to make
calls/video calls :frowning: Did I miss something to set up properly or does
it have to do with the Google dialect Jingle you mentioned below?

Yes. The Gmail and GTalk clients use a variant of Jingle that we are
currently working on but that we don't support yet.

but that brings me to a question that came to my mind: I thought
audio/video functionality over XMPP is _entirely_ dependent on the
client you use, and only the client. So I don't really get the
discussion about "jingle works / doesn't work on this or that
server/provider". It's _just_ the client, isn't it?

Most of the time this is indeed the case. There is however something
that the server can do to make sure that would have a significant impact
on whether or not conversations succeed: providing a (Jingle Nodes)
relay service.

People using jabber.org or talkr.im, both of which offer JN relays,
would be able to always establish a call, no matter what kind of NAT
they are behind.

In the case of other services (including the combination of Gmail and
Jitsi) would only be able to do so if there's a possibility to establish
a direct connection between the two peers, or if they have manually
configured a TURN or a Jingle Nodes relay.

Hope this helps,
Emil

···

Am 07.06.2011 02:10, schrieb rsnewsbox:


#11

Hey Mark,

На 07.06.11 10:07, MarkD написа:

Emil - and all the other developers and supporters,
you guys are doing a REALLY good job. The speed of development of this
project is outstanding.

Thanks! We are doing our best! :slight_smile:

You are mentioning Nimbuzz. I had a look at their website, but it looks
to me as they only support their own proprietary software (although I
read talk about Nimbuzz accounts being just "ordinary" jabber/xmpp
accounts. Has anyone used Nimbuzz with Jitsi ?

It is indeed "ordinary" XMPP, with the exception of their telephony
features which use the Google dialect of Jingle that we are still
working on integrating.

Cheers,
Emil

···

all the best, Mark

On 06/06/11 22:09, Emil Ivov wrote:

Can anyone recommend a provider
that uses XMPP, that works well with jitsi and that is not bound to the
US like Google Voice - I live in the UK.

Since you are mentioning Google Voice, I suppose PSTN access has to be
part of the service. Nimbuzz and Google Talk are hence the only two that
come to my mind right now and they both require the Google dialect of
Jingle. We we are still working on it though.

Emil


#12

Thanks for the clarification, Emil!

I had a browse around the FAQ regarding XMPP/Gmail chat/GoogleTalk but
couldn't find the answer to the problem I have:

If I log in to a Gmail account via Jitsi I'm able to chat to people
signed in to Gmail or GoogleTalk application. But I'm not able to make
calls/video calls :frowning: Did I miss something to set up properly or does
it have to do with the Google dialect Jingle you mentioned below?

Yes. The Gmail and GTalk clients use a variant of Jingle that we are
currently working on but that we don't support yet.

That leaves some hope :slight_smile:
Honestly I see it as vital and I can't wait to see this working. As
much as I'm prepared to tweak stuff, install work arounds... I can't
and don't wanna force my friends doing the same, including installing
a program (in this case Jitsi) as long as their are happy to do all
their communication from within Gmail, which is fair enough. Of course
I'm recommending Jitsi wherever I can :slight_smile: I do understand however it's
Googles Jingle dialect and not Jitsi in the first place.
Just in case you've not heard about a program who is already able to
obviously speak that dialect:
https://www.talkonaut.com
I've got it installed on a Symbian phone and voice calls (via a Gmail
account) to people logged in to Gmail/GTalk work like a charm. Not
sure if the guys from talkonaut/GTalk2VoIP are prepared to share some
knowledge?

People using jabber.org or talkr.im, both of which offer JN relays,
would be able to always establish a call, no matter what kind of NAT
they are behind.

In the case of other services (including the combination of Gmail and
Jitsi) would only be able to do so if there's a possibility to establish
a direct connection between the two peers, or if they have manually
configured a TURN or a Jingle Nodes relay.

So did I get it, as long as people are logged in to Gmail/Gtalk it
doesn't matter what kind of account I use via Jitsi (Gmail,
jabber.org, talkr.im, ...)... at the moment it does not work?
The configuration between the two peers I assume needs work on both
ends I suppose, doesn't it?

Thanks again!
John


#13

Hey John,

На 08.06.11 01:17, Ryan написа:

Yes. The Gmail and GTalk clients use a variant of Jingle that we are
currently working on but that we don't support yet.

That leaves some hope :slight_smile:
Honestly I see it as vital and I can't wait to see this working. As
much as I'm prepared to tweak stuff, install work arounds... I can't
and don't wanna force my friends doing the same, including installing
a program (in this case Jitsi) as long as their are happy to do all
their communication from within Gmail, which is fair enough.

We understand this which is why we are working on the Google Jingle
support. Google are also working on implementing standard Jingle by the
way ... so regardless of whoever makes it first - there will be
interoperability eventually.

People using jabber.org or talkr.im, both of which offer JN relays,
would be able to always establish a call, no matter what kind of NAT
they are behind.

In the case of other services (including the combination of Gmail and
Jitsi) would only be able to do so if there's a possibility to establish
a direct connection between the two peers, or if they have manually
configured a TURN or a Jingle Nodes relay.

So did I get it, as long as people are logged in to Gmail/Gtalk it
doesn't matter what kind of account I use via Jitsi (Gmail,
jabber.org, talkr.im, ...)... at the moment it does not work?

Rather, as long as they are logged in _with_ the GTalk client or via the
Gmail web interface, it wouldn't work (currently).

If they are logged on Gmail using Jitsi, then it would be the same as
with any other XMPP server.

The configuration between the two peers I assume needs work on both
ends I suppose, doesn't it?

Not sure what configuration you are referring to.

Emil