[jitsi-users] Music on Hold problem


#1

Hi all,

I'm using the latest stable Jitsi client version and i've an issue with
Music on Hold. The problem is :

in a call between an hardphone and jitsi, if i put on hold from hardphone ,
into Jitsi client i hear the Music that Asterisk sends, but viceversa if i
put on hold from Jitsi the hardphone doesn't hear the music.
With whireshark i notice that when i put on hold jitsi continue to send rtp
packets to hardphone, so it receives 2 audio streams, one from jitsi and
one from asterisk (hold music).

···

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#2

Hey Rocco,

Hi all,

I'm using the latest stable Jitsi client version and i've an issue with
Music on Hold. The problem is :

in a call between an hardphone and jitsi, if i put on hold from hardphone ,
into Jitsi client i hear the Music that Asterisk sends, but viceversa if i
put on hold from Jitsi the hardphone doesn't hear the music.
With whireshark i notice that when i put on hold jitsi continue to send rtp
packets to hardphone, so it receives 2 audio streams, one from jitsi and one
from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

···

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta <rocco.sansotta@staersistemi.com> wrote:

--
Rocco Sansotta
Staer Sistemi s.r.l.
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

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users mailing list
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#3

Hi Emil,

it sounds strange.
I read this 2 doc
http://www.onsip.com/blog/2011/05/17/music-on-hold-and-the-sip-offeranswer-model-what-were-we-thinking
http://www.in2eps.com/fo-sip/tk-fo-sip-service-03.html

On both i see that during the Hold, there's no rtp stream between 2 clients
but only between asterisk and client on hold.
So I've believed that should be the client to stop his rtp stream and not
Asterisk, in fact when is the hardphone to put on hold, he doesn't send rtp
packets.

Btw, if i'm wrong, ther's a way to stop the stream from Jitsi?

···

2015-06-19 11:07 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta > <rocco.sansotta@staersistemi.com> wrote:
> Hi all,
>
> I'm using the latest stable Jitsi client version and i've an issue with
> Music on Hold. The problem is :
>
> in a call between an hardphone and jitsi, if i put on hold from
hardphone ,
> into Jitsi client i hear the Music that Asterisk sends, but viceversa if
i
> put on hold from Jitsi the hardphone doesn't hear the music.
> With whireshark i notice that when i put on hold jitsi continue to send
rtp
> packets to hardphone, so it receives 2 audio streams, one from jitsi and
one
> from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

>
> --
> Rocco Sansotta
> Staer Sistemi s.r.l.
> Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
> 00131 Roma (Italy)
> Skype : rocco_sansotta
>
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

--
https://jitsi.org

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#4

Hey Rocco,

Hi Emil,

it sounds strange.

It is defined here:

https://tools.ietf.org/html/rfc3264#section-8.4

The text explicitly talks about "muting" the stream (which is what we do).

Note that from a SIP perspective both Asterisk and Jitsi are user agents
(asterisk being a B2BUA) so their behavior for placing someone on hold
should be the same.

I read this 2 doc

http://www.onsip.com/blog/2011/05/17/music-on-hold-and-the-sip-offeranswer-model-what-were-we-thinking
http://www.in2eps.com/fo-sip/tk-fo-sip-service-03.html

Not sending RTP after you put someone on hold is certainly not a protocol
offense so there is nothing wrong with the docs you point to.

I personally wouldn't recommend this as stopping all RTP traffic would let
your NAT bindings expire (unless you are using ICE keep-alives) but
everyone is free to take their own chances.

On thing is for certain: if a server plans on generating MoH it should
definitely not be forwarding client media as well.

On both i see that during the Hold, there's no rtp stream between 2 clients

but only between asterisk and client on hold.

So I've believed that should be the client to stop his rtp stream and not

Asterisk, in fact when is the hardphone to put on hold, he doesn't send rtp
packets.

Btw, if i'm wrong, ther's a way to stop the stream from Jitsi?

There is currently none. I'd really recommend raising an issue with Aterisk.

Emil

···

On Friday, June 19, 2015, Rocco Sansotta <rocco.sansotta@staersistemi.com> wrote:

2015-06-19 11:07 GMT+02:00 Emil Ivov <emcho@jitsi.org
<javascript:_e(%7B%7D,'cvml','emcho@jitsi.org');>>:

Hey Rocco,

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta >> <rocco.sansotta@staersistemi.com >> <javascript:_e(%7B%7D,'cvml','rocco.sansotta@staersistemi.com');>> wrote:
> Hi all,
>
> I'm using the latest stable Jitsi client version and i've an issue with
> Music on Hold. The problem is :
>
> in a call between an hardphone and jitsi, if i put on hold from
hardphone ,
> into Jitsi client i hear the Music that Asterisk sends, but viceversa
if i
> put on hold from Jitsi the hardphone doesn't hear the music.
> With whireshark i notice that when i put on hold jitsi continue to send
rtp
> packets to hardphone, so it receives 2 audio streams, one from jitsi
and one
> from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

>
> --
> Rocco Sansotta
> Staer Sistemi s.r.l.
> Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
> 00131 Roma (Italy)
> Skype : rocco_sansotta
>
>
> _______________________________________________
> users mailing list
> users@jitsi.org <javascript:_e(%7B%7D,'cvml','users@jitsi.org');>
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

--
https://jitsi.org

_______________________________________________
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users@jitsi.org <javascript:_e(%7B%7D,'cvml','users@jitsi.org');>
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

--
sent from my mobile


#5

Thanks for the explanation Emil.

I've forgot one important thing : This problem appears only when Asterisk
is configured with the "RTP direct" option, so, all media packets not pass
through him. In that scenario, i think that, must be the client to stop his
rtp stream or just send few packet as "keep alive".

···

2015-06-20 0:49 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Friday, June 19, 2015, Rocco Sansotta <rocco.sansotta@staersistemi.com> > wrote:

Hi Emil,

it sounds strange.

It is defined here:

https://tools.ietf.org/html/rfc3264#section-8.4

The text explicitly talks about "muting" the stream (which is what we do).

Note that from a SIP perspective both Asterisk and Jitsi are user agents
(asterisk being a B2BUA) so their behavior for placing someone on hold
should be the same.

I read this 2 doc

http://www.onsip.com/blog/2011/05/17/music-on-hold-and-the-sip-offeranswer-model-what-were-we-thinking
http://www.in2eps.com/fo-sip/tk-fo-sip-service-03.html

Not sending RTP after you put someone on hold is certainly not a protocol
offense so there is nothing wrong with the docs you point to.

I personally wouldn't recommend this as stopping all RTP traffic would
let your NAT bindings expire (unless you are using ICE keep-alives) but
everyone is free to take their own chances.

On thing is for certain: if a server plans on generating MoH it should
definitely not be forwarding client media as well.

On both i see that during the Hold, there's no rtp stream between 2

clients but only between asterisk and client on hold.

So I've believed that should be the client to stop his rtp stream and not

Asterisk, in fact when is the hardphone to put on hold, he doesn't send rtp
packets.

Btw, if i'm wrong, ther's a way to stop the stream from Jitsi?

There is currently none. I'd really recommend raising an issue with
Aterisk.

Emil

2015-06-19 11:07 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta >>> <rocco.sansotta@staersistemi.com> wrote:
> Hi all,
>
> I'm using the latest stable Jitsi client version and i've an issue with
> Music on Hold. The problem is :
>
> in a call between an hardphone and jitsi, if i put on hold from
hardphone ,
> into Jitsi client i hear the Music that Asterisk sends, but viceversa
if i
> put on hold from Jitsi the hardphone doesn't hear the music.
> With whireshark i notice that when i put on hold jitsi continue to
send rtp
> packets to hardphone, so it receives 2 audio streams, one from jitsi
and one
> from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

>
> --
> Rocco Sansotta
> Staer Sistemi s.r.l.
> Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
> 00131 Roma (Italy)
> Skype : rocco_sansotta
>
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

--
https://jitsi.org

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

--
sent from my mobile

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#6

If Asterisk doesn't transmit the RTP packets in an active call, then the client, i.e. Jitsi, would be responsible for generating MoH while on hold. If Asterisk sends MoH then it creates a second stream when it IMO shouldn't.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

···

On 23.06.2015, at 13:49, Rocco Sansotta <rocco.sansotta@staersistemi.com> wrote:

Thanks for the explanation Emil.

I've forgot one important thing : This problem appears only when Asterisk is configured with the "RTP direct" option, so, all media packets not pass through him. In that scenario, i think that, must be the client to stop his rtp stream or just send few packet as "keep alive".

2015-06-20 0:49 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Friday, June 19, 2015, Rocco Sansotta <rocco.sansotta@staersistemi.com> wrote:
Hi Emil,

it sounds strange.

It is defined here:

https://tools.ietf.org/html/rfc3264#section-8.4

The text explicitly talks about "muting" the stream (which is what we do).

Note that from a SIP perspective both Asterisk and Jitsi are user agents (asterisk being a B2BUA) so their behavior for placing someone on hold should be the same.

I read this 2 doc
http://www.onsip.com/blog/2011/05/17/music-on-hold-and-the-sip-offeranswer-model-what-were-we-thinking
http://www.in2eps.com/fo-sip/tk-fo-sip-service-03.html

Not sending RTP after you put someone on hold is certainly not a protocol offense so there is nothing wrong with the docs you point to.

I personally wouldn't recommend this as stopping all RTP traffic would let your NAT bindings expire (unless you are using ICE keep-alives) but everyone is free to take their own chances.

On thing is for certain: if a server plans on generating MoH it should definitely not be forwarding client media as well.

On both i see that during the Hold, there's no rtp stream between 2 clients but only between asterisk and client on hold.
So I've believed that should be the client to stop his rtp stream and not Asterisk, in fact when is the hardphone to put on hold, he doesn't send rtp packets.

Btw, if i'm wrong, ther's a way to stop the stream from Jitsi?

There is currently none. I'd really recommend raising an issue with Aterisk.

Emil

2015-06-19 11:07 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta >>>> <rocco.sansotta@staersistemi.com> wrote:
> Hi all,
>
> I'm using the latest stable Jitsi client version and i've an issue with
> Music on Hold. The problem is :
>
> in a call between an hardphone and jitsi, if i put on hold from hardphone ,
> into Jitsi client i hear the Music that Asterisk sends, but viceversa if i
> put on hold from Jitsi the hardphone doesn't hear the music.
> With whireshark i notice that when i put on hold jitsi continue to send rtp
> packets to hardphone, so it receives 2 audio streams, one from jitsi and one
> from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

>
> --
> Rocco Sansotta
> Staer Sistemi s.r.l.
> Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
> 00131 Roma (Italy)
> Skype : rocco_sansotta
>
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

--
https://jitsi.org

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

--
Rocco Sansotta
Staer Sistemi s.r.l.
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

--
sent from my mobile

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

--
Rocco Sansotta
Staer Sistemi s.r.l.
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

_______________________________________________
users mailing list
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#7

But in a large scenario, where there are a lot of different clients
(hardphones, jitsi like etc.) and not all clients have a possibility to
generate a MoH, IMO Asterisk must do the work.

···

2015-06-23 12:23 GMT+02:00 Ingo Bauersachs <ingo@jitsi.org>:

If Asterisk doesn't transmit the RTP packets in an active call, then the
client, i.e. Jitsi, would be responsible for generating MoH while on hold.
If Asterisk sends MoH then it creates a second stream when it IMO shouldn't.

Ingo

Freundliche Grüsse,
Ingo Bauersachs

-- sent from my mobile

On 23.06.2015, at 13:49, Rocco Sansotta <rocco.sansotta@staersistemi.com> > wrote:

Thanks for the explanation Emil.

I've forgot one important thing : This problem appears only when Asterisk
is configured with the "RTP direct" option, so, all media packets not pass
through him. In that scenario, i think that, must be the client to stop his
rtp stream or just send few packet as "keep alive".

2015-06-20 0:49 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Friday, June 19, 2015, Rocco Sansotta <rocco.sansotta@staersistemi.com> >> wrote:

Hi Emil,

it sounds strange.

It is defined here:

https://tools.ietf.org/html/rfc3264#section-8.4

The text explicitly talks about "muting" the stream (which is what we do).

Note that from a SIP perspective both Asterisk and Jitsi are user agents
(asterisk being a B2BUA) so their behavior for placing someone on hold
should be the same.

I read this 2 doc

http://www.onsip.com/blog/2011/05/17/music-on-hold-and-the-sip-offeranswer-model-what-were-we-thinking
http://www.in2eps.com/fo-sip/tk-fo-sip-service-03.html

Not sending RTP after you put someone on hold is certainly not a protocol
offense so there is nothing wrong with the docs you point to.

I personally wouldn't recommend this as stopping all RTP traffic would
let your NAT bindings expire (unless you are using ICE keep-alives) but
everyone is free to take their own chances.

On thing is for certain: if a server plans on generating MoH it should
definitely not be forwarding client media as well.

On both i see that during the Hold, there's no rtp stream between 2

clients but only between asterisk and client on hold.

So I've believed that should be the client to stop his rtp stream and not

Asterisk, in fact when is the hardphone to put on hold, he doesn't send rtp
packets.

Btw, if i'm wrong, ther's a way to stop the stream from Jitsi?

There is currently none. I'd really recommend raising an issue with
Aterisk.

Emil

2015-06-19 11:07 GMT+02:00 Emil Ivov <emcho@jitsi.org>:

Hey Rocco,

On Fri, Jun 19, 2015 at 10:54 AM, Rocco Sansotta >>>> <rocco.sansotta@staersistemi.com> wrote:
> Hi all,
>
> I'm using the latest stable Jitsi client version and i've an issue
with
> Music on Hold. The problem is :
>
> in a call between an hardphone and jitsi, if i put on hold from
hardphone ,
> into Jitsi client i hear the Music that Asterisk sends, but viceversa
if i
> put on hold from Jitsi the hardphone doesn't hear the music.
> With whireshark i notice that when i put on hold jitsi continue to
send rtp
> packets to hardphone, so it receives 2 audio streams, one from jitsi
and one
> from asterisk (hold music).

That's exactly how SIP hold works. The fact that you send packets when
you put someone on hold allows you to play music on hold.

Obviously if Asterisk is planning on generating its own MoH then it
shouldn't be forwarding the RTP from Jitsi.

Hope this helps,
Emil

>
> --
> Rocco Sansotta
> Staer Sistemi s.r.l.
> Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
> 00131 Roma (Italy)
> Skype : rocco_sansotta
>
>
> _______________________________________________
> users mailing list
> users@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/users

--
https://jitsi.org

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

--
sent from my mobile

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta

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

--
*Rocco Sansotta*
*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#8

But in a large scenario, where there are a lot of different clients
(hardphones, jitsi like etc.) and not all clients have a possibility to
generate a MoH, IMO Asterisk must do the work.

Yeah, but then you can't use direct RTP. It's one or the other.

There's another problem if encryption is enabled: the audio stream between two clients is end-to-end encrypted. Asterisk cannot just start sending unencrypted MoH.

Ingo


#9

Yeah, but then you can't use direct RTP. It's one or the other.

For performance reason in that case is prefered "direct RTP".

Then ok, i understand that the only thing i can do is to use the MoH on
jitsi, but i didn't see any option to enable that feature. Any tips?

Thanks

···

2015-06-23 17:30 GMT+02:00 Ingo Bauersachs <ingo@jitsi.org>:

> But in a large scenario, where there are a lot of different clients
> (hardphones, jitsi like etc.) and not all clients have a possibility to
> generate a MoH, IMO Asterisk must do the work.

Yeah, but then you can't use direct RTP. It's one or the other.

There's another problem if encryption is enabled: the audio stream between
two clients is end-to-end encrypted. Asterisk cannot just start sending
unencrypted MoH.

Ingo

_______________________________________________
users mailing list
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Unsubscribe instructions and other list options:
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--
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*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#10

Yeah, but then you can't use direct RTP. It's one or the other.

For performance reason in that case is prefered "direct RTP".

Then ok, i understand that the only thing i can do is to use the MoH on
jitsi, but i didn't see any option to enable that feature. Any tips?

Jitsi has no MoH feature.

Thanks

Ingo


#11

So, all these speeches are useless if jitsi has no MoH.

Back to my first question.. i tried to set MediaDirection.Inactive when the
"on hold" function is called but i have no seen difference, have you any
advice on where i can stop rtp stream?

···

2015-06-23 17:53 GMT+02:00 Ingo Bauersachs <ingo@jitsi.org>:

>> Yeah, but then you can't use direct RTP. It's one or the other.
>
> For performance reason in that case is prefered "direct RTP".
>
> Then ok, i understand that the only thing i can do is to use the MoH on
> jitsi, but i didn't see any option to enable that feature. Any tips?

Jitsi has no MoH feature.

> Thanks

Ingo

_______________________________________________
users mailing list
users@jitsi.org
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--
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*Staer Sistemi s.r.l.*
Via Giacomo Peroni, 400-402, modulo 5B, 1° piano
00131 Roma (Italy)
Skype : rocco_sansotta


#12

So, all these speeches are useless if jitsi has no MoH.

The point of al these "speeches" is to indicate that there is a bug in
Asterisk that you should report on their mailing list. More specifically:
when holding a direct rtp session, if they plan on producing MoH, they
should put themselves on the media path of both agents and not only on that
of the holdee.

Not to mention that the agent on the other side shouldn't be freaking out
because some other party is sending RTP. Otherwise they would be easily
DoS-able.

Back to my first question.. i tried to set MediaDirection.Inactive when the

"on hold" function is called but i have no seen difference, have you any
advice on where i can stop rtp stream?

Depending on whee you did it that should have produced what you are looking
for (and it would also cause NAT traversal issues over the Internet)

Emil

···

On Wednesday, June 24, 2015, Rocco Sansotta <rocco.sansotta@staersistemi.com> wrote:

2015-06-23 17:53 GMT+02:00 Ingo Bauersachs <ingo@jitsi.org
<javascript:_e(%7B%7D,'cvml','ingo@jitsi.org');>>:

>> Yeah, but then you can't use direct RTP. It's one or the other.
>
> For performance reason in that case is prefered "direct RTP".
>
> Then ok, i understand that the only thing i can do is to use the MoH on
> jitsi, but i didn't see any option to enable that feature. Any tips?

Jitsi has no MoH feature.

> Thanks

Ingo

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*Rocco Sansotta*
*Staer Sistemi s.r.l.*
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