[jitsi-dev] SIP calling on jitsi meet


#1

Hello Developers,

I'd like to know how to add sip calling to jitsi meet like seen for the VUC:
https://jitsi.vuc.me

Having this added in may help those who are trying to have "audio only" jitsi
meet conferences.

Thanks,
jungle

···

--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si


#2

That's currently a work in progress and there's a bit more left to do.

Shouldn't be long though, so stay tuned!

--sent from my mobile

···

On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:

Hello Developers,

I'd like to know how to add sip calling to jitsi meet like seen for the
VUC:
https://jitsi.vuc.me

Having this added in may help those who are trying to have "audio only"
jitsi
meet conferences.

Thanks,
jungle
--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

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#3

Dear Emil,

···

--------------------------------------------
From: Emil Ivov <emcho@jitsi.org>
Sent: Fri, 25 Jul 2014 13:24:12 -0400
To: Jitsi Developers
Subject: Re: [jitsi-dev] SIP calling on jitsi meet

That's currently a work in progress and there's a bit more left to do.

Shouldn't be long though, so stay tuned!

You and your team rock! Thanks to you and your team for working so hard to
make Jitsi and JM better every day!

--sent from my mobile

A million thanks,
j

On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:

Hello Developers,

I'd like to know how to add sip calling to jitsi meet like seen for the
VUC:
https://jitsi.vuc.me

Having this added in may help those who are trying to have "audio only"
jitsi
meet conferences.

Thanks,
jungle
--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si


#4

Thank you very much for your kind words and support! We really do
appreciate that!

--sent from my mobile

···

On 25 Jul 2014 1:28 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:

Dear Emil,
--------------------------------------------
From: Emil Ivov <emcho@jitsi.org>
Sent: Fri, 25 Jul 2014 13:24:12 -0400
To: Jitsi Developers
Subject: Re: [jitsi-dev] SIP calling on jitsi meet
>
> That's currently a work in progress and there's a bit more left to do.
>
> Shouldn't be long though, so stay tuned!

You and your team rock! Thanks to you and your team for working so hard to
make Jitsi and JM better every day!

>
> --sent from my mobile

A million thanks,
j

> On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:
>
>> Hello Developers,
>>
>> I'd like to know how to add sip calling to jitsi meet like seen for the
>> VUC:
>> https://jitsi.vuc.me
>>
>> Having this added in may help those who are trying to have "audio only"
>> jitsi
>> meet conferences.
>>
>> Thanks,
>> jungle
>> --
>> inum: 883510009027723
>> sip: jungleboogie@sip2sip.info
>> xmpp: jungle-boogie@jit.si
>>

--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

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#5

Hi All,

That's currently a work in progress and there's a bit more left to do.

Shouldn't be long though, so stay tuned!

Just curious to know how the progress is going with Jitsi meet and sip
calling. Any updates?

···

On 25 July 2014 10:24, Emil Ivov <emcho@jitsi.org> wrote:

--sent from my mobile

On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:

Hello Developers,

I'd like to know how to add sip calling to jitsi meet like seen for the
VUC:
https://jitsi.vuc.me

Having this added in may help those who are trying to have "audio only"
jitsi
meet conferences.

Thanks,
jungle
--
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

--
-------
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si


#6

Hey JB,

You can already dowoad our SIP gateway (Jigasi) and useit fir SIP calls.
Here's more on how to easily do that:

https://jitsi.org/qi

--sent from my mobile

···

On 02 Sep 2014 11:05 PM, "jungle Boogie" <jungleboogie0@gmail.com> wrote:

Hi All,
On 25 July 2014 10:24, Emil Ivov <emcho@jitsi.org> wrote:
> That's currently a work in progress and there's a bit more left to do.
>
> Shouldn't be long though, so stay tuned!

Just curious to know how the progress is going with Jitsi meet and sip
calling. Any updates?

>
> --sent from my mobile
>
> On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> wrote:
>>
>> Hello Developers,
>>
>> I'd like to know how to add sip calling to jitsi meet like seen for the
>> VUC:
>> https://jitsi.vuc.me
>>
>> Having this added in may help those who are trying to have "audio only"
>> jitsi
>> meet conferences.
>>
>> Thanks,
>> jungle
>> --
>> inum: 883510009027723
>> sip: jungleboogie@sip2sip.info
>> xmpp: jungle-boogie@jit.si
>>

--
-------
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

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#7

That sounds really interesting. Would Jitsi be a SIP client that could be called this way? Any limitations in functionality?

Cheers,
Markus

···

On 3 September 2014 22:20:56 CEST, Emil Ivov <emcho@jitsi.org> wrote:

Hey JB,

You can already dowoad our SIP gateway (Jigasi) and useit fir SIP
calls.
Here's more on how to easily do that:

https://jitsi.org/qi

--sent from my mobile
On 02 Sep 2014 11:05 PM, "jungle Boogie" <jungleboogie0@gmail.com> >wrote:

Hi All,
On 25 July 2014 10:24, Emil Ivov <emcho@jitsi.org> wrote:
> That's currently a work in progress and there's a bit more left to

do.

>
> Shouldn't be long though, so stay tuned!

Just curious to know how the progress is going with Jitsi meet and

sip

calling. Any updates?

>
> --sent from my mobile
>
> On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> >wrote:
>>
>> Hello Developers,
>>
>> I'd like to know how to add sip calling to jitsi meet like seen

for the

>> VUC:
>> https://jitsi.vuc.me
>>
>> Having this added in may help those who are trying to have "audio

only"

>> jitsi
>> meet conferences.
>>
>> Thanks,
>> jungle
>> --
>> inum: 883510009027723
>> sip: jungleboogie@sip2sip.info
>> xmpp: jungle-boogie@jit.si
>>

--
-------
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

_______________________________________________
dev mailing list
dev@jitsi.org
Unsubscribe instructions and other list options:
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------------------------------------------------------------------------

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Sent from my Android phone with K-9 Mail. Please excuse my brevity.


#8

Yes, Jigasi registers as a SIP client and can be called. It expects to find
a Jitsi-Conference-Room header in the invite with the name of the Jitsi
Meet conference the call is to be patched through to.

--sent from my mobile

···

On 04 Sep 2014 8:16 AM, "Markus Kilås" <subjunctive.post@gmail.com> wrote:

That sounds really interesting. Would Jitsi be a SIP client that could be
called this way? Any limitations in functionality?

Cheers,
Markus

On 3 September 2014 22:20:56 CEST, Emil Ivov <emcho@jitsi.org> wrote:

Hey JB,

You can already dowoad our SIP gateway (Jigasi) and useit fir SIP calls.
Here's more on how to easily do that:

https://jitsi.org/qi

--sent from my mobile
On 02 Sep 2014 11:05 PM, "jungle Boogie" <jungleboogie0@gmail.com> wrote:

Hi All,
On 25 July 2014 10:24, Emil Ivov <emcho@jitsi.org> wrote:
> That's currently a work in progress and there's a bit more left to do.
>
> Shouldn't be long though, so stay tuned!

Just curious to know how the progress is going with Jitsi meet and sip
calling. Any updates?

>
> --sent from my mobile
>
> On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> >>> wrote:
>>
>> Hello Developers,
>>
>> I'd like to know how to add sip calling to jitsi meet like seen for
the
>> VUC:
>> https://jitsi.vuc.me
>>
>> Having this added in may help those who are trying to have "audio
only"
>> jitsi
>> meet conferences.
>>
>> Thanks,
>> jungle
>> --
>> inum: 883510009027723
>> sip: jungleboogie@sip2sip.info
>> xmpp: jungle-boogie@jit.si
>>

--
-------
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

_______________________________________________
dev mailing list
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Unsubscribe instructions and other list options:
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------------------------------

dev mailing list
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--
Sent from my Android phone with K-9 Mail. Please excuse my brevity.


#9

Very interesting development. I was waiting for this to become available.
Excellent job Jitsi team.
I presume that, for now, SIP calling in Jitsi-meet only works with jigasi.
Is that correct?

If we already run our own SIP proxy or B2BUA, like kamailio or Freeswitch,
is it possible to use that instead of jigasi? Any docs out there I can
study on how to implement this functionality?

Thanks

···

On Thu, Sep 4, 2014 at 7:59 AM, Emil Ivov <emcho@jitsi.org> wrote:

Yes, Jigasi registers as a SIP client and can be called. It expects to
find a Jitsi-Conference-Room header in the invite with the name of the
Jitsi Meet conference the call is to be patched through to.

--sent from my mobile
On 04 Sep 2014 8:16 AM, "Markus Kilås" <subjunctive.post@gmail.com> wrote:

That sounds really interesting. Would Jitsi be a SIP client that could be
called this way? Any limitations in functionality?

Cheers,
Markus

On 3 September 2014 22:20:56 CEST, Emil Ivov <emcho@jitsi.org> wrote:

Hey JB,

You can already dowoad our SIP gateway (Jigasi) and useit fir SIP calls.
Here's more on how to easily do that:

https://jitsi.org/qi

--sent from my mobile
On 02 Sep 2014 11:05 PM, "jungle Boogie" <jungleboogie0@gmail.com> >>> wrote:

Hi All,
On 25 July 2014 10:24, Emil Ivov <emcho@jitsi.org> wrote:
> That's currently a work in progress and there's a bit more left to do.
>
> Shouldn't be long though, so stay tuned!

Just curious to know how the progress is going with Jitsi meet and sip
calling. Any updates?

>
> --sent from my mobile
>
> On 25 Jul 2014 1:19 PM, "Jungle Boogie" <jungleboogie0@gmail.com> >>>> wrote:
>>
>> Hello Developers,
>>
>> I'd like to know how to add sip calling to jitsi meet like seen for
the
>> VUC:
>> https://jitsi.vuc.me
>>
>> Having this added in may help those who are trying to have "audio
only"
>> jitsi
>> meet conferences.
>>
>> Thanks,
>> jungle
>> --
>> inum: 883510009027723
>> sip: jungleboogie@sip2sip.info
>> xmpp: jungle-boogie@jit.si
>>

--
-------
inum: 883510009027723
sip: jungleboogie@sip2sip.info
xmpp: jungle-boogie@jit.si

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

------------------------------

dev mailing list
dev@jitsi.org
Unsubscribe instructions and other list options:
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--
Sent from my Android phone with K-9 Mail. Please excuse my brevity.

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#10

Hi,

Very interesting development. I was waiting for this to become available.
Excellent job Jitsi team.
I presume that, for now, SIP calling in Jitsi-meet only works with jigasi.
Is that correct?

If we already run our own SIP proxy or B2BUA, like kamailio or Freeswitch,
is it possible to use that instead of jigasi? Any docs out there I can study
on how to implement this functionality?

Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
component. After Jigasi receives the IQ it joins the conference room
specified in Rayo IQ and waits for session-initiate from the focus.
Note that altough there is code for hanging up a call through Rayo it
is currently not used and handled by Jignle terminate session.

In opposite direction when Jigasi receives SIP invite it looks for
extra header mentioned by Emil and joins the room, then wait for
invite from the focus.

In your own proxy you will need to handle Jingle session with the
focus, MUC presence, do audio mixing and forward it to/from SIP peer.

Regards,
Pawel

[1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js

···

On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve <petervnv1@gmail.com> wrote:


#11

I think there's some confusion here:

Yes, you can make Jitsi Meet SIP calls through your existing
infrastructure: by making Jigasi connect to it.

Jigasi is NOT a SIP server. It is just a connector that allows SIP
servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP
signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for
them.

It does require a SIP server to then actually make the call and
especially to receive one.

Does this make more sense?

Emil

···

On Thu, Sep 4, 2014 at 2:18 PM, Paweł Domas <pawel.domas@jitsi.org> wrote:

Hi,

On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve <petervnv1@gmail.com> wrote:

Very interesting development. I was waiting for this to become available.
Excellent job Jitsi team.
I presume that, for now, SIP calling in Jitsi-meet only works with jigasi.
Is that correct?

If we already run our own SIP proxy or B2BUA, like kamailio or Freeswitch,
is it possible to use that instead of jigasi? Any docs out there I can study
on how to implement this functionality?

Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
component. After Jigasi receives the IQ it joins the conference room
specified in Rayo IQ and waits for session-initiate from the focus.
Note that altough there is code for hanging up a call through Rayo it
is currently not used and handled by Jignle terminate session.

In opposite direction when Jigasi receives SIP invite it looks for
extra header mentioned by Emil and joins the room, then wait for
invite from the focus.

In your own proxy you will need to handle Jingle session with the
focus, MUC presence, do audio mixing and forward it to/from SIP peer.

Regards,
Pawel

[1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js

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Unsubscribe instructions and other list options:
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--
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#12

(CCing dev as it seems it fell out of the address list)

Hey Peter,

Thanks Emil and Pawel.
Emil's description indeed makes more sense and was pretty much what I had
hoped for.
I'm going to take a look at the code to see if I can figure out what I need
to hack/configure to allow jigasi to connect to my SIP server instead of
using Rayo.

You don't need to worry about Rayo at all. It is part of the way Jitsi
Meet controls Jigasi but it is not exposed to the SIP infrastructure
at all

Is there any documentation available yet that I could study?

The following page tells you how to do a quick install of the entire
system: https://jitsi.org/qi

The debian package for Jigasi pops up a screen that asks you to enter
a user address (e.g. alice@example.com) and password that it would
then use to connect to the example.com SIP service. Exactly the way
the Jitsi client would. These two fields are the only configuration
you need to do in order to connect Jigasi to your infrastructure.

Emil

···

On Thu, Sep 4, 2014 at 3:55 PM, Peter Villeneuve <petervnv1@gmail.com> wrote:

Thanks

On Thu, Sep 4, 2014 at 1:36 PM, Emil Ivov <emcho@jitsi.org> wrote:

I think there's some confusion here:

Yes, you can make Jitsi Meet SIP calls through your existing
infrastructure: by making Jigasi connect to it.

Jigasi is NOT a SIP server. It is just a connector that allows SIP
servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP
signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for
them.

It does require a SIP server to then actually make the call and
especially to receive one.

Does this make more sense?

Emil

On Thu, Sep 4, 2014 at 2:18 PM, Paweł Domas <pawel.domas@jitsi.org> wrote:
> Hi,
>
> On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve <petervnv1@gmail.com> >> > wrote:
>> Very interesting development. I was waiting for this to become
>> available.
>> Excellent job Jitsi team.
>> I presume that, for now, SIP calling in Jitsi-meet only works with
>> jigasi.
>> Is that correct?
>>
>> If we already run our own SIP proxy or B2BUA, like kamailio or
>> Freeswitch,
>> is it possible to use that instead of jigasi? Any docs out there I can
>> study
>> on how to implement this functionality?
>
> Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
> starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
> component. After Jigasi receives the IQ it joins the conference room
> specified in Rayo IQ and waits for session-initiate from the focus.
> Note that altough there is code for hanging up a call through Rayo it
> is currently not used and handled by Jignle terminate session.
>
> In opposite direction when Jigasi receives SIP invite it looks for
> extra header mentioned by Emil and joins the room, then wait for
> invite from the focus.
>
> In your own proxy you will need to handle Jingle session with the
> focus, MUC presence, do audio mixing and forward it to/from SIP peer.
>
> Regards,
> Pawel
>
> [1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

--
https://jitsi.org

--
https://jitsi.org


#13

Nice. Thanks Emil.
I'll give it a try.

So to join a jitmeet conference, bob@example.com would just call
alice@example.com and would be bridged into the jitmeet conference?

···

On Thu, Sep 4, 2014 at 2:08 PM, Emil Ivov <emcho@jitsi.org> wrote:

(CCing dev as it seems it fell out of the address list)

Hey Peter,

On Thu, Sep 4, 2014 at 3:55 PM, Peter Villeneuve <petervnv1@gmail.com> > wrote:
> Thanks Emil and Pawel.
> Emil's description indeed makes more sense and was pretty much what I had
> hoped for.
> I'm going to take a look at the code to see if I can figure out what I
need
> to hack/configure to allow jigasi to connect to my SIP server instead of
> using Rayo.

You don't need to worry about Rayo at all. It is part of the way Jitsi
Meet controls Jigasi but it is not exposed to the SIP infrastructure
at all

> Is there any documentation available yet that I could study?

The following page tells you how to do a quick install of the entire
system: https://jitsi.org/qi

The debian package for Jigasi pops up a screen that asks you to enter
a user address (e.g. alice@example.com) and password that it would
then use to connect to the example.com SIP service. Exactly the way
the Jitsi client would. These two fields are the only configuration
you need to do in order to connect Jigasi to your infrastructure.

Emil

> Thanks
>
>
> On Thu, Sep 4, 2014 at 1:36 PM, Emil Ivov <emcho@jitsi.org> wrote:
>>
>> I think there's some confusion here:
>>
>> Yes, you can make Jitsi Meet SIP calls through your existing
>> infrastructure: by making Jigasi connect to it.
>>
>> Jigasi is NOT a SIP server. It is just a connector that allows SIP
>> servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP
>> signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for
>> them.
>>
>> It does require a SIP server to then actually make the call and
>> especially to receive one.
>>
>> Does this make more sense?
>>
>> Emil
>>
>> On Thu, Sep 4, 2014 at 2:18 PM, Paweł Domas <pawel.domas@jitsi.org> > wrote:
>> > Hi,
>> >
>> > On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve <petervnv1@gmail.com > > > >> > wrote:
>> >> Very interesting development. I was waiting for this to become
>> >> available.
>> >> Excellent job Jitsi team.
>> >> I presume that, for now, SIP calling in Jitsi-meet only works with
>> >> jigasi.
>> >> Is that correct?
>> >>
>> >> If we already run our own SIP proxy or B2BUA, like kamailio or
>> >> Freeswitch,
>> >> is it possible to use that instead of jigasi? Any docs out there I
can
>> >> study
>> >> on how to implement this functionality?
>> >
>> > Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
>> > starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
>> > component. After Jigasi receives the IQ it joins the conference room
>> > specified in Rayo IQ and waits for session-initiate from the focus.
>> > Note that altough there is code for hanging up a call through Rayo it
>> > is currently not used and handled by Jignle terminate session.
>> >
>> > In opposite direction when Jigasi receives SIP invite it looks for
>> > extra header mentioned by Emil and joins the room, then wait for
>> > invite from the focus.
>> >
>> > In your own proxy you will need to handle Jingle session with the
>> > focus, MUC presence, do audio mixing and forward it to/from SIP peer.
>> >
>> > Regards,
>> > Pawel
>> >
>> > [1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js
>> >
>> > _______________________________________________
>> > dev mailing list
>> > dev@jitsi.org
>> > Unsubscribe instructions and other list options:
>> > http://lists.jitsi.org/mailman/listinfo/dev
>>
>>
>>
>> --
>> https://jitsi.org
>
>

--
https://jitsi.org


#14

Nice. Thanks Emil.
I'll give it a try.

So to join a jitmeet conference, bob@example.com would just call
alice@example.com and would be bridged into the jitmeet conference?

Jigasi will register on your SIP server with some identity and it will
accept calls. Let's say that the address is jigasi@example.com.

Every time you send an INVITE to it, it will expect to find inside it
a SIP header called Jitsi-Conference-Room. That's where it understands
what room you are trying to connect to.

Exactly how you add that header (e.g. based on policy or DTMF) is
entirely up to you.

Does this make sense?

Emil

···

On Thu, Sep 4, 2014 at 4:19 PM, Peter Villeneuve <petervnv1@gmail.com> wrote:

On Thu, Sep 4, 2014 at 2:08 PM, Emil Ivov <emcho@jitsi.org> wrote:

(CCing dev as it seems it fell out of the address list)

Hey Peter,

On Thu, Sep 4, 2014 at 3:55 PM, Peter Villeneuve <petervnv1@gmail.com> >> wrote:
> Thanks Emil and Pawel.
> Emil's description indeed makes more sense and was pretty much what I
> had
> hoped for.
> I'm going to take a look at the code to see if I can figure out what I
> need
> to hack/configure to allow jigasi to connect to my SIP server instead of
> using Rayo.

You don't need to worry about Rayo at all. It is part of the way Jitsi
Meet controls Jigasi but it is not exposed to the SIP infrastructure
at all

> Is there any documentation available yet that I could study?

The following page tells you how to do a quick install of the entire
system: https://jitsi.org/qi

The debian package for Jigasi pops up a screen that asks you to enter
a user address (e.g. alice@example.com) and password that it would
then use to connect to the example.com SIP service. Exactly the way
the Jitsi client would. These two fields are the only configuration
you need to do in order to connect Jigasi to your infrastructure.

Emil

> Thanks
>
>
> On Thu, Sep 4, 2014 at 1:36 PM, Emil Ivov <emcho@jitsi.org> wrote:
>>
>> I think there's some confusion here:
>>
>> Yes, you can make Jitsi Meet SIP calls through your existing
>> infrastructure: by making Jigasi connect to it.
>>
>> Jigasi is NOT a SIP server. It is just a connector that allows SIP
>> servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP
>> signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for
>> them.
>>
>> It does require a SIP server to then actually make the call and
>> especially to receive one.
>>
>> Does this make more sense?
>>
>> Emil
>>
>> On Thu, Sep 4, 2014 at 2:18 PM, Paweł Domas <pawel.domas@jitsi.org> >> >> wrote:
>> > Hi,
>> >
>> > On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve >> >> > <petervnv1@gmail.com> >> >> > wrote:
>> >> Very interesting development. I was waiting for this to become
>> >> available.
>> >> Excellent job Jitsi team.
>> >> I presume that, for now, SIP calling in Jitsi-meet only works with
>> >> jigasi.
>> >> Is that correct?
>> >>
>> >> If we already run our own SIP proxy or B2BUA, like kamailio or
>> >> Freeswitch,
>> >> is it possible to use that instead of jigasi? Any docs out there I
>> >> can
>> >> study
>> >> on how to implement this functionality?
>> >
>> > Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
>> > starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
>> > component. After Jigasi receives the IQ it joins the conference room
>> > specified in Rayo IQ and waits for session-initiate from the focus.
>> > Note that altough there is code for hanging up a call through Rayo it
>> > is currently not used and handled by Jignle terminate session.
>> >
>> > In opposite direction when Jigasi receives SIP invite it looks for
>> > extra header mentioned by Emil and joins the room, then wait for
>> > invite from the focus.
>> >
>> > In your own proxy you will need to handle Jingle session with the
>> > focus, MUC presence, do audio mixing and forward it to/from SIP peer.
>> >
>> > Regards,
>> > Pawel
>> >
>> > [1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js
>> >
>> > _______________________________________________
>> > dev mailing list
>> > dev@jitsi.org
>> > Unsubscribe instructions and other list options:
>> > http://lists.jitsi.org/mailman/listinfo/dev
>>
>>
>>
>> --
>> https://jitsi.org
>
>

--
https://jitsi.org

--
https://jitsi.org


#15

Yes it does. Thanks

···

On Thu, Sep 4, 2014 at 2:23 PM, Emil Ivov <emcho@jitsi.org> wrote:

On Thu, Sep 4, 2014 at 4:19 PM, Peter Villeneuve <petervnv1@gmail.com> > wrote:
> Nice. Thanks Emil.
> I'll give it a try.
>
> So to join a jitmeet conference, bob@example.com would just call
> alice@example.com and would be bridged into the jitmeet conference?

Jigasi will register on your SIP server with some identity and it will
accept calls. Let's say that the address is jigasi@example.com.

Every time you send an INVITE to it, it will expect to find inside it
a SIP header called Jitsi-Conference-Room. That's where it understands
what room you are trying to connect to.

Exactly how you add that header (e.g. based on policy or DTMF) is
entirely up to you.

Does this make sense?

Emil

>
> On Thu, Sep 4, 2014 at 2:08 PM, Emil Ivov <emcho@jitsi.org> wrote:
>>
>> (CCing dev as it seems it fell out of the address list)
>>
>> Hey Peter,
>>
>> On Thu, Sep 4, 2014 at 3:55 PM, Peter Villeneuve <petervnv1@gmail.com> > >> wrote:
>> > Thanks Emil and Pawel.
>> > Emil's description indeed makes more sense and was pretty much what I
>> > had
>> > hoped for.
>> > I'm going to take a look at the code to see if I can figure out what I
>> > need
>> > to hack/configure to allow jigasi to connect to my SIP server instead
of
>> > using Rayo.
>>
>> You don't need to worry about Rayo at all. It is part of the way Jitsi
>> Meet controls Jigasi but it is not exposed to the SIP infrastructure
>> at all
>>
>> > Is there any documentation available yet that I could study?
>>
>> The following page tells you how to do a quick install of the entire
>> system: https://jitsi.org/qi
>>
>> The debian package for Jigasi pops up a screen that asks you to enter
>> a user address (e.g. alice@example.com) and password that it would
>> then use to connect to the example.com SIP service. Exactly the way
>> the Jitsi client would. These two fields are the only configuration
>> you need to do in order to connect Jigasi to your infrastructure.
>>
>> Emil
>>
>> > Thanks
>> >
>> >
>> > On Thu, Sep 4, 2014 at 1:36 PM, Emil Ivov <emcho@jitsi.org> wrote:
>> >>
>> >> I think there's some confusion here:
>> >>
>> >> Yes, you can make Jitsi Meet SIP calls through your existing
>> >> infrastructure: by making Jigasi connect to it.
>> >>
>> >> Jigasi is NOT a SIP server. It is just a connector that allows SIP
>> >> servers and B2BUAs to connect to Jitsi Meet. It handles the XMPP
>> >> signalling, ICE, DTLS/SRTP termination and multiple-SSRC handling for
>> >> them.
>> >>
>> >> It does require a SIP server to then actually make the call and
>> >> especially to receive one.
>> >>
>> >> Does this make more sense?
>> >>
>> >> Emil
>> >>
>> >> On Thu, Sep 4, 2014 at 2:18 PM, Paweł Domas <pawel.domas@jitsi.org> > >> >> wrote:
>> >> > Hi,
>> >> >
>> >> > On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve > >> >> > <petervnv1@gmail.com> > >> >> > wrote:
>> >> >> Very interesting development. I was waiting for this to become
>> >> >> available.
>> >> >> Excellent job Jitsi team.
>> >> >> I presume that, for now, SIP calling in Jitsi-meet only works with
>> >> >> jigasi.
>> >> >> Is that correct?
>> >> >>
>> >> >> If we already run our own SIP proxy or B2BUA, like kamailio or
>> >> >> Freeswitch,
>> >> >> is it possible to use that instead of jigasi? Any docs out there I
>> >> >> can
>> >> >> study
>> >> >> on how to implement this functionality?
>> >> >
>> >> > Jitsi Meet JavaSscript app is using part of Rayo signaling[1] for
>> >> > starting a call. It sends dial IQ to "config.hosts.call_control"
XMPP
>> >> > component. After Jigasi receives the IQ it joins the conference
room
>> >> > specified in Rayo IQ and waits for session-initiate from the focus.
>> >> > Note that altough there is code for hanging up a call through Rayo
it
>> >> > is currently not used and handled by Jignle terminate session.
>> >> >
>> >> > In opposite direction when Jigasi receives SIP invite it looks for
>> >> > extra header mentioned by Emil and joins the room, then wait for
>> >> > invite from the focus.
>> >> >
>> >> > In your own proxy you will need to handle Jingle session with the
>> >> > focus, MUC presence, do audio mixing and forward it to/from SIP
peer.
>> >> >
>> >> > Regards,
>> >> > Pawel
>> >> >
>> >> > [1]: https://github.com/jitsi/jitsi-meet/blob/master/libs/rayo.js
>> >> >
>> >> > _______________________________________________
>> >> > dev mailing list
>> >> > dev@jitsi.org
>> >> > Unsubscribe instructions and other list options:
>> >> > http://lists.jitsi.org/mailman/listinfo/dev
>> >>
>> >>
>> >>
>> >> --
>> >> https://jitsi.org
>> >
>> >
>>
>>
>>
>> --
>> https://jitsi.org
>
>

--
https://jitsi.org