Re: [asterisk-users] ConfBridge user joining not getting video
On Thu, Jan 13, 2022 at 10:45 AM Jerry Geis wrote: > >> Hi Josh > > >chan_sip did not add a video stream. What is the actual configuration for > > it? What is the actual call file used for it? > > sip.conf has videosupport in the general section. > > I did find that where I am "joining" the person in the conference I did > not have the Codecs: set. I added that - doing better - its negotiating > video now - but still not showing me video for a conference. > What video mode are you using? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
On Thursday 13 January 2022 at 15:45:02, Jerry Geis wrote: > > Hi Josh > > > >chan_sip did not add a video stream. What is the actual configuration for > > > > it? What is the actual call file used for it? > > sip.conf has videosupport in the general section. > > I did find that where I am "joining" the person in the conference I did not > have the Codecs: set. I added that - doing better - its negotiating video > now - but still not showing me video for a conference. Can you make a 1-to-1 video call between two of the devices (which I assume does give you video?), and then get just those two to join a conference, and see the difference in SDP? Antony. -- The first fifty percent of an engineering project takes ninety percent of the time, and the remaining fifty percent takes another ninety percent of the time. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
> > > Hi Josh >chan_sip did not add a video stream. What is the actual configuration for > it? What is the actual call file used for it? sip.conf has videosupport in the general section. I did find that where I am "joining" the person in the conference I did not have the Codecs: set. I added that - doing better - its negotiating video now - but still not showing me video for a conference. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
On Thu, Jan 13, 2022 at 10:01 AM Jerry Geis wrote: > > > On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis wrote: > >> I am running 18.8.0 - videosupport is enabled. I get video calls no >> problem. >> >> However when I make a call file to a soft phone and include: >> Codecs: ulaw,h264 >> in the call file... >> >> sip show channels - shows: >> 4013c15f1f4cdff (ulaw|h264) No Tx: ACK >> so clearly the caller has h264. >> >> Then when I "automatically" request another softphone to join my conf >> bridge... >> the soft phone rings, and answers - all I get is audio and sip show >> channels for that device: >> 5c77cf1455e4afc (ulaw) No Tx: ACK >> >> How do I get Video in the confbridge ? >> >> Thanks >> >> Jerry >> > > > > hi Josh, > > here is the sip debug... It shows the the first call negotiate video - but > the second call to bring the end video device into the conf - no video > negotitation. > > Audio is at 15542 > Adding codec ulaw to SDP > Adding codec alaw to SDP > Adding codec gsm to SDP > chan_sip did not add a video stream. What is the actual configuration for it? What is the actual call file used for it? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis wrote: > I am running 18.8.0 - videosupport is enabled. I get video calls no > problem. > > However when I make a call file to a soft phone and include: > Codecs: ulaw,h264 > in the call file... > > sip show channels - shows: > 4013c15f1f4cdff (ulaw|h264) No Tx: ACK > so clearly the caller has h264. > > Then when I "automatically" request another softphone to join my conf > bridge... > the soft phone rings, and answers - all I get is audio and sip show > channels for that device: > 5c77cf1455e4afc (ulaw) No Tx: ACK > > How do I get Video in the confbridge ? > > Thanks > > Jerry > hi Josh, here is the sip debug... It shows the the first call negotiate video - but the second call to bring the end video device into the conf - no video negotitation. Audio is at 15542 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Thanks, Jerry Asterisk 18.8.0, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Running as user 'silentm' Running under group 'silentm' Connected to Asterisk 18.8.0 currently running on DevKaufer (pid = 597669) Really destroying SIP dialog 'c24843e8-d7f1-0740-08dd-8b79fe39a15a' Method: REGISTER <--- SIP read from UDP:192.168.2.22:5060 ---> <-> == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Audio is at 17816 Video is at 192.168.1.6:10746 Adding video codec vp8 to SDP Adding codec ulaw to SDP Adding codec opus to SDP Reliably Transmitting (NAT) to 192.168.1.6:48124: INVITE sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK73b689b9;rport Max-Forwards: 70 From: "Mason Kaufer 34" ;tag=as101db932 To: Contact: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 18.8.0 Date: Thu, 13 Jan 2022 13:46:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1106 v=0 o=root 1174630673 1174630673 IN IP4 192.168.1.6 s=Asterisk PBX 18.8.0 c=IN IP4 192.168.1.6 b=CT:5120 t=0 0 m=audio 17816 UDP/TLS/RTP/SAVPF 0 107 a=rtpmap:0 PCMU/8000 a=rtpmap:107 opus/48000/2 a=maxptime:60 a=ice-ufrag:4ff9bfd157a3896a6bc7f86d312dde00 a=ice-pwd:2c8b8f052875a1cd7096d71478ff3567 a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 17816 typ host a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 17817 typ host a=connection:new a=setup:passive a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF a=rtcp-mux a=sendrecv m=video 10746 UDP/TLS/RTP/SAVPF 100 a=ice-ufrag:0c2c1ae221c4578666475d5455d11e6f a=ice-pwd:7eadf35c40a33c2f2bd87431669b60b2 a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 10746 typ host a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 10747 typ host a=connection:new a=setup:passive a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF a=rtpmap:100 VP8/9 a=rtcp-fb:* ccm fir a=rtcp-mux a=sendrecv --- -- Called mason.kaufer.visualcampus <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34";tag=as101db932 To: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE Content-Length: 0 <-> --- (7 headers 0 lines) --- <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34";tag=as101db932 To: ;tag=HULiDWhvD78SNfAPBUqC Contact: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/mason.kaufer.visualcampus-004b is ringing > 0x7f8eac0141f0 -- Strict RTP learning after remote address set to: 192.168.1.6:56634 > 0x7f8eac00b800 -- Strict RTP learning after remote address set to: 192.168.1.6:32953 <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34";tag=as101db932 To: ;tag=HULiDWhvD78SNfAPBUqC Contact: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE Content-Type: application/sdp
Re: [asterisk-users] ConfBridge user joining not getting video
On Wed, Jan 12, 2022 at 6:09 PM Jerry Geis wrote: > I am running 18.8.0 - videosupport is enabled. I get video calls no > problem. > > However when I make a call file to a soft phone and include: > Codecs: ulaw,h264 > in the call file... > > sip show channels - shows: > 4013c15f1f4cdff (ulaw|h264) No Tx: ACK > so clearly the caller has h264. > > Then when I "automatically" request another softphone to join my conf > bridge... > the soft phone rings, and answers - all I get is audio and sip show > channels for that device: > 5c77cf1455e4afc (ulaw) No Tx: ACK > > How do I get Video in the confbridge ? > Have you looked at the actual SIP trace to see what is negotiated? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge user joining not getting video
I am running 18.8.0 - videosupport is enabled. I get video calls no problem. However when I make a call file to a soft phone and include: Codecs: ulaw,h264 in the call file... sip show channels - shows: 4013c15f1f4cdff (ulaw|h264) No Tx: ACK so clearly the caller has h264. Then when I "automatically" request another softphone to join my conf bridge... the soft phone rings, and answers - all I get is audio and sip show channels for that device: 5c77cf1455e4afc (ulaw) No Tx: ACK How do I get Video in the confbridge ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
