[jitsi-users] Direct SIP calling?


#1

Hello,

Is direct SIP IP calling supported in Jitsi? In other words, the
ability to make or receive calls using the SIP protocol to their IP
address.

I understand Jitsi has something called registrarless SIP support, but
it's only intended for the LAN, and there haven't been reports of it
working on the Internet. :frowning:

Gene


#2

In Internet who will handle NAT problems? Why not to use one of the free VOIP services?
For example www.sipmobile.org?
I apologize for the ad.

20 марта 2014 г., в 11:58, Gene Wu <genewu7@gmail.com> написал(а):

···

Hello,

Is direct SIP IP calling supported in Jitsi? In other words, the
ability to make or receive calls using the SIP protocol to their IP
address.

I understand Jitsi has something called registrarless SIP support, but
it's only intended for the LAN, and there haven't been reports of it
working on the Internet. :frowning:

Gene

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#3

Here's an idea for seamless entering of an ip address without significantly
changing the Jitsi code or user interface.
Define a character sequence of # and * that means to Jitsi "the following
is an IP address".

Ie, you define "***" as the IP address "escape code". And then you dial ***
200*219*32*32#
(trailing # signalling the end of the ip address)

Would it work? comments? (I remembered this because I had a SIP client a
long long time ago which allowed to direct dial an IP address if you
preceded it with an "escape sequence" (I don't remember what it was at this
point, but something like ###)

In any case, directly calling an ip address might be useful for a very
short time, until IPv4 is still used. When everyone switches to ipv6, have
you seen the lenght of a IPv6 address?. ie 2001:4860:4001:4002::14
Hard to remember, for sure.

.. but it would still be doable with the suggested method, by replacing :
with asterisks.

...Just an idea...

FC

···

On Thu, Mar 20, 2014 at 7:58 AM, Gene Wu <genewu7@gmail.com> wrote:

Is direct SIP IP calling supported in Jitsi? In other words, the
ability to make or receive calls using the SIP protocol to their IP
address.

--
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


#4

I would also like to know how to make a call without registering.
In answer to Nikita's question the peers should handle the NAT traversal
issues.

···

On Thu, Mar 20, 2014 at 7:13 AM, Nikita Stashkov <snl@sipmobile.org> wrote:

In Internet who will handle NAT problems? Why not to use one of the free
VOIP services?
For example www.sipmobile.org?
I apologize for the ad.

20 марта 2014 г., в 11:58, Gene Wu <genewu7@gmail.com> написал(а):

> Hello,
>
> Is direct SIP IP calling supported in Jitsi? In other words, the
> ability to make or receive calls using the SIP protocol to their IP
> address.
>
> I understand Jitsi has something called registrarless SIP support, but
> it's only intended for the LAN, and there haven't been reports of it
> working on the Internet. :frowning:
>
> Gene
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

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users@jitsi.org
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#5

Yes. Between two WAN IP addresses, I imagine Jitsi should be able to
call each other's IP address. But if this is not possible, it would be
unfortunate as it seems to be a basic function.

Gene

···

On Thu, Mar 20, 2014 at 7:20 AM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

I would also like to know how to make a call without registering.
In answer to Nikita's question the peers should handle the NAT traversal
issues.

On Thu, Mar 20, 2014 at 7:13 AM, Nikita Stashkov <snl@sipmobile.org> wrote:

In Internet who will handle NAT problems? Why not to use one of the free
VOIP services?
For example <XXXXXXX>?
I apologize for the ad.

20 марта 2014 г., в 11:58, Gene Wu <genewu7@gmail.com> написал(а):

> Hello,
>
> Is direct SIP IP calling supported in Jitsi? In other words, the
> ability to make or receive calls using the SIP protocol to their IP
> address.
>
> I understand Jitsi has something called registrarless SIP support, but
> it's only intended for the LAN, and there haven't been reports of it
> working on the Internet. :frowning:
>
> Gene
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users


#6

Maybe you don't want others to monitor your communication.

···

2014-03-20 12:13 GMT+01:00 Nikita Stashkov <snl@sipmobile.org>:

In Internet who will handle NAT problems? Why not to use one of the free VOIP services?
For example www.sipmobile.org?
I apologize for the ad.

20 марта 2014 г., в 11:58, Gene Wu <genewu7@gmail.com> написал(а):

Hello,

Is direct SIP IP calling supported in Jitsi? In other words, the
ability to make or receive calls using the SIP protocol to their IP
address.

I understand Jitsi has something called registrarless SIP support, but
it's only intended for the LAN, and there haven't been reports of it
working on the Internet. :frowning:

Gene

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users


#7

Is direct SIP IP calling supported in Jitsi? In other words, the
  ability to make or receive calls using the SIP protocol to their IP
  address.

Here's an idea for seamless entering of an ip address without significantly
changing the Jitsi code or user interface.
Define a character sequence of # and * that means to Jitsi "the following is
an IP address".

Just enter someone@ip, e.g. john@192.168.1.10 or jane@2001:123:456::10 (not sure right now if the brackets would be required/accepted). This assumes you're using the default port of 5060 for SIP and have it opened on your computer's firewall (e.g. on Windows, allow Jitsi.exe). But as already said multiple times, this won't help you if NAT is involved.

[...]
FC

Ingo


#8

Great! is the username before the @ irrelevant?

FC

···

On Thu, Mar 20, 2014 at 4:41 PM, Ingo Bauersachs <ingo@jitsi.org> wrote:

Just enter someone@ip, e.g. john@192.168.1.10 or jane@2001:123:456::10 (

--
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


#9

This assumes
you're using the default port of 5060 for SIP and have it opened on your
computer's firewall (e.g. on Windows, allow Jitsi.exe). But as already said
multiple times, this won't help you if NAT is involved.

The thing that nobody seems to explain is why. Why should it not work
if jitsi is configured to use only specific ports that are forwarded by the
NAT router? Maybe it just that the repeatedly given and maybe just a
little incomplete answer, does not help as much as it could.


#10

https://jitsi.org/Documentation/RegistrarlessSIPAccount[1]

Ciao,
  Matteo

···

-----------------------------------------------------------------------------
From: Fernando Cassia <fcassia@gmail.com>
Sent: giovedì 20 marzo 2014
To: Jitsi Users <users@jitsi.org>
Cc:
Subject: Re: [jitsi-users] Direct SIP calling?

On Thu, Mar 20, 2014 at 4:41 PM, Ingo Bauersachs <ingo@jitsi.org> wrote:

Just enter someone@ip, e.g. john@192.168.1.10 or jane@2001:123:456::10 (

Great! is the username before the @ irrelevant?

FC

--------
[1] https://jitsi.org/Documentation/RegistrarlessSIPAccount


#11

This assumes
you're using the default port of 5060 for SIP and have it opened on your
computer's firewall (e.g. on Windows, allow Jitsi.exe). But as already

said

multiple times, this won't help you if NAT is involved.

The thing that nobody seems to explain is why. Why should it not work
if jitsi is configured to use only specific ports that are forwarded by

the

NAT router? Maybe it just that the repeatedly given and maybe just a
little incomplete answer, does not help as much as it could.

You probably can establish a call if you forward the port 5060 on your NAT
box, but you won't have media. This is because SIP contains your local IP
address in the INVITE SDP (e.g. 192.168.1.10). The remote peer doesn't know
that this address should actually be the public IP of the NAT box. Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP. But then again, this only does the job for
one single instance behind the NAT (the one you forward the ports to).

And thanks Matteo, I forgot that David wrote a doc page for the regless
accounts.

Ingo


#12

Another method is to establish some VPN connection between two computers. So they would be in one network.
For example OpenVPN.

Regards,
Nikita Stashkov
21 марта 2014 г., в 9:40, Ingo Bauersachs <ingo@jitsi.org> написал(а):

···

This assumes
you're using the default port of 5060 for SIP and have it opened on your
computer's firewall (e.g. on Windows, allow Jitsi.exe). But as already

said

multiple times, this won't help you if NAT is involved.

The thing that nobody seems to explain is why. Why should it not work
if jitsi is configured to use only specific ports that are forwarded by

the

NAT router? Maybe it just that the repeatedly given and maybe just a
little incomplete answer, does not help as much as it could.

You probably can establish a call if you forward the port 5060 on your NAT
box, but you won't have media. This is because SIP contains your local IP
address in the INVITE SDP (e.g. 192.168.1.10). The remote peer doesn't know
that this address should actually be the public IP of the NAT box. Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP. But then again, this only does the job for
one single instance behind the NAT (the one you forward the ports to).

And thanks Matteo, I forgot that David wrote a doc page for the regless
accounts.

Ingo

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users mailing list
users@jitsi.org
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http://lists.jitsi.org/mailman/listinfo/users


#13

Thank you, that is definitely part of the complete answer!

You probably can establish a call if you forward the port 5060 on your NAT
box, but you won't have media.

I see now, https://jitsi.org/Documentation/RegistrarlessSIPAccount suggests 5061 (UDP)
is the default media port.

This is because SIP contains your local IP
address in the INVITE SDP (e.g. 192.168.1.10). The remote peer doesn't know
that this address should actually be the public IP of the NAT box.

Many thanks for explaning that jitsi still sends false (private) IP addresses
in SIP messages to public IP addresses.

What about direct "jingle" calls (instead of SIP) is that possible with ICE?

Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP. But then again, this only does the job for
one single instance behind the NAT (the one you forward the ports to).

OK, NAT will of course require to separate port forwarding for each client.


#14

Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP.

Be carefull in selecting, never to send media to all possible addresses
(the local IP may be valid on the remote side and point to somebody else)


#15

Thank you, that is definitely part of the complete answer!

You probably can establish a call if you forward the port 5060 on your

NAT

box, but you won't have media.

I see now, https://jitsi.org/Documentation/RegistrarlessSIPAccount

suggests

5061 (UDP) is the default media port.

I don't think this is correct. 5061 is used for TLS connections, even in
p2p-mode (which wouldn't work because we don't set up the listener with a
certificate). 5060 is used for SIP over UDP and TCP as long as no other
application has already claimed it (and it's adjustable in the advanced
settings).

This is because SIP contains your local IP
address in the INVITE SDP (e.g. 192.168.1.10). The remote peer doesn't

know

that this address should actually be the public IP of the NAT box.

Many thanks for explaning that jitsi still sends false (private) IP

addresses

in SIP messages to public IP addresses.

Well, they're not "false". They're the addresses Jitsi actually listens on
and the whole nature why NAT is such a bad thing.

What about direct "jingle" calls (instead of SIP) is that possible with

ICE?

I don't know of any p2p mode for XMPP, and there is definitely none in
Jitsi.

Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP. But then again, this only does the job for
one single instance behind the NAT (the one you forward the ports to).

OK, NAT will of course require to separate port forwarding for each

client.

Ingo


#16

Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP.

Be carefull in selecting, never to send media to all possible addresses
(the local IP may be valid on the remote side and point to somebody else)

Media will always be sent to only one address. ICE just figures out all
possible addresses and checks over which of these the two clients can
actually communicate with each other. This pair is then used for the media.

Ingo


#17

Hey there,

  Once ICE
is also implemented for SIP, this should work as it will include all
possible addresses in the SDP.

Be carefull in selecting, never to send media to all possible addresses
(the local IP may be valid on the remote side and point to somebody else)

You seem to be repeatedly asking the same questions and yet put very little effort in understanding the answers. This doesn't prevent you from making assertions based on false assumptions.

The ICE specification is here:

http://tools.ietf.org/html/rfc5245

I recommend you read that.

Emil

···

On 21.03.14, 10:17, ca2013@arcor.de wrote:

--
https://jitsi.org