[jitsi-dev] Questions on Jigasi multiple SIP call support


#1

Hi all,

I have some questions about Jigasi multiple SIP call support,

Can focus in the conference invite multiple SIP clients to this conference?
Does Jigasi mix the audio? What is the limitation about the number of
concurrent SIP calls for every conference?

Per How it works on https://github.com/jitsi/jigasi

"It handles the XMPP signalling, ICE, DTLS/SRTP termination and
multiple-SSRC handling for them."

Can you provide more details about the multiple-SSRC handling? Does Jigasi
mix the audio from webrtc participants? My understanding is that in the
conference where all parties are webrtc participants, the browser itself
mixes the audio.

Thanks for the help,

/Kaiduan


#2

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

···

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Hi all,

I have some questions about Jigasi multiple SIP call support,

Can focus in the conference invite multiple SIP clients to this conference?
Does Jigasi mix the audio? What is the limitation about the number of
concurrent SIP calls for every conference?

Per How it works on https://github.com/jitsi/jigasi

"It handles the XMPP signalling, ICE, DTLS/SRTP termination and
multiple-SSRC handling for them."

Can you provide more details about the multiple-SSRC handling? Does Jigasi
mix the audio from webrtc participants? My understanding is that in the
conference where all parties are webrtc participants, the browser itself
mixes the audio.

Thanks for the help,

/Kaiduan

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#3

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from different

SIP clients in the case where Jigasi calls out multiple SIP clients?

Thanks,

/Kaiduan

···

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
conference?
> Does Jigasi mix the audio? What is the limitation about the number of
> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling? Does
Jigasi
> mix the audio from webrtc participants? My understanding is that in the
> conference where all parties are webrtc participants, the browser itself
> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

_______________________________________________
dev mailing list
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#4

I did one test where jigasi called two SIP clients registered to Freeswitch
in a webrtc conference, the webrtc clients and SIP can hear each other
without issues.

So it looks like Jigasi also mixes audio streams from SIP world.

Regards,

/Kaiduan

···

On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from different
SIP clients in the case where Jigasi calls out multiple SIP clients?

Thanks,

/Kaiduan

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> > wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
conference?
> Does Jigasi mix the audio? What is the limitation about the number of
> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling? Does
Jigasi
> mix the audio from webrtc participants? My understanding is that in the
> conference where all parties are webrtc participants, the browser itself
> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> 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
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#5

Hi Kaiduan,

The test you did by making jigasi to call two SIP clients is very interesting for me.

Could you help me to reproduce such test at my side?

I would be very thankful if you could describe the steps briefly.

Regards,
Arthur Petrosyan

···

On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie <kaiduanx@gmail.com> wrote:

I did one test where jigasi called two SIP clients registered to
Freeswitch
in a webrtc conference, the webrtc clients and SIP can hear each other
without issues.

So it looks like Jigasi also mixes audio streams from SIP world.

Regards,

/Kaiduan

On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> >wrote:

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from

different

SIP clients in the case where Jigasi calls out multiple SIP clients?

Thanks,

/Kaiduan

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> >> wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have

at

the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio

streams

to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams

to

voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> >wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
conference?
> Does Jigasi mix the audio? What is the limitation about the number

of

> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling?

Does

Jigasi
> mix the audio from webrtc participants? My understanding is that

in the

> conference where all parties are webrtc participants, the browser

itself

> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> 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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#6

Arthur,

The test is described as below,

1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001 SIP
accounts respectively.

2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu, Jigasi
registers with 1004 SIP account to Freeswitch,

3. Two parties join the conference from two Chrome browsers.

4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.

5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.

6. Two webrtc participants and two SIP phone are in the conference.

/Kaiduan

···

On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> wrote:

Hi Kaiduan,

The test you did by making jigasi to call two SIP clients is very
interesting for me.

Could you help me to reproduce such test at my side?

I would be very thankful if you could describe the steps briefly.

Regards,
Arthur Petrosyan

On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie <kaiduanx@gmail.com> > wrote:

I did one test where jigasi called two SIP clients registered to
Freeswitch in a webrtc conference, the webrtc clients and SIP can hear each
other without issues.

So it looks like Jigasi also mixes audio streams from SIP world.

Regards,

/Kaiduan

On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from different
SIP clients in the case where Jigasi calls out multiple SIP clients?

Thanks,

/Kaiduan

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> >>> wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> >>>> wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
conference?
> Does Jigasi mix the audio? What is the limitation about the number of
> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling? Does
Jigasi
> mix the audio from webrtc participants? My understanding is that in
the
> conference where all parties are webrtc participants, the browser
itself
> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> 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:
http://lists.jitsi.org/mailman/listinfo/dev


#7

Damencho,

Can you kindly point out the source code where Jigasi mixes the audio
streams coming from the jitsi-meet conference and sends one audio stream to
the sip side?

Thanks,

/Kaiduan

···

On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Arthur,

The test is described as below,

1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001 SIP
accounts respectively.

2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu, Jigasi
registers with 1004 SIP account to Freeswitch,

3. Two parties join the conference from two Chrome browsers.

4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.

5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.

6. Two webrtc participants and two SIP phone are in the conference.

/Kaiduan

On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> wrote:

Hi Kaiduan,

The test you did by making jigasi to call two SIP clients is very
interesting for me.

Could you help me to reproduce such test at my side?

I would be very thankful if you could describe the steps briefly.

Regards,
Arthur Petrosyan

On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie <kaiduanx@gmail.com> >> wrote:

I did one test where jigasi called two SIP clients registered to
Freeswitch in a webrtc conference, the webrtc clients and SIP can hear each
other without issues.

So it looks like Jigasi also mixes audio streams from SIP world.

Regards,

/Kaiduan

On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> >>> wrote:

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from
different SIP clients in the case where Jigasi calls out multiple SIP
clients?

Thanks,

/Kaiduan

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> >>>> wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> >>>>> wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
conference?
> Does Jigasi mix the audio? What is the limitation about the number of
> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling? Does
Jigasi
> mix the audio from webrtc participants? My understanding is that in
the
> conference where all parties are webrtc participants, the browser
itself
> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

_______________________________________________
dev mailing list
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#8

Hi,

Here by default it creates AudioMixer for the audio:
https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L312-L312

If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a translator:
https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L385-L385
And it just forwards RTP, the same way jitsi-videobridge uses libjitsi.

Regards
damencho

···

On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Damencho,

Can you kindly point out the source code where Jigasi mixes the audio
streams coming from the jitsi-meet conference and sends one audio stream to
the sip side?

Thanks,

/Kaiduan

On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Arthur,

The test is described as below,

1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001 SIP
accounts respectively.

2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu, Jigasi
registers with 1004 SIP account to Freeswitch,

3. Two parties join the conference from two Chrome browsers.

4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.

5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.

6. Two webrtc participants and two SIP phone are in the conference.

/Kaiduan

On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> wrote:

Hi Kaiduan,

The test you did by making jigasi to call two SIP clients is very
interesting for me.

Could you help me to reproduce such test at my side?

I would be very thankful if you could describe the steps briefly.

Regards,
Arthur Petrosyan

On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie <kaiduanx@gmail.com> >>> wrote:

I did one test where jigasi called two SIP clients registered to
Freeswitch in a webrtc conference, the webrtc clients and SIP can hear each
other without issues.

So it looks like Jigasi also mixes audio streams from SIP world.

Regards,

/Kaiduan

On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> >>>> wrote:

Damencho,

Thanks for the swift response.

So Jigasi does mix audio from the webrtc side.

From SIP side, does Jigasi mix the multiple audio streams from
different SIP clients in the case where Jigasi calls out multiple SIP
clients?

Thanks,

/Kaiduan

On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> >>>>> wrote:

Hi,

By default Jigasi mixes the audio streams coming from the jitsi-meet
conference and sends one audio stream to the sip side.
When mixing audio this is the limitation that jigasi can handle a
number of conferences till it hits a CPU limitation, we do not have at
the moment any estimations and it depends on the resources of the
machine running the jigasi server.

There is an option to bypass this and just send multiple audio streams
to the sip side, but there is no legacy sip server(asterisk or
freeswitch) at the moment that handles that. We deployed jigasi
servers to meet.jit.si to call in, which are forwarding all streams to
voximplant (the provider we use for those jigasi servers there) and
they handle that.

Hope this clears things a bit and helps.

Regards
damencho

On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> >>>>>> wrote:
> Hi all,
>
> I have some questions about Jigasi multiple SIP call support,
>
> Can focus in the conference invite multiple SIP clients to this
> conference?
> Does Jigasi mix the audio? What is the limitation about the number
> of
> concurrent SIP calls for every conference?
>
> Per How it works on https://github.com/jitsi/jigasi
>
> "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
> multiple-SSRC handling for them."
>
> Can you provide more details about the multiple-SSRC handling? Does
> Jigasi
> mix the audio from webrtc participants? My understanding is that in
> the
> conference where all parties are webrtc participants, the browser
> itself
> mixes the audio.
>
> Thanks for the help,
>
> /Kaiduan
>
> _______________________________________________
> 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:
http://lists.jitsi.org/mailman/listinfo/dev

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#9

Thanks Damencho.

So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix audio
from jitsi-meet conference, right?

/Kaiduan

···

On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

Here by default it creates AudioMixer for the audio:
https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L312-L312

If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
translator:
https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L385-L385
And it just forwards RTP, the same way jitsi-videobridge uses libjitsi.

Regards
damencho

On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Damencho,
>
> Can you kindly point out the source code where Jigasi mixes the audio
> streams coming from the jitsi-meet conference and sends one audio stream
to
> the sip side?
>
> Thanks,
>
> /Kaiduan
>
> On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> > wrote:
>>
>> Arthur,
>>
>> The test is described as below,
>>
>> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001 SIP
>> accounts respectively.
>>
>> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu,
Jigasi
>> registers with 1004 SIP account to Freeswitch,
>>
>> 3. Two parties join the conference from two Chrome browsers.
>>
>> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.
>>
>> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.
>>
>> 6. Two webrtc participants and two SIP phone are in the conference.
>>
>> /Kaiduan
>>
>>
>>
>> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> > wrote:
>>>
>>> Hi Kaiduan,
>>>
>>> The test you did by making jigasi to call two SIP clients is very
>>> interesting for me.
>>>
>>> Could you help me to reproduce such test at my side?
>>>
>>> I would be very thankful if you could describe the steps briefly.
>>>
>>> Regards,
>>> Arthur Petrosyan
>>>
>>>
>>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie < > kaiduanx@gmail.com> > >>> wrote:
>>>>
>>>> I did one test where jigasi called two SIP clients registered to
>>>> Freeswitch in a webrtc conference, the webrtc clients and SIP can
hear each
>>>> other without issues.
>>>>
>>>> So it looks like Jigasi also mixes audio streams from SIP world.
>>>>
>>>> Regards,
>>>>
>>>> /Kaiduan
>>>>
>>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> > >>>> wrote:
>>>>>
>>>>> Damencho,
>>>>>
>>>>> Thanks for the swift response.
>>>>>
>>>>> So Jigasi does mix audio from the webrtc side.
>>>>>
>>>>> From SIP side, does Jigasi mix the multiple audio streams from
>>>>> different SIP clients in the case where Jigasi calls out multiple SIP
>>>>> clients?
>>>>>
>>>>> Thanks,
>>>>>
>>>>> /Kaiduan
>>>>>
>>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> > >>>>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> By default Jigasi mixes the audio streams coming from the jitsi-meet
>>>>>> conference and sends one audio stream to the sip side.
>>>>>> When mixing audio this is the limitation that jigasi can handle a
>>>>>> number of conferences till it hits a CPU limitation, we do not have
at
>>>>>> the moment any estimations and it depends on the resources of the
>>>>>> machine running the jigasi server.
>>>>>>
>>>>>> There is an option to bypass this and just send multiple audio
streams
>>>>>> to the sip side, but there is no legacy sip server(asterisk or
>>>>>> freeswitch) at the moment that handles that. We deployed jigasi
>>>>>> servers to meet.jit.si to call in, which are forwarding all
streams to
>>>>>> voximplant (the provider we use for those jigasi servers there) and
>>>>>> they handle that.
>>>>>>
>>>>>> Hope this clears things a bit and helps.
>>>>>>
>>>>>> Regards
>>>>>> damencho
>>>>>>
>>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> > >>>>>> wrote:
>>>>>> > Hi all,
>>>>>> >
>>>>>> > I have some questions about Jigasi multiple SIP call support,
>>>>>> >
>>>>>> > Can focus in the conference invite multiple SIP clients to this
>>>>>> > conference?
>>>>>> > Does Jigasi mix the audio? What is the limitation about the number
>>>>>> > of
>>>>>> > concurrent SIP calls for every conference?
>>>>>> >
>>>>>> > Per How it works on https://github.com/jitsi/jigasi
>>>>>> >
>>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
>>>>>> > multiple-SSRC handling for them."
>>>>>> >
>>>>>> > Can you provide more details about the multiple-SSRC handling?
Does
>>>>>> > Jigasi
>>>>>> > mix the audio from webrtc participants? My understanding is that
in
>>>>>> > the
>>>>>> > conference where all parties are webrtc participants, the browser
>>>>>> > itself
>>>>>> > mixes the audio.
>>>>>> >
>>>>>> > Thanks for the help,
>>>>>> >
>>>>>> > /Kaiduan
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > 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:
>>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>>>>>
>>>>>
>>>>
>>
>
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

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#10

Yep that is correct, but I'm not sure there is some open source sip
pbx supporting multi-streams.

···

On Thu, Mar 23, 2017 at 12:47 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Thanks Damencho.

So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix audio
from jitsi-meet conference, right?

/Kaiduan

On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

Here by default it creates AudioMixer for the audio:

https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L312-L312

If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
translator:

https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L385-L385
And it just forwards RTP, the same way jitsi-videobridge uses libjitsi.

Regards
damencho

On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Damencho,
>
> Can you kindly point out the source code where Jigasi mixes the audio
> streams coming from the jitsi-meet conference and sends one audio stream
> to
> the sip side?
>
> Thanks,
>
> /Kaiduan
>
> On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> >> > wrote:
>>
>> Arthur,
>>
>> The test is described as below,
>>
>> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001 SIP
>> accounts respectively.
>>
>> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu,
>> Jigasi
>> registers with 1004 SIP account to Freeswitch,
>>
>> 3. Two parties join the conference from two Chrome browsers.
>>
>> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.
>>
>> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.
>>
>> 6. Two webrtc participants and two SIP phone are in the conference.
>>
>> /Kaiduan
>>
>>
>>
>> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> >> >> wrote:
>>>
>>> Hi Kaiduan,
>>>
>>> The test you did by making jigasi to call two SIP clients is very
>>> interesting for me.
>>>
>>> Could you help me to reproduce such test at my side?
>>>
>>> I would be very thankful if you could describe the steps briefly.
>>>
>>> Regards,
>>> Arthur Petrosyan
>>>
>>>
>>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie >> >>> <kaiduanx@gmail.com> >> >>> wrote:
>>>>
>>>> I did one test where jigasi called two SIP clients registered to
>>>> Freeswitch in a webrtc conference, the webrtc clients and SIP can
>>>> hear each
>>>> other without issues.
>>>>
>>>> So it looks like Jigasi also mixes audio streams from SIP world.
>>>>
>>>> Regards,
>>>>
>>>> /Kaiduan
>>>>
>>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> >> >>>> wrote:
>>>>>
>>>>> Damencho,
>>>>>
>>>>> Thanks for the swift response.
>>>>>
>>>>> So Jigasi does mix audio from the webrtc side.
>>>>>
>>>>> From SIP side, does Jigasi mix the multiple audio streams from
>>>>> different SIP clients in the case where Jigasi calls out multiple
>>>>> SIP
>>>>> clients?
>>>>>
>>>>> Thanks,
>>>>>
>>>>> /Kaiduan
>>>>>
>>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov <damencho@jitsi.org> >> >>>>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> By default Jigasi mixes the audio streams coming from the
>>>>>> jitsi-meet
>>>>>> conference and sends one audio stream to the sip side.
>>>>>> When mixing audio this is the limitation that jigasi can handle a
>>>>>> number of conferences till it hits a CPU limitation, we do not have
>>>>>> at
>>>>>> the moment any estimations and it depends on the resources of the
>>>>>> machine running the jigasi server.
>>>>>>
>>>>>> There is an option to bypass this and just send multiple audio
>>>>>> streams
>>>>>> to the sip side, but there is no legacy sip server(asterisk or
>>>>>> freeswitch) at the moment that handles that. We deployed jigasi
>>>>>> servers to meet.jit.si to call in, which are forwarding all streams
>>>>>> to
>>>>>> voximplant (the provider we use for those jigasi servers there) and
>>>>>> they handle that.
>>>>>>
>>>>>> Hope this clears things a bit and helps.
>>>>>>
>>>>>> Regards
>>>>>> damencho
>>>>>>
>>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com> >> >>>>>> wrote:
>>>>>> > Hi all,
>>>>>> >
>>>>>> > I have some questions about Jigasi multiple SIP call support,
>>>>>> >
>>>>>> > Can focus in the conference invite multiple SIP clients to this
>>>>>> > conference?
>>>>>> > Does Jigasi mix the audio? What is the limitation about the
>>>>>> > number
>>>>>> > of
>>>>>> > concurrent SIP calls for every conference?
>>>>>> >
>>>>>> > Per How it works on https://github.com/jitsi/jigasi
>>>>>> >
>>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
>>>>>> > multiple-SSRC handling for them."
>>>>>> >
>>>>>> > Can you provide more details about the multiple-SSRC handling?
>>>>>> > Does
>>>>>> > Jigasi
>>>>>> > mix the audio from webrtc participants? My understanding is that
>>>>>> > in
>>>>>> > the
>>>>>> > conference where all parties are webrtc participants, the browser
>>>>>> > itself
>>>>>> > mixes the audio.
>>>>>> >
>>>>>> > Thanks for the help,
>>>>>> >
>>>>>> > /Kaiduan
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > 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:
>>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>>>>>
>>>>>
>>>>
>>
>
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

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#11

Damencho,

Can you explain how the media is forwarded between SIP and jitsi-meet in
Jigasi? The signalling handling between SIP and jitsi-meet is obvious from
reading the code, but the media processing is not that clear.

Thanks,

/Kaiduan

···

On Thu, Mar 23, 2017 at 1:51 PM, Damian Minkov <damencho@jitsi.org> wrote:

Yep that is correct, but I'm not sure there is some open source sip
pbx supporting multi-streams.

On Thu, Mar 23, 2017 at 12:47 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Thanks Damencho.
>
> So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix audio
> from jitsi-meet conference, right?
>
> /Kaiduan
>
> On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> > wrote:
>>
>> Hi,
>>
>> Here by default it creates AudioMixer for the audio:
>>
>> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L312-L312
>>
>> If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
>> translator:
>>
>> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L385-L385
>> And it just forwards RTP, the same way jitsi-videobridge uses libjitsi.
>>
>> Regards
>> damencho
>>
>>
>> On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> > wrote:
>> > Damencho,
>> >
>> > Can you kindly point out the source code where Jigasi mixes the audio
>> > streams coming from the jitsi-meet conference and sends one audio
stream
>> > to
>> > the sip side?
>> >
>> > Thanks,
>> >
>> > /Kaiduan
>> >
>> > On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> > >> > wrote:
>> >>
>> >> Arthur,
>> >>
>> >> The test is described as below,
>> >>
>> >> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001
SIP
>> >> accounts respectively.
>> >>
>> >> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu,
>> >> Jigasi
>> >> registers with 1004 SIP account to Freeswitch,
>> >>
>> >> 3. Two parties join the conference from two Chrome browsers.
>> >>
>> >> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.
>> >>
>> >> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.
>> >>
>> >> 6. Two webrtc participants and two SIP phone are in the conference.
>> >>
>> >> /Kaiduan
>> >>
>> >>
>> >>
>> >> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> > >> >> wrote:
>> >>>
>> >>> Hi Kaiduan,
>> >>>
>> >>> The test you did by making jigasi to call two SIP clients is very
>> >>> interesting for me.
>> >>>
>> >>> Could you help me to reproduce such test at my side?
>> >>>
>> >>> I would be very thankful if you could describe the steps briefly.
>> >>>
>> >>> Regards,
>> >>> Arthur Petrosyan
>> >>>
>> >>>
>> >>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie > >> >>> <kaiduanx@gmail.com> > >> >>> wrote:
>> >>>>
>> >>>> I did one test where jigasi called two SIP clients registered to
>> >>>> Freeswitch in a webrtc conference, the webrtc clients and SIP can
>> >>>> hear each
>> >>>> other without issues.
>> >>>>
>> >>>> So it looks like Jigasi also mixes audio streams from SIP world.
>> >>>>
>> >>>> Regards,
>> >>>>
>> >>>> /Kaiduan
>> >>>>
>> >>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> > >> >>>> wrote:
>> >>>>>
>> >>>>> Damencho,
>> >>>>>
>> >>>>> Thanks for the swift response.
>> >>>>>
>> >>>>> So Jigasi does mix audio from the webrtc side.
>> >>>>>
>> >>>>> From SIP side, does Jigasi mix the multiple audio streams from
>> >>>>> different SIP clients in the case where Jigasi calls out multiple
>> >>>>> SIP
>> >>>>> clients?
>> >>>>>
>> >>>>> Thanks,
>> >>>>>
>> >>>>> /Kaiduan
>> >>>>>
>> >>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov < > damencho@jitsi.org> > >> >>>>> wrote:
>> >>>>>>
>> >>>>>> Hi,
>> >>>>>>
>> >>>>>> By default Jigasi mixes the audio streams coming from the
>> >>>>>> jitsi-meet
>> >>>>>> conference and sends one audio stream to the sip side.
>> >>>>>> When mixing audio this is the limitation that jigasi can handle a
>> >>>>>> number of conferences till it hits a CPU limitation, we do not
have
>> >>>>>> at
>> >>>>>> the moment any estimations and it depends on the resources of the
>> >>>>>> machine running the jigasi server.
>> >>>>>>
>> >>>>>> There is an option to bypass this and just send multiple audio
>> >>>>>> streams
>> >>>>>> to the sip side, but there is no legacy sip server(asterisk or
>> >>>>>> freeswitch) at the moment that handles that. We deployed jigasi
>> >>>>>> servers to meet.jit.si to call in, which are forwarding all
streams
>> >>>>>> to
>> >>>>>> voximplant (the provider we use for those jigasi servers there)
and
>> >>>>>> they handle that.
>> >>>>>>
>> >>>>>> Hope this clears things a bit and helps.
>> >>>>>>
>> >>>>>> Regards
>> >>>>>> damencho
>> >>>>>>
>> >>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie <kaiduanx@gmail.com > > > >> >>>>>> wrote:
>> >>>>>> > Hi all,
>> >>>>>> >
>> >>>>>> > I have some questions about Jigasi multiple SIP call support,
>> >>>>>> >
>> >>>>>> > Can focus in the conference invite multiple SIP clients to this
>> >>>>>> > conference?
>> >>>>>> > Does Jigasi mix the audio? What is the limitation about the
>> >>>>>> > number
>> >>>>>> > of
>> >>>>>> > concurrent SIP calls for every conference?
>> >>>>>> >
>> >>>>>> > Per How it works on https://github.com/jitsi/jigasi
>> >>>>>> >
>> >>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination and
>> >>>>>> > multiple-SSRC handling for them."
>> >>>>>> >
>> >>>>>> > Can you provide more details about the multiple-SSRC handling?
>> >>>>>> > Does
>> >>>>>> > Jigasi
>> >>>>>> > mix the audio from webrtc participants? My understanding is
that
>> >>>>>> > in
>> >>>>>> > the
>> >>>>>> > conference where all parties are webrtc participants, the
browser
>> >>>>>> > itself
>> >>>>>> > mixes the audio.
>> >>>>>> >
>> >>>>>> > Thanks for the help,
>> >>>>>> >
>> >>>>>> > /Kaiduan
>> >>>>>> >
>> >>>>>> > _______________________________________________
>> >>>>>> > 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:
>> >>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>> >>>>>
>> >>>>>
>> >>>>
>> >>
>> >
>> >
>> > _______________________________________________
>> > 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:
>> http://lists.jitsi.org/mailman/listinfo/dev
>
>
>
> _______________________________________________
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> dev@jitsi.org
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#12

Hi,

Jigasi uses protocol providers from Jitsi desktop client. In the
desktop client you can create xmpp(jingle) and sip calls, and there is
an option to merge those calls.
https://github.com/jitsi/jigasi/blob/900d746db2a08b5d90c416e6693cc27360c37397/src/main/java/org/jitsi/jigasi/GatewaySession.java#L307-L307
Jigasi creates a sip call and sets conference object from xmpp call to
the sip call, this instructs it to merge both calls. In case of
mixing, what it does is that it creates a mixer and all the streams
(let's say A and B) got merged and are sent to the sip side, what is
received from sip side, let's say stream C is sent to the bridge
(after transcoding to appropriate codec) and the bridge send it to the
other endpoints.
In case of relaying, no mixing and whatever is received it is sent, no
decoding, so you need to setup with sip side same codec, opus.
Hope this clears a bit, it is very complicated thing so its a lot of
code, handling it :slight_smile:

Regards
damencho

···

On Thu, Mar 23, 2017 at 1:56 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Damencho,

Can you explain how the media is forwarded between SIP and jitsi-meet in
Jigasi? The signalling handling between SIP and jitsi-meet is obvious from
reading the code, but the media processing is not that clear.

Thanks,

/Kaiduan

On Thu, Mar 23, 2017 at 1:51 PM, Damian Minkov <damencho@jitsi.org> wrote:

Yep that is correct, but I'm not sure there is some open source sip
pbx supporting multi-streams.

On Thu, Mar 23, 2017 at 12:47 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Thanks Damencho.
>
> So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix
> audio
> from jitsi-meet conference, right?
>
> /Kaiduan
>
> On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> >> > wrote:
>>
>> Hi,
>>
>> Here by default it creates AudioMixer for the audio:
>>
>>
>> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L312-L312
>>
>> If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
>> translator:
>>
>>
>> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b4166566546/src/net/java/sip/communicator/service/protocol/media/MediaAwareCallConference.java#L385-L385
>> And it just forwards RTP, the same way jitsi-videobridge uses libjitsi.
>>
>> Regards
>> damencho
>>
>>
>> On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> >> >> wrote:
>> > Damencho,
>> >
>> > Can you kindly point out the source code where Jigasi mixes the audio
>> > streams coming from the jitsi-meet conference and sends one audio
>> > stream
>> > to
>> > the sip side?
>> >
>> > Thanks,
>> >
>> > /Kaiduan
>> >
>> > On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> >> >> > wrote:
>> >>
>> >> Arthur,
>> >>
>> >> The test is described as below,
>> >>
>> >> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with 1000/1001
>> >> SIP
>> >> accounts respectively.
>> >>
>> >> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on Ubuntu,
>> >> Jigasi
>> >> registers with 1004 SIP account to Freeswitch,
>> >>
>> >> 3. Two parties join the conference from two Chrome browsers.
>> >>
>> >> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the call.
>> >>
>> >> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the call.
>> >>
>> >> 6. Two webrtc participants and two SIP phone are in the conference.
>> >>
>> >> /Kaiduan
>> >>
>> >>
>> >>
>> >> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am> >> >> >> wrote:
>> >>>
>> >>> Hi Kaiduan,
>> >>>
>> >>> The test you did by making jigasi to call two SIP clients is very
>> >>> interesting for me.
>> >>>
>> >>> Could you help me to reproduce such test at my side?
>> >>>
>> >>> I would be very thankful if you could describe the steps briefly.
>> >>>
>> >>> Regards,
>> >>> Arthur Petrosyan
>> >>>
>> >>>
>> >>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie >> >> >>> <kaiduanx@gmail.com> >> >> >>> wrote:
>> >>>>
>> >>>> I did one test where jigasi called two SIP clients registered to
>> >>>> Freeswitch in a webrtc conference, the webrtc clients and SIP can
>> >>>> hear each
>> >>>> other without issues.
>> >>>>
>> >>>> So it looks like Jigasi also mixes audio streams from SIP world.
>> >>>>
>> >>>> Regards,
>> >>>>
>> >>>> /Kaiduan
>> >>>>
>> >>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie <kaiduanx@gmail.com> >> >> >>>> wrote:
>> >>>>>
>> >>>>> Damencho,
>> >>>>>
>> >>>>> Thanks for the swift response.
>> >>>>>
>> >>>>> So Jigasi does mix audio from the webrtc side.
>> >>>>>
>> >>>>> From SIP side, does Jigasi mix the multiple audio streams from
>> >>>>> different SIP clients in the case where Jigasi calls out multiple
>> >>>>> SIP
>> >>>>> clients?
>> >>>>>
>> >>>>> Thanks,
>> >>>>>
>> >>>>> /Kaiduan
>> >>>>>
>> >>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov >> >> >>>>> <damencho@jitsi.org> >> >> >>>>> wrote:
>> >>>>>>
>> >>>>>> Hi,
>> >>>>>>
>> >>>>>> By default Jigasi mixes the audio streams coming from the
>> >>>>>> jitsi-meet
>> >>>>>> conference and sends one audio stream to the sip side.
>> >>>>>> When mixing audio this is the limitation that jigasi can handle
>> >>>>>> a
>> >>>>>> number of conferences till it hits a CPU limitation, we do not
>> >>>>>> have
>> >>>>>> at
>> >>>>>> the moment any estimations and it depends on the resources of
>> >>>>>> the
>> >>>>>> machine running the jigasi server.
>> >>>>>>
>> >>>>>> There is an option to bypass this and just send multiple audio
>> >>>>>> streams
>> >>>>>> to the sip side, but there is no legacy sip server(asterisk or
>> >>>>>> freeswitch) at the moment that handles that. We deployed jigasi
>> >>>>>> servers to meet.jit.si to call in, which are forwarding all
>> >>>>>> streams
>> >>>>>> to
>> >>>>>> voximplant (the provider we use for those jigasi servers there)
>> >>>>>> and
>> >>>>>> they handle that.
>> >>>>>>
>> >>>>>> Hope this clears things a bit and helps.
>> >>>>>>
>> >>>>>> Regards
>> >>>>>> damencho
>> >>>>>>
>> >>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie >> >> >>>>>> <kaiduanx@gmail.com> >> >> >>>>>> wrote:
>> >>>>>> > Hi all,
>> >>>>>> >
>> >>>>>> > I have some questions about Jigasi multiple SIP call support,
>> >>>>>> >
>> >>>>>> > Can focus in the conference invite multiple SIP clients to
>> >>>>>> > this
>> >>>>>> > conference?
>> >>>>>> > Does Jigasi mix the audio? What is the limitation about the
>> >>>>>> > number
>> >>>>>> > of
>> >>>>>> > concurrent SIP calls for every conference?
>> >>>>>> >
>> >>>>>> > Per How it works on https://github.com/jitsi/jigasi
>> >>>>>> >
>> >>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination
>> >>>>>> > and
>> >>>>>> > multiple-SSRC handling for them."
>> >>>>>> >
>> >>>>>> > Can you provide more details about the multiple-SSRC handling?
>> >>>>>> > Does
>> >>>>>> > Jigasi
>> >>>>>> > mix the audio from webrtc participants? My understanding is
>> >>>>>> > that
>> >>>>>> > in
>> >>>>>> > the
>> >>>>>> > conference where all parties are webrtc participants, the
>> >>>>>> > browser
>> >>>>>> > itself
>> >>>>>> > mixes the audio.
>> >>>>>> >
>> >>>>>> > Thanks for the help,
>> >>>>>> >
>> >>>>>> > /Kaiduan
>> >>>>>> >
>> >>>>>> > _______________________________________________
>> >>>>>> > 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:
>> >>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>> >>>>>
>> >>>>>
>> >>>>
>> >>
>> >
>> >
>> > _______________________________________________
>> > dev mailing list
>> > dev@jitsi.org
>> > Unsubscribe instructions and other list options:
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>> _______________________________________________
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#13

Ahha, call.setConference is the key point, thanks a lot Damencho for the
pointer.

/Kaiduan

···

On Thu, Mar 23, 2017 at 3:13 PM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

Jigasi uses protocol providers from Jitsi desktop client. In the
desktop client you can create xmpp(jingle) and sip calls, and there is
an option to merge those calls.
https://github.com/jitsi/jigasi/blob/900d746db2a08b5d90c416e6693cc2
7360c37397/src/main/java/org/jitsi/jigasi/GatewaySession.java#L307-L307
Jigasi creates a sip call and sets conference object from xmpp call to
the sip call, this instructs it to merge both calls. In case of
mixing, what it does is that it creates a mixer and all the streams
(let's say A and B) got merged and are sent to the sip side, what is
received from sip side, let's say stream C is sent to the bridge
(after transcoding to appropriate codec) and the bridge send it to the
other endpoints.
In case of relaying, no mixing and whatever is received it is sent, no
decoding, so you need to setup with sip side same codec, opus.
Hope this clears a bit, it is very complicated thing so its a lot of
code, handling it :slight_smile:

Regards
damencho

On Thu, Mar 23, 2017 at 1:56 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Damencho,
>
> Can you explain how the media is forwarded between SIP and jitsi-meet in
> Jigasi? The signalling handling between SIP and jitsi-meet is obvious
from
> reading the code, but the media processing is not that clear.
>
> Thanks,
>
> /Kaiduan
>
> On Thu, Mar 23, 2017 at 1:51 PM, Damian Minkov <damencho@jitsi.org> > wrote:
>>
>> Yep that is correct, but I'm not sure there is some open source sip
>> pbx supporting multi-streams.
>>
>> On Thu, Mar 23, 2017 at 12:47 PM, Kaiduan Xie <kaiduanx@gmail.com> > wrote:
>> > Thanks Damencho.
>> >
>> > So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix
>> > audio
>> > from jitsi-meet conference, right?
>> >
>> > /Kaiduan
>> >
>> > On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> > >> > wrote:
>> >>
>> >> Hi,
>> >>
>> >> Here by default it creates AudioMixer for the audio:
>> >>
>> >>
>> >> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L312-L312
>> >>
>> >> If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
>> >> translator:
>> >>
>> >>
>> >> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199b1883b
4166566546/src/net/java/sip/communicator/service/protocol/media/
MediaAwareCallConference.java#L385-L385
>> >> And it just forwards RTP, the same way jitsi-videobridge uses
libjitsi.
>> >>
>> >> Regards
>> >> damencho
>> >>
>> >>
>> >> On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> > >> >> wrote:
>> >> > Damencho,
>> >> >
>> >> > Can you kindly point out the source code where Jigasi mixes the
audio
>> >> > streams coming from the jitsi-meet conference and sends one audio
>> >> > stream
>> >> > to
>> >> > the sip side?
>> >> >
>> >> > Thanks,
>> >> >
>> >> > /Kaiduan
>> >> >
>> >> > On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com> > >> >> > wrote:
>> >> >>
>> >> >> Arthur,
>> >> >>
>> >> >> The test is described as below,
>> >> >>
>> >> >> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with
1000/1001
>> >> >> SIP
>> >> >> accounts respectively.
>> >> >>
>> >> >> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on
Ubuntu,
>> >> >> Jigasi
>> >> >> registers with 1004 SIP account to Freeswitch,
>> >> >>
>> >> >> 3. Two parties join the conference from two Chrome browsers.
>> >> >>
>> >> >> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the
call.
>> >> >>
>> >> >> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the
call.
>> >> >>
>> >> >> 6. Two webrtc participants and two SIP phone are in the
conference.
>> >> >>
>> >> >> /Kaiduan
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan <arthur@sci.am > > > >> >> >> wrote:
>> >> >>>
>> >> >>> Hi Kaiduan,
>> >> >>>
>> >> >>> The test you did by making jigasi to call two SIP clients is very
>> >> >>> interesting for me.
>> >> >>>
>> >> >>> Could you help me to reproduce such test at my side?
>> >> >>>
>> >> >>> I would be very thankful if you could describe the steps briefly.
>> >> >>>
>> >> >>> Regards,
>> >> >>> Arthur Petrosyan
>> >> >>>
>> >> >>>
>> >> >>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie > >> >> >>> <kaiduanx@gmail.com> > >> >> >>> wrote:
>> >> >>>>
>> >> >>>> I did one test where jigasi called two SIP clients registered to
>> >> >>>> Freeswitch in a webrtc conference, the webrtc clients and SIP
can
>> >> >>>> hear each
>> >> >>>> other without issues.
>> >> >>>>
>> >> >>>> So it looks like Jigasi also mixes audio streams from SIP world.
>> >> >>>>
>> >> >>>> Regards,
>> >> >>>>
>> >> >>>> /Kaiduan
>> >> >>>>
>> >> >>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie < > kaiduanx@gmail.com> > >> >> >>>> wrote:
>> >> >>>>>
>> >> >>>>> Damencho,
>> >> >>>>>
>> >> >>>>> Thanks for the swift response.
>> >> >>>>>
>> >> >>>>> So Jigasi does mix audio from the webrtc side.
>> >> >>>>>
>> >> >>>>> From SIP side, does Jigasi mix the multiple audio streams from
>> >> >>>>> different SIP clients in the case where Jigasi calls out
multiple
>> >> >>>>> SIP
>> >> >>>>> clients?
>> >> >>>>>
>> >> >>>>> Thanks,
>> >> >>>>>
>> >> >>>>> /Kaiduan
>> >> >>>>>
>> >> >>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov > >> >> >>>>> <damencho@jitsi.org> > >> >> >>>>> wrote:
>> >> >>>>>>
>> >> >>>>>> Hi,
>> >> >>>>>>
>> >> >>>>>> By default Jigasi mixes the audio streams coming from the
>> >> >>>>>> jitsi-meet
>> >> >>>>>> conference and sends one audio stream to the sip side.
>> >> >>>>>> When mixing audio this is the limitation that jigasi can
handle
>> >> >>>>>> a
>> >> >>>>>> number of conferences till it hits a CPU limitation, we do not
>> >> >>>>>> have
>> >> >>>>>> at
>> >> >>>>>> the moment any estimations and it depends on the resources of
>> >> >>>>>> the
>> >> >>>>>> machine running the jigasi server.
>> >> >>>>>>
>> >> >>>>>> There is an option to bypass this and just send multiple audio
>> >> >>>>>> streams
>> >> >>>>>> to the sip side, but there is no legacy sip server(asterisk or
>> >> >>>>>> freeswitch) at the moment that handles that. We deployed
jigasi
>> >> >>>>>> servers to meet.jit.si to call in, which are forwarding all
>> >> >>>>>> streams
>> >> >>>>>> to
>> >> >>>>>> voximplant (the provider we use for those jigasi servers
there)
>> >> >>>>>> and
>> >> >>>>>> they handle that.
>> >> >>>>>>
>> >> >>>>>> Hope this clears things a bit and helps.
>> >> >>>>>>
>> >> >>>>>> Regards
>> >> >>>>>> damencho
>> >> >>>>>>
>> >> >>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie > >> >> >>>>>> <kaiduanx@gmail.com> > >> >> >>>>>> wrote:
>> >> >>>>>> > Hi all,
>> >> >>>>>> >
>> >> >>>>>> > I have some questions about Jigasi multiple SIP call
support,
>> >> >>>>>> >
>> >> >>>>>> > Can focus in the conference invite multiple SIP clients to
>> >> >>>>>> > this
>> >> >>>>>> > conference?
>> >> >>>>>> > Does Jigasi mix the audio? What is the limitation about the
>> >> >>>>>> > number
>> >> >>>>>> > of
>> >> >>>>>> > concurrent SIP calls for every conference?
>> >> >>>>>> >
>> >> >>>>>> > Per How it works on https://github.com/jitsi/jigasi
>> >> >>>>>> >
>> >> >>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination
>> >> >>>>>> > and
>> >> >>>>>> > multiple-SSRC handling for them."
>> >> >>>>>> >
>> >> >>>>>> > Can you provide more details about the multiple-SSRC
handling?
>> >> >>>>>> > Does
>> >> >>>>>> > Jigasi
>> >> >>>>>> > mix the audio from webrtc participants? My understanding is
>> >> >>>>>> > that
>> >> >>>>>> > in
>> >> >>>>>> > the
>> >> >>>>>> > conference where all parties are webrtc participants, the
>> >> >>>>>> > browser
>> >> >>>>>> > itself
>> >> >>>>>> > mixes the audio.
>> >> >>>>>> >
>> >> >>>>>> > Thanks for the help,
>> >> >>>>>> >
>> >> >>>>>> > /Kaiduan
>> >> >>>>>> >
>> >> >>>>>> > _______________________________________________
>> >> >>>>>> > 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:
>> >> >>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>> >> >>>>>
>> >> >>>>>
>> >> >>>>
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > 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:
>> >> http://lists.jitsi.org/mailman/listinfo/dev
>> >
>> >
>> >
>> > _______________________________________________
>> > 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:
>> http://lists.jitsi.org/mailman/listinfo/dev
>
>
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
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#14

Damencho,

After reading the source code of libjisti, I have some rough ideas about
the incoming and outgoing SRTP processing in libjitsi.

However I am not clear where media is received on SIP leg, and how media is
forwarded between SIP leg and webrtc leg. Can you kindly point some
important points in the source code?

1. The code where UDP socket is created for SIP leg, and the code where the
incoming RTP packet is received on the UDP socket?

2. The code where the incoming RTP packet from SIP leg is forwarded to the
webrtc leg?

3. The code where the RTP packet from webrtc leg is forwarded to the SIP
leg?

4. The code where the RTP packet is sent to peer SIP phone from the UDP
socket created in step 1?

I think these important points will help undestanding the internal media
processing.

Thanks for help,

/Kaiduan

···

On Thu, Mar 23, 2017 at 4:09 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Ahha, call.setConference is the key point, thanks a lot Damencho for the
pointer.

/Kaiduan

On Thu, Mar 23, 2017 at 3:13 PM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

Jigasi uses protocol providers from Jitsi desktop client. In the
desktop client you can create xmpp(jingle) and sip calls, and there is
an option to merge those calls.
https://github.com/jitsi/jigasi/blob/900d746db2a08b5d90c416e
6693cc27360c37397/src/main/java/org/jitsi/jigasi/
GatewaySession.java#L307-L307
Jigasi creates a sip call and sets conference object from xmpp call to
the sip call, this instructs it to merge both calls. In case of
mixing, what it does is that it creates a mixer and all the streams
(let's say A and B) got merged and are sent to the sip side, what is
received from sip side, let's say stream C is sent to the bridge
(after transcoding to appropriate codec) and the bridge send it to the
other endpoints.
In case of relaying, no mixing and whatever is received it is sent, no
decoding, so you need to setup with sip side same codec, opus.
Hope this clears a bit, it is very complicated thing so its a lot of
code, handling it :slight_smile:

Regards
damencho

On Thu, Mar 23, 2017 at 1:56 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Damencho,
>
> Can you explain how the media is forwarded between SIP and jitsi-meet in
> Jigasi? The signalling handling between SIP and jitsi-meet is obvious
from
> reading the code, but the media processing is not that clear.
>
> Thanks,
>
> /Kaiduan
>
> On Thu, Mar 23, 2017 at 1:51 PM, Damian Minkov <damencho@jitsi.org> >> wrote:
>>
>> Yep that is correct, but I'm not sure there is some open source sip
>> pbx supporting multi-streams.
>>
>> On Thu, Mar 23, 2017 at 12:47 PM, Kaiduan Xie <kaiduanx@gmail.com> >> wrote:
>> > Thanks Damencho.
>> >
>> > So if USE_TRANSLATOR_IN_CONFERENCE is set, then Jigasi does not mix
>> > audio
>> > from jitsi-meet conference, right?
>> >
>> > /Kaiduan
>> >
>> > On Thu, Mar 23, 2017 at 1:39 PM, Damian Minkov <damencho@jitsi.org> >> >> > wrote:
>> >>
>> >> Hi,
>> >>
>> >> Here by default it creates AudioMixer for the audio:
>> >>
>> >>
>> >> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199
b1883b4166566546/src/net/java/sip/communicator/service/
protocol/media/MediaAwareCallConference.java#L312-L312
>> >>
>> >> If account property USE_TRANSLATOR_IN_CONFERENCE is set it uses a
>> >> translator:
>> >>
>> >>
>> >> https://github.com/jitsi/jitsi/blob/3f28fcef07c39ab735917199
b1883b4166566546/src/net/java/sip/communicator/service/
protocol/media/MediaAwareCallConference.java#L385-L385
>> >> And it just forwards RTP, the same way jitsi-videobridge uses
libjitsi.
>> >>
>> >> Regards
>> >> damencho
>> >>
>> >>
>> >> On Thu, Mar 23, 2017 at 12:27 PM, Kaiduan Xie <kaiduanx@gmail.com> >> >> >> wrote:
>> >> > Damencho,
>> >> >
>> >> > Can you kindly point out the source code where Jigasi mixes the
audio
>> >> > streams coming from the jitsi-meet conference and sends one audio
>> >> > stream
>> >> > to
>> >> > the sip side?
>> >> >
>> >> > Thanks,
>> >> >
>> >> > /Kaiduan
>> >> >
>> >> > On Mon, Mar 20, 2017 at 11:12 AM, Kaiduan Xie <kaiduanx@gmail.com >> > >> >> >> > wrote:
>> >> >>
>> >> >> Arthur,
>> >> >>
>> >> >> The test is described as below,
>> >> >>
>> >> >> 1. Two X-Lite soft phones register to FreeSwitch 1.6 with
1000/1001
>> >> >> SIP
>> >> >> accounts respectively.
>> >> >>
>> >> >> 2. Jitsi-meet installation with jigasi_1.0-126_amd64.deb on
Ubuntu,
>> >> >> Jigasi
>> >> >> registers with 1004 SIP account to Freeswitch,
>> >> >>
>> >> >> 3. Two parties join the conference from two Chrome browsers.
>> >> >>
>> >> >> 4. Webrtc participant A dials 1000 SIP phone, 1000 answers the
call.
>> >> >>
>> >> >> 5. Webrtc participant A dials 1001 SIP phone, 1001 answers the
call.
>> >> >>
>> >> >> 6. Two webrtc participants and two SIP phone are in the
conference.
>> >> >>
>> >> >> /Kaiduan
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Sun, Mar 19, 2017 at 12:33 PM, Arthur Petrosyan < >> arthur@sci.am> >> >> >> >> wrote:
>> >> >>>
>> >> >>> Hi Kaiduan,
>> >> >>>
>> >> >>> The test you did by making jigasi to call two SIP clients is
very
>> >> >>> interesting for me.
>> >> >>>
>> >> >>> Could you help me to reproduce such test at my side?
>> >> >>>
>> >> >>> I would be very thankful if you could describe the steps
briefly.
>> >> >>>
>> >> >>> Regards,
>> >> >>> Arthur Petrosyan
>> >> >>>
>> >> >>>
>> >> >>> On March 17, 2017 10:19:21 PM GMT+04:00, Kaiduan Xie >> >> >> >>> <kaiduanx@gmail.com> >> >> >> >>> wrote:
>> >> >>>>
>> >> >>>> I did one test where jigasi called two SIP clients registered
to
>> >> >>>> Freeswitch in a webrtc conference, the webrtc clients and SIP
can
>> >> >>>> hear each
>> >> >>>> other without issues.
>> >> >>>>
>> >> >>>> So it looks like Jigasi also mixes audio streams from SIP
world.
>> >> >>>>
>> >> >>>> Regards,
>> >> >>>>
>> >> >>>> /Kaiduan
>> >> >>>>
>> >> >>>> On Fri, Mar 17, 2017 at 11:43 AM, Kaiduan Xie < >> kaiduanx@gmail.com> >> >> >> >>>> wrote:
>> >> >>>>>
>> >> >>>>> Damencho,
>> >> >>>>>
>> >> >>>>> Thanks for the swift response.
>> >> >>>>>
>> >> >>>>> So Jigasi does mix audio from the webrtc side.
>> >> >>>>>
>> >> >>>>> From SIP side, does Jigasi mix the multiple audio streams from
>> >> >>>>> different SIP clients in the case where Jigasi calls out
multiple
>> >> >>>>> SIP
>> >> >>>>> clients?
>> >> >>>>>
>> >> >>>>> Thanks,
>> >> >>>>>
>> >> >>>>> /Kaiduan
>> >> >>>>>
>> >> >>>>> On Fri, Mar 17, 2017 at 11:08 AM, Damian Minkov >> >> >> >>>>> <damencho@jitsi.org> >> >> >> >>>>> wrote:
>> >> >>>>>>
>> >> >>>>>> Hi,
>> >> >>>>>>
>> >> >>>>>> By default Jigasi mixes the audio streams coming from the
>> >> >>>>>> jitsi-meet
>> >> >>>>>> conference and sends one audio stream to the sip side.
>> >> >>>>>> When mixing audio this is the limitation that jigasi can
handle
>> >> >>>>>> a
>> >> >>>>>> number of conferences till it hits a CPU limitation, we do
not
>> >> >>>>>> have
>> >> >>>>>> at
>> >> >>>>>> the moment any estimations and it depends on the resources of
>> >> >>>>>> the
>> >> >>>>>> machine running the jigasi server.
>> >> >>>>>>
>> >> >>>>>> There is an option to bypass this and just send multiple
audio
>> >> >>>>>> streams
>> >> >>>>>> to the sip side, but there is no legacy sip server(asterisk
or
>> >> >>>>>> freeswitch) at the moment that handles that. We deployed
jigasi
>> >> >>>>>> servers to meet.jit.si to call in, which are forwarding all
>> >> >>>>>> streams
>> >> >>>>>> to
>> >> >>>>>> voximplant (the provider we use for those jigasi servers
there)
>> >> >>>>>> and
>> >> >>>>>> they handle that.
>> >> >>>>>>
>> >> >>>>>> Hope this clears things a bit and helps.
>> >> >>>>>>
>> >> >>>>>> Regards
>> >> >>>>>> damencho
>> >> >>>>>>
>> >> >>>>>> On Fri, Mar 17, 2017 at 9:38 AM, Kaiduan Xie >> >> >> >>>>>> <kaiduanx@gmail.com> >> >> >> >>>>>> wrote:
>> >> >>>>>> > Hi all,
>> >> >>>>>> >
>> >> >>>>>> > I have some questions about Jigasi multiple SIP call
support,
>> >> >>>>>> >
>> >> >>>>>> > Can focus in the conference invite multiple SIP clients to
>> >> >>>>>> > this
>> >> >>>>>> > conference?
>> >> >>>>>> > Does Jigasi mix the audio? What is the limitation about the
>> >> >>>>>> > number
>> >> >>>>>> > of
>> >> >>>>>> > concurrent SIP calls for every conference?
>> >> >>>>>> >
>> >> >>>>>> > Per How it works on https://github.com/jitsi/jigasi
>> >> >>>>>> >
>> >> >>>>>> > "It handles the XMPP signalling, ICE, DTLS/SRTP termination
>> >> >>>>>> > and
>> >> >>>>>> > multiple-SSRC handling for them."
>> >> >>>>>> >
>> >> >>>>>> > Can you provide more details about the multiple-SSRC
handling?
>> >> >>>>>> > Does
>> >> >>>>>> > Jigasi
>> >> >>>>>> > mix the audio from webrtc participants? My understanding is
>> >> >>>>>> > that
>> >> >>>>>> > in
>> >> >>>>>> > the
>> >> >>>>>> > conference where all parties are webrtc participants, the
>> >> >>>>>> > browser
>> >> >>>>>> > itself
>> >> >>>>>> > mixes the audio.
>> >> >>>>>> >
>> >> >>>>>> > Thanks for the help,
>> >> >>>>>> >
>> >> >>>>>> > /Kaiduan
>> >> >>>>>> >
>> >> >>>>>> > _______________________________________________
>> >> >>>>>> > 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:
>> >> >>>>>> http://lists.jitsi.org/mailman/listinfo/dev
>> >> >>>>>
>> >> >>>>>
>> >> >>>>
>> >> >>
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > 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:
>> >> http://lists.jitsi.org/mailman/listinfo/dev
>> >
>> >
>> >
>> > _______________________________________________
>> > 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:
>> http://lists.jitsi.org/mailman/listinfo/dev
>
>
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

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