[jitsi-users] WebRTC SFU


#1

So, I'd like to try Jitsi as an SFU for WebRTC broadcasing.

I am totally confused on what to download and install on an Ubuntu server?

Apparently there is major lack of documentation for new users. Most
examples are

using Windows, which I am not understanding at all?

Can someone step me through what to install and how to set up Jitsi for SFU?
I would appreciate that.

Thanks,

Ray


#2

https://github.com/jitsi/jitsi-meet/#installation

···

On 01/05/2017 14:40, Ray Jender wrote:

So, I’d like to try Jitsi as an SFU for WebRTC broadcasing.

I am totally confused on what to download and install on an Ubuntu server?

Apparently there is major lack of documentation for new users. Most
examples are

using Windows, which I am not understanding at all?

Can someone step me through what to install and how to set up Jitsi for
SFU? I would appreciate that.

Thanks,

Ray

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#3

Thanks, but I still do know how to interface my WebRTC broadcast to the Jitsi and distribute it to viewers?

Ray

···

-----Original Message-----
From: users [mailto:users-bounces@jitsi.org] On Behalf Of Boris Grozev
Sent: Monday, May 01, 2017 3:55 PM
To: Jitsi Users <users@jitsi.org>
Subject: Re: [jitsi-users] WebRTC SFU

https://github.com/jitsi/jitsi-meet/#installation

On 01/05/2017 14:40, Ray Jender wrote:

So, I’d like to try Jitsi as an SFU for WebRTC broadcasing.

I am totally confused on what to download and install on an Ubuntu server?

Apparently there is major lack of documentation for new users. Most
examples are

using Windows, which I am not understanding at all?

Can someone step me through what to install and how to set up Jitsi
for SFU? I would appreciate that.

Thanks,

Ray

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

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#4

So, I'd like to try Jitsi as an SFU for WebRTC broadcasing.

What's an SFU?

https://en.wikipedia.org/wiki/Sfu

I am totally confused on what to download and install on an Ubuntu server?

Apparently there is major lack of documentation for new users. Most
examples are using Windows, which I am not understanding at all?

Are you talking about client installation, or server?

Can someone step me through what to install and how to set up Jitsi for
SFU? I would appreciate that.

Hm, what's an SFU?

Antony.

···

On Monday 01 May 2017 at 20:40:24, Ray Jender wrote:

--
"this restriction will not apply in the event of the occurrence (certified by
the United States Centers for Disease Control or successor body) of a
widespread viral infection transmitted via bites or contact with bodily fluids
that causes human corpses to reanimate and seek to consume living human flesh,
blood, brain or nerve tissue and is likely to result in the fall of organized
civilization."

- https://aws.amazon.com/service-terms/ paragraph 57.10

                                                   Please reply to the list;
                                                         please *don't* CC me.


#5

SFU = https://webrtcglossary.com/sfu/

Client or server? Probably server. The client will be the WebRTC traffic.

Ray

···

-----Original Message-----
From: users [mailto:users-bounces@jitsi.org] On Behalf Of Antony Stone
Sent: Monday, May 01, 2017 4:52 PM
To: Jitsi Users <users@jitsi.org>
Subject: Re: [jitsi-users] WebRTC SFU

On Monday 01 May 2017 at 20:40:24, Ray Jender wrote:

So, I'd like to try Jitsi as an SFU for WebRTC broadcasing.

What's an SFU?

https://en.wikipedia.org/wiki/Sfu

I am totally confused on what to download and install on an Ubuntu server?

Apparently there is major lack of documentation for new users. Most
examples are using Windows, which I am not understanding at all?

Are you talking about client installation, or server?

Can someone step me through what to install and how to set up Jitsi
for SFU? I would appreciate that.

Hm, what's an SFU?

Antony.

--
"this restriction will not apply in the event of the occurrence (certified by the United States Centers for Disease Control or successor body) of a widespread viral infection transmitted via bites or contact with bodily fluids that causes human corpses to reanimate and seek to consume living human flesh, blood, brain or nerve tissue and is likely to result in the fall of organized civilization."

- https://aws.amazon.com/service-terms/ paragraph 57.10

                                                   Please reply to the list;
                                                         please *don't* CC me.

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#6

I'll try one more time.

I am working on some of Muaz’s experiments and was wondering if you have any manuals or documents on how to use Jitsi and its SFU functionality?
I’d like to try this with some of Muaz’s experiments.

I have installed Jitsi on a Ubuntu 16.04 server, but I’m pretty much at a standstill as I am lacking documentation and a user manual for Ubuntu installs.

Everything I come up with shows a Windows install and a management GUI. How do I get that with Unbuntu?

Thanks,

Ray


#7

I have no idea what windows documentation you are referring to. I already pointed you the document that you need, which explains how to get the whole environment (including the SFU) set up on Debian or Ubuntu:
https://github.com/jitsi/jitsi-meet/#installation

Boris

···

On 03/05/2017 11:55, Ray Jender wrote:

I'll try one more time.

I am working on some of Muaz’s experiments and was wondering if you have any manuals or documents on how to use Jitsi and its SFU functionality?
I’d like to try this with some of Muaz’s experiments.

I have installed Jitsi on a Ubuntu 16.04 server, but I’m pretty much at a standstill as I am lacking documentation and a user manual for Ubuntu installs.

Everything I come up with shows a Windows install and a management GUI. How do I get that with Unbuntu?


#8

It only tells how to install. I have no idea how to actually use it? How do I send a WebRTC broadcast to multiple viewers for instance?

Ray

···

-----Original Message-----
From: users [mailto:users-bounces@jitsi.org] On Behalf Of Boris Grozev
Sent: Wednesday, May 03, 2017 5:37 PM
To: Jitsi Users <users@jitsi.org>
Subject: Re: [jitsi-users] WebRTC SFU

On 03/05/2017 11:55, Ray Jender wrote:

I'll try one more time.

I am working on some of Muaz’s experiments and was wondering if you have any manuals or documents on how to use Jitsi and its SFU functionality?
I’d like to try this with some of Muaz’s experiments.

I have installed Jitsi on a Ubuntu 16.04 server, but I’m pretty much at a standstill as I am lacking documentation and a user manual for Ubuntu installs.

Everything I come up with shows a Windows install and a management GUI. How do I get that with Unbuntu?

I have no idea what windows documentation you are referring to. I already pointed you the document that you need, which explains how to get the whole environment (including the SFU) set up on Debian or Ubuntu:
https://github.com/jitsi/jitsi-meet/#installation

Boris

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#9

Hi,

would like to know any performance report for Jigasi?
We face an issue that Jigasi start replying SIP 486 error after sending around 28 INVITEs to Jigasi.
We created 2 rooms, sending SIP INVITE at rate 10 INVITEs / second, and expect 20 participants join via Jigasi in each room.

Any idea what’s the potential cause?

Thanks,
Wing


#10

Do you have any logs?

···

On May 4, 2017 04:00, "Wing Chan" <WingChan@maaii.com> wrote:

Hi,

would like to know any performance report for Jigasi?
We face an issue that Jigasi start replying SIP 486 error after sending
around 28 INVITEs to Jigasi.
We created 2 rooms, sending SIP INVITE at rate 10 INVITEs / second, and
expect 20 participants join via Jigasi in each room.

Any idea what’s the potential cause?

Thanks,
Wing
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users@jitsi.org
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http://lists.jitsi.org/mailman/listinfo/users


#11

Hi,

I see this NullPointerException in Jigasi:
SEVERE: [8616] impl.protocol.sip.SipStackSharing.logApplicationException().1116 An error occurred while processing event of type: javax.sip.DialogTerminatedEvent
12876 java.lang.NullPointerException
12877 at net.sf.fmj.media.rtp.RTPSessionMgr.createSendStream(RTPSessionMgr.java:902)
12878 at net.sf.fmj.media.rtp.RTPSessionMgr.createSendStream(RTPSessionMgr.java:985)
12879 at org.jitsi.impl.neomedia.rtp.StreamRTPManager.createSendStream(StreamRTPManager.java:121)
12880 at org.jitsi.impl.neomedia.MediaStreamImpl.createSendStreams(MediaStreamImpl.java:903)
12881 at org.jitsi.impl.neomedia.MediaStreamImpl.startSendStreams(MediaStreamImpl.java:2971)
12882 at org.jitsi.impl.neomedia.MediaStreamImpl.start(MediaStreamImpl.java:2860)
12883 at org.jitsi.impl.neomedia.MediaStreamImpl.start(MediaStreamImpl.java:2823)
12884 at net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.start(CallPeerMediaHandler.java:1948)
12885 at net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.start(CallPeerMediaHandlerSipImpl.java:1909)
12886 at net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.processAck(CallPeerSipImpl.java:724)
12887 at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.processAck(OperationSetBasicTelephonySipImpl.java:1312)
12888 at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.processRequest(OperationSetBasicTelephonySipImpl.java:359)
12889 at net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.processRequest(ProtocolProviderServiceSipImpl.java:1044)
12890 at net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest(SipStackSharing.java:673)
12891 at gov.nist.javax.sip.EventScanner.deliverRequestEvent(EventScanner.java:250)
12892 at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:146)
12893 at gov.nist.javax.sip.SipProviderImpl.handleEvent(SipProviderImpl.java:185)
12894 at gov.nist.javax.sip.DialogFilter.processRequest(DialogFilter.java:1324)
12895 at gov.nist.javax.sip.stack.SIPServerTransactionImpl.processRequest(SIPServerTransactionImpl.java:834)
12896 at gov.nist.javax.sip.stack.UDPMessageChannel.processMessage(UDPMessageChannel.java:568)
12897 at gov.nist.javax.sip.stack.UDPMessageChannel.processIncomingDataPacket(UDPMessageChannel.java:514)
12898 at gov.nist.javax.sip.stack.UDPMessageChannel.run(UDPMessageChannel.java:319)
12899 at java.lang.Thread.run(Thread.java:745)

I create 5 rooms, try to send 20 INVITE for each room, using sipp, at rate 10.

Regards,
Wing

···

On May 4, 2017, at 8:18 PM, Damian Minkov <damencho@damencho.com<mailto:damencho@damencho.com>> wrote:

Do you have any logs?

On May 4, 2017 04:00, "Wing Chan" <WingChan@maaii.com<mailto:WingChan@maaii.com>> wrote:
Hi,

would like to know any performance report for Jigasi?
We face an issue that Jigasi start replying SIP 486 error after sending around 28 INVITEs to Jigasi.
We created 2 rooms, sending SIP INVITE at rate 10 INVITEs / second, and expect 20 participants join via Jigasi in each room.

Any idea what’s the potential cause?

Thanks,
Wing
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users mailing list
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#12

Hi,

I haven't seen this, it needs more debugging to identify what's wrong.
I would enable finest logging for net.sf.fmj.media.rtp package and
check the code, I see the problem is that a remoteAddress field is
null, and there is a log when this one is created ...

Regards
damencho

···

On Fri, May 5, 2017 at 1:56 AM, Wing Chan <WingChan@maaii.com> wrote:

Hi,

I see this NullPointerException in Jigasi:
SEVERE: [8616]
impl.protocol.sip.SipStackSharing.logApplicationException().1116 An error
occurred while processing event of type: javax.sip.DialogTerminatedEvent
12876 java.lang.NullPointerException
12877 at
net.sf.fmj.media.rtp.RTPSessionMgr.createSendStream(RTPSessionMgr.java:902)
12878 at
net.sf.fmj.media.rtp.RTPSessionMgr.createSendStream(RTPSessionMgr.java:985)
12879 at
org.jitsi.impl.neomedia.rtp.StreamRTPManager.createSendStream(StreamRTPManager.java:121)
12880 at
org.jitsi.impl.neomedia.MediaStreamImpl.createSendStreams(MediaStreamImpl.java:903)
12881 at
org.jitsi.impl.neomedia.MediaStreamImpl.startSendStreams(MediaStreamImpl.java:2971)
12882 at
org.jitsi.impl.neomedia.MediaStreamImpl.start(MediaStreamImpl.java:2860)
12883 at
org.jitsi.impl.neomedia.MediaStreamImpl.start(MediaStreamImpl.java:2823)
12884 at
net.java.sip.communicator.service.protocol.media.CallPeerMediaHandler.start(CallPeerMediaHandler.java:1948)
12885 at
net.java.sip.communicator.impl.protocol.sip.CallPeerMediaHandlerSipImpl.start(CallPeerMediaHandlerSipImpl.java:1909)
12886 at
net.java.sip.communicator.impl.protocol.sip.CallPeerSipImpl.processAck(CallPeerSipImpl.java:724)
12887 at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.processAck(OperationSetBasicTelephonySipImpl.java:1312)
12888 at
net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.processRequest(OperationSetBasicTelephonySipImpl.java:359)
12889 at
net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.processRequest(ProtocolProviderServiceSipImpl.java:1044)
12890 at
net.java.sip.communicator.impl.protocol.sip.SipStackSharing.processRequest(SipStackSharing.java:673)
12891 at
gov.nist.javax.sip.EventScanner.deliverRequestEvent(EventScanner.java:250)
12892 at
gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:146)
12893 at
gov.nist.javax.sip.SipProviderImpl.handleEvent(SipProviderImpl.java:185)
12894 at
gov.nist.javax.sip.DialogFilter.processRequest(DialogFilter.java:1324)
12895 at
gov.nist.javax.sip.stack.SIPServerTransactionImpl.processRequest(SIPServerTransactionImpl.java:834)
12896 at
gov.nist.javax.sip.stack.UDPMessageChannel.processMessage(UDPMessageChannel.java:568)
12897 at
gov.nist.javax.sip.stack.UDPMessageChannel.processIncomingDataPacket(UDPMessageChannel.java:514)
12898 at
gov.nist.javax.sip.stack.UDPMessageChannel.run(UDPMessageChannel.java:319)
12899 at java.lang.Thread.run(Thread.java:745)

I create 5 rooms, try to send 20 INVITE for each room, using sipp, at rate
10.

Regards,
Wing

On May 4, 2017, at 8:18 PM, Damian Minkov <damencho@damencho.com> wrote:

Do you have any logs?

On May 4, 2017 04:00, "Wing Chan" <WingChan@maaii.com> wrote:

Hi,

would like to know any performance report for Jigasi?
We face an issue that Jigasi start replying SIP 486 error after sending
around 28 INVITEs to Jigasi.
We created 2 rooms, sending SIP INVITE at rate 10 INVITEs / second, and
expect 20 participants join via Jigasi in each room.

Any idea what’s the potential cause?

Thanks,
Wing
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#13

Hi ,

Is the audio mixing done in Jigasi?
If so, is there any way I can set the mixing done in JVB?

Thanks,
Wing


#14

Hi,

Jigasi will mix the audio when bridging with SIP endpoints. The JVB does (or did?) support mixing, but it’s currently used in Jitsi Meet. What is your usage scenario?

Cheers,

···

On May 8, 2017, at 09:43, Wing Chan <WingChan@maaii.com> wrote:

Hi ,

Is the audio mixing done in Jigasi?
If so, is there any way I can set the mixing done in JVB?

--
Saúl


#15

In our case, we just need audio and we will not use any web client, we just use a XMPP client sending ConferenceIq to Jicofo in order to create room in Jitsi, XMPP clients are just responsible for signaling the users and monitoring.

The actual users will join the conference room via Jigasi. The audio mixing part is essential in our case, we prefer the transcoding and mixing logic performed in JVB and scale JVB if necessary. Is it possible?

Regards,
Wing

···

On May 8, 2017, at 5:12 PM, Saúl Ibarra Corretgé <scorretge@atlassian.com> wrote:

On May 8, 2017, at 09:43, Wing Chan <WingChan@maaii.com> wrote:

Hi ,

Is the audio mixing done in Jigasi?
If so, is there any way I can set the mixing done in JVB?

Hi,

Jigasi will mix the audio when bridging with SIP endpoints. The JVB does (or did?) support mixing, but it’s currently used in Jitsi Meet. What is your usage scenario?

Cheers,

--
Saúl

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#16

In our case, we just need audio and we will not use any web client, we just use a XMPP client sending ConferenceIq to Jicofo in order to create room in Jitsi, XMPP clients are just responsible for signaling the users and monitoring.

The actual users will join the conference room via Jigasi. The audio mixing part is essential in our case, we prefer the transcoding and mixing logic performed in JVB and scale JVB if necessary. Is it possible?

I’m not sure that’s possible today, Boris, maybe you can confirm? At any rate, we have make some great progress towards Jigasi scalability, so you could scale it up just like you scale the JVB up.

Cheers,

···

On May 8, 2017, at 11:19, Wing Chan <WingChan@maaii.com> wrote:

Regards,
Wing

On May 8, 2017, at 5:12 PM, Saúl Ibarra Corretgé <scorretge@atlassian.com> wrote:

On May 8, 2017, at 09:43, Wing Chan <WingChan@maaii.com> wrote:

Hi ,

Is the audio mixing done in Jigasi?
If so, is there any way I can set the mixing done in JVB?

Hi,

Jigasi will mix the audio when bridging with SIP endpoints. The JVB does (or did?) support mixing, but it’s currently used in Jitsi Meet. What is your usage scenario?

Cheers,

--
Saúl

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--
Saúl


#17

The bridge supports mixing, but jitsi-meet/jicofo doesn't use it (the needed modifications should be minor).

Boris

···

On 15/05/2017 06:22, Saúl Ibarra Corretgé wrote:

On May 8, 2017, at 11:19, Wing Chan <WingChan@maaii.com> wrote:

In our case, we just need audio and we will not use any web
client, we just use a XMPP client sending ConferenceIq to Jicofo
in order to create room in Jitsi, XMPP clients are just responsible
for signaling the users and monitoring.

The actual users will join the conference room via Jigasi. The
audio mixing part is essential in our case, we prefer the
transcoding and mixing logic performed in JVB and scale JVB if
necessary. Is it possible?

I’m not sure that’s possible today, Boris, maybe you can confirm? At
any rate, we have make some great progress towards Jigasi
scalability, so you could scale it up just like you scale the JVB
up.