Jitsi, Jigasi integration with asterisk:

Hi, I am trying to use video conferencing(sip) device to join a meeting on self hosted jitsi server. I have installed jigasi (on same machine where jitsi is hosted) and asterisk along with freepbx (on a different machine). Now the situation is, jigasi and sip device is registered through pjsip, can make a call to jigasi from sip device and connect to it (using default room).

But the problem is, no video is streamed from sip device to jisti and vice-versa. Sip device is displayed with phone icon in conference.

Following are the logs from jigasi:

2021-11-10 19:21:43.601 INFO: [97] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1683 [ctx=16365523014261935569167] SIP peer state: Connecting*
2021-11-10 19:21:43.605 INFO: [105] org.jitsi.jigasi.SipGatewaySession.handleCallState().1600 [ctx=16365523014261935569167] Sip call IN_PROGRESS: Call: id=1636552301394198235602 peers=1
2021-11-10 19:21:43.605 INFO: [105] org.jitsi.jigasi.SipGatewaySession.handleCallState().1609 [ctx=16365523014261935569167] SIP call format used: rtpmap:0 PCMU/8000
2021-11-10 19:21:43.606 INFO: [105] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1683 [ctx=16365523014261935569167] SIP peer state: Connected
2021-11-10 19:21:43.615 INFO: [105] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
2021-11-10 19:21:43.616 INFO: [97] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 126=rtpmap:-1 telephone-event/8000; 111=rtpmap:-1 opus/48000/2 fmtp:useinbandfec=1;minptime=10; 103=rtpmap:-1 unknown/90000;
2021-11-10 19:21:43.616 INFO: [97] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides [103->104 ]
2021-11-10 19:21:43.637 INFO: [97] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
2021-11-10 19:21:43.809 INFO: [114] org.jitsi.srtp.crypto.OpenSslWrapperLoader.log() jitsisrtp successfully loaded
2021-11-10 19:21:43.819 INFO: [97] org.jitsi.jigasi.JvbConference.callStateChanged().1530 [ctx=16365523014261935569167] JVB conference call IN_PROGRESS.
2021-11-10 19:21:43.820 INFO: [105] service.protocol.media.TransportManager.sendHolePunchPacket().552 Send NAT hole punch packets
2021-11-10 19:21:43.822 INFO: [106] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1003 Dynamic PT map: 101=rtpmap:-1 telephone-event/8000;
2021-11-10 19:21:43.822 INFO: [106] service.protocol.media.MediaHandler.registerDynamicPTsWithStream().1020 PT overrides
2021-11-10 19:21:43.830 INFO: [150] service.protocol.media.CallPeerMediaHandler.start().1961 Starting
2021-11-10 19:21:43.872 SEVERE: [184] net.sf.fmj.media.Log.error() Unable to handle format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
2021-11-10 19:21:43.873 SEVERE: [184] net.sf.fmj.media.Log.error() Failed to prefetch: net.sf.fmj.media.ProcessEngine@746b0147
2021-11-10 19:21:43.876 SEVERE: [180] net.sf.fmj.media.Log.error() Error: Unable to prefetch net.sf.fmj.media.ProcessEngine@746b0147

Following error is displayed on asterisl cli: (888 is extension used for jigasi)

[2021-11-10 13:51:41] WARNING[6636][C-0000002a]: chan_sip.c:23254 func_hea This function can only be used on SIP channels.
[2021-11-10 13:51:43] ERROR[21305]: res_pjsip_session.c:2058 resolve_refrestates: PJSIP/888-00000051: Attempting to remove stream 2:-2 but’t exist anywhere.
[2021-11-10 13:51:43] WARNING[21305]: res_pjsip_session.c:2299 sip_session PJSIP/888-00000051: Unable to merge media states

Can someone help in understanding the problem first and then troubleshooting it.

Thanks.

This is not possible through Jigasi. Jibri potentially has the capability to send SIP video but you’ll need to work through setting the feature up.

Thanks for replying @Freddie.

“But the problem is, no video is streamed from sip device to jisti and vice-versa. Sip device is displayed with phone icon in conference.”

What I meant with above statement was, I am not able to get video but audio is working fine in the conference. Sorry, that sentence made you think I was trying to stream the conference. I want to join conference through sip device.

Jigasi is audio-only and does not support video.

As @Freddie said: jibri supports adding sip video devices in a meeting, and its main features are recording and streaming.

Hi,sir.
Could you tell me how to use Jibri to support adding SIP video device?
I did not find documentation on this

We don’t have currently a doc describing that, I remember there was something simple in the forum somewhere …
We will work on adding a doc there, we just don’t have the resources right now …
The setup is like jibri, you just need to setup a second loop device for the audio and for the video and run a second Xserver.

It’s actually slightly different than standard jibri: the calls may need to be triggered via REST rather than xmpp. In addition to a new alsa audio loopback device, each jibri also needs two V4L loopback video devices, fed by two ffmpeg processes running in the background as services, providing the video for pjsua and chrome. Our custom branch of pjsua may also be required. So it’s not immediately easy to set up, but it’s not terrible. We also aren’t doing a lot of testing here currently, it’s an experimental mode still, although stable in most cases.

As per this tutorial from @Freddie: https://community.jitsi.org/t/tutorial-jibri-overview-troubleshooting-tips-tricks-solve-your-jibri-problems-quickly/86054.

Jibri is the recording and streaming instrument for Jitsi Meet conferences. Its capabilities include:

  1. Record a Jitsi meeting, capturing both audio and video
  2. Livestream a Jitsi meeting (through any RTMP)
  3. Do a local (audio) recording of a Jitsi meeting

My aim is neither to livestream nor record. I need to join a jitsi room and participate using a conventional video conferencing system(VCS) through SIP. After going through various post in this forum, I understood that we can use jigasi to get registered with a sip server and calling from the VCS can join a meeting(siptest - default). Now , I’m able to join the meeting but with only audio.
I am following this tutorial: Tutorial - Jitsi / Jigasi & FreePBX integration. Along with Asterisk IVR to use Jitsi conference mapper API.

As per suggestions from @Freddie and @damencho, jibri is also necessary for adding a sip device with video. Could you clarify if my understanding is correct?
Also any document for implementing this.

That post has been due an update for a while. This feature is actually not a function of Jibri.

But you’re correct on everything else. And your question has already been answered in the comments.

I saw in other subjects that if you want jibri to support SIP video access, you need to set up some important things in jicofo. Now,I have no idea to config Jibri or Jicofo.
Do you have any suggestions or demo about this?
https://community.jitsi.org/t/what-standard-vaules-are-required-for-sip-control-muc/79319
https://community.jitsi.org/t/is-it-possible-to-setup-jitsi-freeswitch-or-any-other-sip-server-multi-user-video-conference/22005/4