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Jerry Geis

2022-Jan-13 14:01 UTC

[asterisk-users] ConfBridge user joining not getting video

On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis <jerry.geis at gmail.com>
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 <markster at digium.com>
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 at
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" <sip:527 at 192.168.1.6>;tag=as101db932
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>
Contact: <sip:527 at 192.168.1.6:5060;transport=ws>
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 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 VP8/90000
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"<sip:527 at 192.168.1.6>;tag=as101db932
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 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"<sip:527 at 192.168.1.6>;tag=as101db932
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC
Contact: <sips:mason.kaufer.visualcampus at
df7jal23ls0d.invalid;transport=wss>
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 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:
<sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid;transport=wss>
    -- SIP/mason.kaufer.visualcampus-0000004b 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"<sip:527 at 192.168.1.6>;tag=as101db932
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC
Contact: <sips:mason.kaufer.visualcampus at
df7jal23ls0d.invalid;transport=wss>
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 192.168.1.6:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 1824
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER,
UPDATE

v=0
o=- 6700265476590515000 2 IN IP4 127.0.0.1
s=Cloudonix WebRTC Client - chrome
t=0 0
a=msid-semantic: WMS NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
m=audio 56634 UDP/TLS/RTP/SAVPF 0 107
c=IN IP4 192.168.1.6
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:505434299 1 udp 2122260223 192.168.1.6 56634 typ host
generation 0 network-id 1
a=ice-ufrag:tt5f
a=ice-pwd:t6HFMwvAhMBcsLzbNv6ZsBN7
a=ice-options:trickle
a=fingerprint:sha-256
BC:DF:CE:46:D5:23:0D:50:52:1D:9A:E8:5C:ED:66:B9:4D:8A:73:8C:83:3C:20:75:8E:BC:D5:19:A4:28:50:74
a=setup:active
a=mid:0
a=sendrecv
a=msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
eca112c3-1a46-4c88-8ab0-822a8db6f24e
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=ssrc:902560899 cname:g/TRvw9o4VgxF0Qi
a=ssrc:902560899 msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
eca112c3-1a46-4c88-8ab0-822a8db6f24e
a=ssrc:902560899 mslabel:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
a=ssrc:902560899 label:eca112c3-1a46-4c88-8ab0-822a8db6f24e
m=video 32953 UDP/TLS/RTP/SAVPF 100
c=IN IP4 192.168.1.6
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:505434299 1 udp 2122260223 192.168.1.6 32953 typ host
generation 0 network-id 1
a=ice-ufrag:iQuk
a=ice-pwd:V3WS4tt1M2TwSqCs+sNWzhXP
a=ice-options:trickle
a=fingerprint:sha-256
BC:DF:CE:46:D5:23:0D:50:52:1D:9A:E8:5C:ED:66:B9:4D:8A:73:8C:83:3C:20:75:8E:BC:D5:19:A4:28:50:74
a=setup:active
a=mid:1
a=sendrecv
a=msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
1ed3f295-431e-492d-a1a3-cfa356566ea1
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=ssrc:3172627963 cname:g/TRvw9o4VgxF0Qi
a=ssrc:3172627963 msid:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
1ed3f295-431e-492d-a1a3-cfa356566ea1
a=ssrc:3172627963 mslabel:NROTCuqCKE7FDYU2K1IROgw1nIijAhkwDhWE
a=ssrc:3172627963 label:1ed3f295-431e-492d-a1a3-cfa356566ea1

<------------->
--- (10 headers 44 lines) ---
Got SDP version 2 and unique parts [- 6700265476590515000 IN IP4 127.0.0.1]
Found RTP audio format 0
Found RTP audio format 107
Found audio description format PCMU for ID 0
Found audio description format opus for ID 107
Found RTP video format 100
Found video description format VP8 for ID 100
Capabilities: us - (ulaw|opus|vp8|h264), peer -
audio=(ulaw|opus)/video=(vp8)/text=(nothing), combined - (ulaw|opus|vp8)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.6:56634
Peer video RTP is at port 192.168.1.6:32953
sip_route_dump: route/path hop:
<sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid;transport=wss>
Transmitting (NAT) to 192.168.1.6:48124:
ACK sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid;transport=wss
SIP/2.0
Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK65afda31;rport
Max-Forwards: 70
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as101db932
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>;tag=HULiDWhvD78SNfAPBUqC
Contact: <sip:527 at 192.168.1.6:5060;transport=ws>
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 192.168.1.6:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0


---
    -- SIP/mason.kaufer.visualcampus-0000004b answered
    -- Executing [smvoice_callprogress at smvoice-dialout:1]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"0?smvoice-analog,s,1") in
new stack
    -- Executing [smvoice_callprogress at smvoice-dialout:2]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?smvoice_callprogress,4:smvoice_callprogress,3") in new stack
    -- Goto (smvoice-dialout,smvoice_callprogress,4)
    -- Executing [smvoice_callprogress at smvoice-dialout:4]
AGI("SIP/mason.kaufer.visualcampus-0000004b",
"smvoice,-digium_asterisk")
in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
       > 0x7f8eac00b800 -- Strict RTP learning after ICE completion
       > 0x7f8eac0141f0 -- Strict RTP learning after ICE completion
       > 0x7f8eac00b800 -- Strict RTP learning after remote address set to:
192.168.1.6:32953
       > 0x7f8eac0141f0 -- Strict RTP learning after remote address set to:
192.168.1.6:56634
       > 0x7f8eac0141f0 -- Strict RTP switching to RTP target address
192.168.1.6:56634 as source
       > 0x7f8eac00b800 -- Strict RTP switching to RTP target address
192.168.1.6:32953 as source
    -- <SIP/mason.kaufer.visualcampus-0000004b>AGI Script smvoice
completed, returning 0
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:1]
Set("SIP/mason.kaufer.visualcampus-0000004b",
"agi_use_meetme=0") in new
stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:2]
Set("SIP/mason.kaufer.visualcampus-0000004b",
"agi_use_confbridge=1") in
new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:3]
AGI("SIP/mason.kaufer.visualcampus-0000004b",
"smvoice,-digium_success,-pa_list") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
    -- <SIP/mason.kaufer.visualcampus-0000004b>AGI Script smvoice
completed, returning 0
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:4]
Wait("SIP/mason.kaufer.visualcampus-0000004b", "1") in new
stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:5]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "SETTING SPEAK
LIVE GAIN")
in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:6]
Set("SIP/mason.kaufer.visualcampus-0000004b",
"VOLUME(TX)=0") in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:7]
Set("SIP/mason.kaufer.visualcampus-0000004b",
"VOLUME(RX)=0") in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:8]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "START SPEAK LIVE
OPTIONS")
in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:9]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_delay")
in new stack
    -- Goto (smvoice-local-public-address,app_confbridge_call_out,11)
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:11]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "Skipped speak
live delay")
in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:12]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_preamble")
in new stack
    -- Goto (smvoice-local-public-address,app_confbridge_call_out,18)
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:18]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "Skipped speak
live
preamble") in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:19]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "END SPEAK LIVE
OPTIONS") in
new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:20]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "START SPEAK LIVE
BEEP") in
new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:21]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"0?smvoice-local-public-address,app_confbridge_call_out,skip_speak_live_beep")
in new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:22]
Playback("SIP/mason.kaufer.visualcampus-0000004b", "beep")
in new stack
    -- <SIP/mason.kaufer.visualcampus-0000004b> Playing 'beep.gsm'
(language 'en')
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:23]
NoOp("SIP/mason.kaufer.visualcampus-0000004b", "END SPEAK LIVE
BEEP") in
new stack
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:24]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?smvoice-local-public-address,app_confbridge_call_out,skip_record")
in
new stack
    -- Goto (smvoice-local-public-address,app_confbridge_call_out,26)
    -- Executing [app_confbridge_call_out at smvoice-local-public-address:26]
ConfBridge("SIP/mason.kaufer.visualcampus-0000004b",
"PA0003,LayeredSolutionsConfBridge,LayeredSolutionsConfUser") in new
stack
    -- Channel CBAnn/PA0003-00000ae7;2 joined 'softmix' base-bridge
<920968b7-3db6-4d15-ab7f-f123d585d98e>
    -- Channel SIP/mason.kaufer.visualcampus-0000004b joined 'softmix'
base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e>
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 15542
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.102:5063:
INVITE sip:5124 at 192.168.1.102:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf
Max-Forwards: 70
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>
Contact: <sip:527 at 192.168.1.6:5060>
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.8.0
Date: Thu, 13 Jan 2022 13:46:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Alert-Info: Ring Answer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1569896537 1569896537 IN IP4 192.168.1.6
s=Asterisk PBX 18.8.0
c=IN IP4 192.168.1.6
t=0 0
m=audio 15542 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called 5124

<--- SIP read from UDP:192.168.1.102:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 102 INVITE
User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.102:5063 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>;tag=735442138
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 102 INVITE
Contact: <sip:5124 at 192.168.1.102:5063>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106
Allow-Events: talk,hold,conference,refer,check-sync
P-Asserted-Identity: "5124"<sip:5124 at 192.168.1.6>
Privacy: none
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:5124 at 192.168.1.102:5063>
    -- SIP/5124-0000004c is ringing

<--- SIP read from UDP:192.168.1.102:5063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4fa302bf
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>;tag=735442138
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 102 INVITE
Contact: <sip:5124 at 192.168.1.102:5063>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106
Content-Length:   210

v=0
o=5124 5000 5000 IN IP4 192.168.1.102
s=Talk
c=IN IP4 192.168.1.102
t=0 0
m=audio 11868 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
--- (11 headers 11 lines) ---
Got SDP version 5000 and unique parts [5124 5000 IN IP4 192.168.1.102]
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (h264|ulaw|alaw|gsm|vp8), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f8f400292c0 -- Strict RTP learning after remote address set to:
192.168.1.102:11868
Peer audio RTP is at port 192.168.1.102:11868
sip_route_dump: route/path hop: <sip:5124 at 192.168.1.102:5063>
set_destination: Parsing <sip:5124 at 192.168.1.102:5063> for address/port
to
send to
set_destination: set destination to 192.168.1.102:5063
Transmitting (no NAT) to 192.168.1.102:5063:
ACK sip:5124 at 192.168.1.102:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK18a38671
Max-Forwards: 70
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>;tag=735442138
Contact: <sip:527 at 192.168.1.6:5060>
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0


---
    -- SIP/5124-0000004c answered
    -- Executing [smvoice_pa_app_confbridge_twoway at smvoice-transfers:1]
GotoIf("SIP/5124-0000004c", "1?skip_dtmf:use_dtmf") in new
stack
    -- Goto (smvoice-transfers,smvoice_pa_app_confbridge_twoway,4)
    -- Executing [smvoice_pa_app_confbridge_twoway at smvoice-transfers:4]
ConfBridge("SIP/5124-0000004c",
"PA0003,LayeredSolutionsConfBridge,LayeredSolutionsConfUser") in new
stack
    -- Channel SIP/5124-0000004c joined 'softmix' base-bridge
<920968b7-3db6-4d15-ab7f-f123d585d98e>
Reliably Transmitting (NAT) to 192.168.1.6:48124:
OPTIONS sips:mason.kaufer.visualcampus at
df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss
SIP/2.0
Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK3c0c1808;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.1.6>;tag=as0ceb5b65
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>
Contact: <sip:asterisk at 192.168.1.6:5060;transport=ws>
Call-ID: 3f760a323f6d68fa68274e1c6512bfa4 at 192.168.1.6:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.8.0
Date: Thu, 13 Jan 2022 13:46:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from WS:192.168.1.6:48124 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK3c0c1808
From: "asterisk"<sip:asterisk at 192.168.1.6>;tag=as0ceb5b65
To: <sips:mason.kaufer.visualcampus at df7jal23ls0d.invalid
;rtcweb-breaker=yes;transport=wss>
Call-ID: 3f760a323f6d68fa68274e1c6512bfa4 at 192.168.1.6:5060
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
       > 0x7f8f400292c0 -- Strict RTP switching to RTP target address
192.168.1.102:11868 as source
       > 0x7f8eac00b800 -- Strict RTP learning complete - Locking on source
address 192.168.1.6:32953
       > 0x7f8eac0141f0 -- Strict RTP learning complete - Locking on source
address 192.168.1.6:56634
Really destroying SIP dialog '
3f760a323f6d68fa68274e1c6512bfa4 at 192.168.1.6:5060' Method: OPTIONS
       > 0x7f8f400292c0 -- Strict RTP learning complete - Locking on source
address 192.168.1.102:11868

<--- SIP read from WS:192.168.1.6:48124 --->
BYE sip:527 at 192.168.1.6:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKXZvId1AmTWItdTSFURnuLNJdEg8esTwa;rport
From: <sips:mason.kaufer.visualcampus at
df7jal23ls0d.invalid>;tag=HULiDWhvD78SNfAPBUqCTo: "Mason Kaufer 34"<sip:527 at 192.168.1.6>;tag=as101db932
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 192.168.1.6:5060
CSeq: 60034 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom


<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog '
4c4ab96b31b9fa52416eacf563ae2bf1 at 192.168.1.6:5060' in 6400 ms (Method:
BYE)

<--- Transmitting (NAT) to 192.168.1.6:48124 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;branch=z9hG4bKXZvId1AmTWItdTSFURnuLNJdEg8esTwa;received=192.168.1.6;rport=48124
From: <sips:mason.kaufer.visualcampus at
df7jal23ls0d.invalid>;tag=HULiDWhvD78SNfAPBUqCTo: "Mason Kaufer 34"<sip:527 at 192.168.1.6>;tag=as101db932
Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1 at 192.168.1.6:5060
CSeq: 60034 BYE
Server: Asterisk PBX 18.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/mason.kaufer.visualcampus-0000004b left 'softmix'
base-bridge <920968b7-3db6-4d15-ab7f-f123d585d98e>
    -- Executing [h at smvoice-local-public-address:1]
NoOp("SIP/mason.kaufer.visualcampus-0000004b",
"agi_pa_meetme=PA0003
agi_use_meetme0 agi_use_confbridge=1") in new stack
    -- Executing [h at smvoice-local-public-address:2]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?h_app_conference,1") in
new stack
    -- Goto (smvoice-local-public-address,h_app_conference,1)
    -- Executing [h_app_conference at smvoice-local-public-address:1]
GotoIf("SIP/mason.kaufer.visualcampus-0000004b",
"1?h_app_confbridge,1") in
new stack
    -- Goto (smvoice-local-public-address,h_app_confbridge,1)
    -- Executing [h_app_confbridge at smvoice-local-public-address:1]
AGI("SIP/mason.kaufer.visualcampus-0000004b",
"smvoice,-digium_success,-pa_done,241,PA0003") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
    -- Channel CBAnn/PA0003-00000ae7;2 left 'softmix' base-bridge
<920968b7-3db6-4d15-ab7f-f123d585d98e>
    -- Executing [h at smvoice-transfers:1]
GotoIf("SIP/5124-0000004c",
"1?h,s,4") in new stack
    -- Goto (h,s,4)
Scheduling destruction of SIP dialog '
240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060' in 32000 ms (Method:
INVITE)
[Jan 13 08:46:28] NOTICE[1097928]: manager.c:4499 action_hangup: Request to
hangup non-existent channel: SIP/5124-0000004c
set_destination: Parsing <sip:5124 at 192.168.1.102:5063> for address/port
to
send to
set_destination: set destination to 192.168.1.102:5063
Reliably Transmitting (no NAT) to 192.168.1.102:5063:
BYE sip:5124 at 192.168.1.102:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6e2de9fa
Max-Forwards: 70
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>;tag=735442138
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 18.8.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.102:5063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6e2de9fa
From: "Mason Kaufer 34" <sip:527 at 192.168.1.6>;tag=as3729f342
To: <sip:5124 at 192.168.1.102:5063>;tag=735442138
Call-ID: 240dde9e3a94137916b97b2d60264c53 at 192.168.1.6:5060
CSeq: 103 BYE
User-Agent: TOA N-SP80VS1 21.192.1.167 0005F9300106
Content-Length: 0
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Joshua C. Colp

2022-Jan-13 14:12 UTC

head link

[asterisk-users] ConfBridge user joining not getting video

On Thu, Jan 13, 2022 at 10:01 AM Jerry Geis <jerry.geis at gmail.com>
wrote:
>
>
> On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis <jerry.geis at gmail.com>
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
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Jerry Geis

2022-Jan-13 14:45 UTC

head link

[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
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asterisk users - Jan 2022 - ConfBridge user joining not getting video

[asterisk-users] ConfBridge user joining not getting video

[asterisk-users] ConfBridge user joining not getting video

[asterisk-users] ConfBridge user joining not getting video