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
>>
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
>>
--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
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 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
>>
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 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
>>
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.
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.
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.
(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
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
>
>
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
>
>
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
>
>