[jitsi-dev] SIP calls out using Jitsu-Meet interrupt conference RTP


#1

Hi,
Just installed the latest jitsi-meet nightly build including Jigasi support. All are installed on the same Debian server.
If I start a conference between two web clients and then add a SIP client to the conference (call out to the client) the conference audio & RTP will stop. Wireshark running on the debian machine shows this as well.
The SIP connection will show connected on the jitsi-meet UI but there is still no audio. If the SIP call is terminated, audio between the two connected web clients will be restored.

Previously, SIP out calls from Jigasi worked..
Thx,
Tom


#2

I should add that the working version is running on a VM while this new version is on a stand-alone server

  working version:
    root@jitsi:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-160-1 amd64 JItsi Meet COnference FOcus
    ii jigasi 1.0-91 amd64 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.711-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.711-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 549-1 amd64 WebRTC compatible Selective Forwarding Unit (SFU)

  not working:
    root@conference:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-197-1 i386 JItsi Meet COnference FOcus
    ii jigasi 1.0-90 i386 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.803-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.803-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 627-1 i386 WebRTC compatible Selective Forwarding Unit (SFU)

···

On 1 Feb 2016, at 6:14 PM, Tom McGuinness <tommcguinness@maaii.com> wrote:

Hi,
Just installed the latest jitsi-meet nightly build including Jigasi support. All are installed on the same Debian server.
If I start a conference between two web clients and then add a SIP client to the conference (call out to the client) the conference audio & RTP will stop. Wireshark running on the debian machine shows this as well.
The SIP connection will show connected on the jitsi-meet UI but there is still no audio. If the SIP call is terminated, audio between the two connected web clients will be restored.

Previously, SIP out calls from Jigasi worked..
Thx,
Tom


#3

Hi Tom,

I think I experiences the same issue since a while but I'm not able since when this issue is occurring.

Best regards
Christoph

···

Am 01.02.2016 um 11:52 schrieb Tom McGuinness:

I should add that the working version is running on a VM while this new version is on a stand-alone server

  working version:
    root@jitsi:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-160-1 amd64 JItsi Meet COnference FOcus
    ii jigasi 1.0-91 amd64 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.711-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.711-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 549-1 amd64 WebRTC compatible Selective Forwarding Unit (SFU)

  not working:
    root@conference:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-197-1 i386 JItsi Meet COnference FOcus
    ii jigasi 1.0-90 i386 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.803-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.803-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 627-1 i386 WebRTC compatible Selective Forwarding Unit (SFU)
  

On 1 Feb 2016, at 6:14 PM, Tom McGuinness <tommcguinness@maaii.com> wrote:

Hi,
Just installed the latest jitsi-meet nightly build including Jigasi support. All are installed on the same Debian server.
If I start a conference between two web clients and then add a SIP client to the conference (call out to the client) the conference audio & RTP will stop. Wireshark running on the debian machine shows this as well.
The SIP connection will show connected on the jitsi-meet UI but there is still no audio. If the SIP call is terminated, audio between the two connected web clients will be restored.

Previously, SIP out calls from Jigasi worked..
Thx,
Tom

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#4

Digging a bit deeper through the JVB logs today. On the failing system I see this log at the end of ICE:

2016-02-02 10:47:43.223 INFO: [58] org.ice4j.ice.harvest.SinglePortUdpHarvester.runInHarvesterThread() Failed to handle new socket: java.io.IOException: Agent state is Completed. Cannot add socket.
2016-02-02 10:47:44.843 INFO: [153] org.ice4j.ice.Agent.setState() ICE state changed from Completed to Terminated
2016-02-02 10:47:44.844 INFO: [153] org.jitsi.videobridge.IceUdpTransportManager.info() ICE processing state of IceUdpTransportManager #32dac4 (for channels a60c7fedb9b10438 ddc9aad392fee31e 49bcf1649be36417) of conference b65c2564bd57569 changed from Completed to Terminated.
2016-02-02 10:47:45.687 INFO: [177] org.ice4j.ice.Agent.setState() ICE state changed from Completed to Terminated
2016-02-02 10:47:45.687 INFO: [177] org.jitsi.videobridge.IceUdpTransportManager.info() ICE processing state of IceUdpTransportManager #6b1cdc (for channels dffdd9820ed8b16f) of conference b65c2564bd57569 changed from Completed to Terminated.
2016-02-02 10:47:47.888 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.908 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.928 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.948 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.

The "Will not translate RTP packet” are printed at about 50/sec ..

Any ideas why socket creation fails inside ice4j?

Thanks,
Tom

···

On 1 Feb 2016, at 6:52 PM, Tom McGuinness <tommcguinness@maaii.com> wrote:

I should add that the working version is running on a VM while this new version is on a stand-alone server

  working version:
    root@jitsi:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-160-1 amd64 JItsi Meet COnference FOcus
    ii jigasi 1.0-91 amd64 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.711-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.711-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 549-1 amd64 WebRTC compatible Selective Forwarding Unit (SFU)

  not working:
    root@conference:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
    ii jicofo 1.0-197-1 i386 JItsi Meet COnference FOcus
    ii jigasi 1.0-90 i386 Jitsi Gateway for SIP
    ii jitsi-meet 1.0.803-1 all WebRTC JavaScript video conferences
    ii jitsi-meet-prosody 1.0.803-1 all Prosody configuration for Jitsi Meet
    ii jitsi-videobridge 627-1 i386 WebRTC compatible Selective Forwarding Unit (SFU)
  

On 1 Feb 2016, at 6:14 PM, Tom McGuinness <tommcguinness@maaii.com> wrote:

Hi,
Just installed the latest jitsi-meet nightly build including Jigasi support. All are installed on the same Debian server.
If I start a conference between two web clients and then add a SIP client to the conference (call out to the client) the conference audio & RTP will stop. Wireshark running on the debian machine shows this as well.
The SIP connection will show connected on the jitsi-meet UI but there is still no audio. If the SIP call is terminated, audio between the two connected web clients will be restored.

Previously, SIP out calls from Jigasi worked..
Thx,
Tom


#5

I am not alone: https://github.com/jitsi/jitsi-videobridge/issues/124

I downgraded jitsi-videobridge to 527-1 (latest stable version) and audio for SIP calls is working again.

Latest nightly version still has this problem.

···

On 2 Feb 2016, at 11:42 AM, Tom McGuinness <tommcguinness@maaii.com<mailto:tommcguinness@maaii.com>> wrote:

Digging a bit deeper through the JVB logs today. On the failing system I see this log at the end of ICE:

2016-02-02 10:47:43.223 INFO: [58] org.ice4j.ice.harvest.SinglePortUdpHarvester.runInHarvesterThread() Failed to handle new socket: java.io.IOException: Agent state is Completed. Cannot add socket.
2016-02-02 10:47:44.843 INFO: [153] org.ice4j.ice.Agent.setState() ICE state changed from Completed to Terminated
2016-02-02 10:47:44.844 INFO: [153] org.jitsi.videobridge.IceUdpTransportManager.info<http://org.jitsi.videobridge.iceudptransportmanager.info>() ICE processing state of IceUdpTransportManager #32dac4 (for channels a60c7fedb9b10438 ddc9aad392fee31e 49bcf1649be36417) of conference b65c2564bd57569 changed from Completed to Terminated.
2016-02-02 10:47:45.687 INFO: [177] org.ice4j.ice.Agent.setState() ICE state changed from Completed to Terminated
2016-02-02 10:47:45.687 INFO: [177] org.jitsi.videobridge.IceUdpTransportManager.info<http://org.jitsi.videobridge.iceudptransportmanager.info>() ICE processing state of IceUdpTransportManager #6b1cdc (for channels dffdd9820ed8b16f) of conference b65c2564bd57569 changed from Completed to Terminated.
2016-02-02 10:47:47.888 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.908 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.928 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.
2016-02-02 10:47:47.948 WARNING: [156] org.jitsi.impl.neomedia.rtp.translator.OutputDataStreamImpl.warn() Will not translate RTP packet.

The "Will not translate RTP packet” are printed at about 50/sec ..

Any ideas why socket creation fails inside ice4j?

Thanks,
Tom

On 1 Feb 2016, at 6:52 PM, Tom McGuinness <tommcguinness@maaii.com<mailto:tommcguinness@maaii.com>> wrote:

I should add that the working version is running on a VM while this new version is on a stand-alone server

working version:
root@jitsi:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
ii jicofo 1.0-160-1 amd64 JItsi Meet COnference FOcus
ii jigasi 1.0-91 amd64 Jitsi Gateway for SIP
ii jitsi-meet 1.0.711-1 all WebRTC JavaScript video conferences
ii jitsi-meet-prosody 1.0.711-1 all Prosody configuration for Jitsi Meet
ii jitsi-videobridge 549-1 amd64 WebRTC compatible Selective Forwarding Unit (SFU)

not working:
root@conference:/home/mcguinn# dpkg-query -l | grep -i 'jitsi'
ii jicofo 1.0-197-1 i386 JItsi Meet COnference FOcus
ii jigasi 1.0-90 i386 Jitsi Gateway for SIP
ii jitsi-meet 1.0.803-1 all WebRTC JavaScript video conferences
ii jitsi-meet-prosody 1.0.803-1 all Prosody configuration for Jitsi Meet
ii jitsi-videobridge 627-1 i386 WebRTC compatible Selective Forwarding Unit (SFU)

On 1 Feb 2016, at 6:14 PM, Tom McGuinness <tommcguinness@maaii.com<mailto:tommcguinness@maaii.com>> wrote:

Hi,
Just installed the latest jitsi-meet nightly build including Jigasi support. All are installed on the same Debian server.
If I start a conference between two web clients and then add a SIP client to the conference (call out to the client) the conference audio & RTP will stop. Wireshark running on the debian machine shows this as well.
The SIP connection will show connected on the jitsi-meet UI but there is still no audio. If the SIP call is terminated, audio between the two connected web clients will be restored.

Previously, SIP out calls from Jigasi worked..
Thx,
Tom