[Free-RTC] Introduction and request for help

[Emil Ivov]

On Sat, Sep 21, 2013 at 9:54 PM, Bas Wijnen <wijnen at debian.org> wrote:
> On Sat, Sep 21, 2013 at 10:25:44AM +0300, Emil Ivov wrote:
>> On 21 Sep 2013 02:22, "Bas Wijnen" <wijnen at debian.org> wrote:
>>
>> > Jitsi (SIP and XMPP): Jitsi is written
>> > in Java, which seems to be a
>> > problem on its own.
>>
>> Well, I am afraid there's no changing this, so if you have an aversion to
>> the language, you would simply need to use something else.
>
> I didn't mean to suggest it should be changed.  I was just getting frustrated
> that nothing worked, and a program which throws errors before it even started
> isn't helping. ;-)

Exceptions printed by Jitsi on stderr are often mostly meant as debug
output and you should really worry about them unless you know exactly
what they mean.

> On Sat, Sep 21, 2013 at 09:44:30AM +0200, Daniel Pocock wrote:
>> Java can also be a good thing, as it means users on Linux, Mac and
>> Windows all get the same code and it is easier for the developers to
>> make sure that every release is interoperable across all platforms.
>>
>> For communications software, interoperability is crucial and Java helps
>> with that.
>
> Which is entirely true.  But it would be good if it would indeed work on all
> platforms.  Random crashes are extremely frustrating, especially after trying
> so many things for so long, to achieve something which really should be simple.

It does work for many people.

> I mean, what is the big problem?

There is no one big problem. It's always about the specifics of the deployment.

> I have two computers I control.  I can
> connect() from one to the other and have a working network connection.  I have
> working audio and video hardware on both sides.  How hard can it be to set up a
> link to make us see and hear each other?

OK. If we had to really quote one big problem then I suppose it would
be "diversity". Diversity in hardware, diversity of OSes and OS
flavors, diversity of audio/video capture/render libs, diversity in
devices and of course, diversity in network topologies.

Note that RTC applications are not generally written for the use case
you describe: two machines with direct IP connectivity. This is simply
one of the multitude of cases where RTC has to work.

I haven't seen you describing a specific problem (other than the
language of the application and the fact that you find standard output
to be unsettling) so it's hard for me to be any more specific on your
situation.

>> Since upgrading to wheezy, I can't communicate reliably with Empathy.
>> This is a big no-no for any communications product, once people lose
>> faith in it, it is very hard to recover
>
> In their favor, there aren't any working alternatives.  Which isn't a good
> thing, of course.

It also isn't true.

>> > Empathy (SIP): SIP connections just don't work with masquerading, it seems.
>> > I've set up an asterisk server, which works as long as no masqueraded hosts are
>> > using it.  The masqueraded hosts can call the server, get automated answers,
>> > but cannot be connected to other users (which doesn't make much sense to me,
>> > because asterisk would pass on all the traffic, so I don't see the difference
>> > between an automated answer generated by asterisk itself and a real person
>> > answering).  Anyway, it didn't work well.  Also, Asterisk seems to have very
>> > limited support for video, but I didn't really get around to trying it out.
>>
>> I don't know if they support ICE with SIP
>
> Asterisk does;

No, I don't think that's true either. At least not in the way Daniel
meant it. Asterisk is a Back-to-Back User Agent (B2BUA). It is meant
to be an endpoint. As such it is not really meant to let endpoints
talk directly to each other. Asterisk being in the middle of all
communication is its default mode of operation.

As a result, using ICE to establish a direct connection between two
asterisk connected clients is, as far as I know, not possible.

> it just ignores the reported return address and uses
> getpeername() instead.  And it works, too; it's used when making a call to an
> auto-responder, and that works.
>
>> > but it does seem to work.  Then again, it crashes at random times.  It seems I
>> > need to disable several settings to make sure it doesn't contact Google; I
>>
>> Can you raise a bug against the package listing those things you had to
>> disable?
>
> Sure, but perhaps I should first ask if I am right.  In my XMPP account
> settings, "Enable Google Contacts search" was on.

This option is only used in case you actually have a Google account.
It has no impact otherwise.

> I originally also disabled
> "Use Google's Jingle"

This wouldn't hurt. Although, once again, Jitsi only uses this variant
of the Jingle protocol in case it detects that you are using a Google
account.

> and "Use JingleNodes",

I am interested to know why you chose to do this.

Jingle Nodes is one of the technologies that Jitsi uses for NAT
traversal. Disabling it only increases your chances of  not being able
to establish a connection.

> but now I see it again, I suppose
> that's just technology designed by Google, not actually contacting their
> servers.  Or is it?
>
>> > really don't like that.  As for how it works: so far it doesn't really (or at
>> > least not with the masquerated hosts), but perhaps I have to try harder.
>>
>> Did you put in the TURN server settings?
>
> I did, but on localhost it doesn't care

Who is it?

> (it works, even if the password is
> wrong; of course on localhost I don't need the TURN server anyway, but I had
> expected an error nonetheless) and from the masqueraded host it says the
> password is wrong even though it isn't.

I seem to be missing a part of the thread here but, if you are using a
TURN server then the client needs to be able to contact it *before* it
makes a call.

>> Make sure ejabberd works well for chat first
>>
>> Make sure you have the SRV records, e.g. for pocock.com.au:
>>
>> _xmpp-client._tcp       IN      SRV     5 0 5222 jabber.trendhosting.net.
>> _xmpp-server._tcp       IN      SRV     5 0 5269 jabber.trendhosting.net.
>
> Wait, what?  Do I need to change my DNS records just to start a one-on-one
> video call with a computer that can already contact me?  I can do it; I happen
> to have control over my DNS, but I really wouldn't expect to need it.

Most of the time things would work even if you only had an A record,
but yes, you do need to control your DNS.

Also, if all you need is for two IP-s to talk to each other, then you
can simply take VLC and stream your microphone and webcam in each
direction. That's a solution specific to your use case.

VoIP isn't.

Protocols like XMPP and SIP and their accompanying methodologies are
meant to cover the Internet as a whole. To do that you need to have
servers at least at the start of your conversation.

>> From Debian, you can use CaCert.org certificates and the clients will
>> trust them.
>
> Yes, I'm using those already.  Unfortunately Ubuntu, Mac and Windows aren't
> accepting them. :-(
>
>> Multi-domain and IPv6 are slightly tricky, let me know if you need
>> examples to make them work.  I also have it linked to LDAP.
>
> No, I don't need anything complex, thanks.
>
>> Skip Asterisk for now, it is overkill, and as you point out, no video.
>>
>> It is very good for building a business phone system though
>
> Yes, it looks like it, and it works well so far, except for NAT traversal.

It actually does handle NAT traversal quite reliably (although not in
a very bandwidth efficient way due to its always-relay-everything
model of operation that I mentioned above).

> And
> its documentation is fantastic (except that its focus is not where I needed it).
>
>> The config files are fully commented with instructions, but that was
>> only done after the wheezy freeze began.
>>
>> The release team have so far not approved further uploads
>> http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=717420
>>
>> but upstream and the package maintainer (myself) are quite happy to
>> upload those changes when authorised to do so.
>
> While I understand your reasoning, the usual way in Debian is not to change
> anything once it is released, except for security and really urgent things
> (virus definition updates; timezone changes).  If you have a much better
> version of the package, it might be considered for the next point release, but
> not for stable itself.  The downside of this method is that stable is always
> outdated (you're left with the state when it was frozen).  The upside is that
> it means things are indeed really stable (new "fixes" often introduce new bugs
> as well).
>
> Anyway, to get back on topic, all this doesn't matter for me, as I'm running
> the TURN server on unstable, so I have version 1.8.13-1.
>
> I see you have uploaded 1.9.0~beta1-1 to experimental, at high urgency, with no
> information in the changelog why this was so urgent.  Is there a big difference
> with 1.8.13?  Why didn't you upload to unstable?
>
>> You can view the commented version of the file here:
>>
>> https://github.com/resiprocate/resiprocate/blob/master/reTurn/reTurnServer.config
>
> The comments didn't make it to unstable, indeed.  Thanks.
>
>> Note that trunk has slightly changed, the version in wheezy does not
>> recognise the option "UserDatabaseFile = users.txt" and you need to keep
>> using a single set of credentials for all users, by specifying something
>> like this in reTurnServer.config:
>>
>> LongTermAuthUsername = test
>> LongTermAuthPassword = password
>
> It seems to be the same in sid, looking at the config file?
>
>> I've been getting more recent versions of packages into jessie and I am
>> very hopeful that with feedback from people like yourself we will be
>> able to fine tune them further to "just work" and jessie will make a big
>> impact in this space.
>
> I would definitely like that.  My main goal would be that two people, where at
> least one of them is reachable on a public IP (which may not be the IP the host
> thinks it has), can make an encrypted video call by:
> - both installing a package
> - one person filling in the IP (and possibly the name) of the other person
> and nothing more.  They should not need to set up any server, and they
> certainly shouldn't need to edit DNS entries.

You might want to download and try Openfire (although, note that it is
also written in Java, which may be a problem for you). It comes with
an easy to install JingleNodes plugin and, together with Jitsi, will
give you what you are looking for.

You can also try using http://jit.si to get a feel for how all this works.

Cheers,
Emil

>> Please put specific comments about the problems into the bug tracker,
>> that definitely helps us too.
>
> I'll do that.
>
> Thanks,
> Bas
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-- 
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho at jitsi.org                        PHONE: +33.1.77.62.43.30
https://jitsi.org                       FAX:   +33.1.77.62.47.31