Sound issues with Dahua VTO/VTH connected on freepbx 17

fourat · September 20, 2024, 9:02pm

Hello,

I’ve been trying for some days to connect my Dahua’s VTO-2211g (door ring) and Dahua’s VTH (screen) through freepbx17 with no success so far.

Here’s my configuration:

Freepbx: 10.0.2.16
Dahua’s VTO: 10.0.2.99, with extension 8001
Dahua’s VTH: 10.0.2.98, with extension 8011

Test scenarios:

When I call VTO from VTH I hear scratching sound, It’s like a codec negociation issue.
When I call VTO from a Jitsi (extension 100), sound is good !
When I call VTH from the Jitsi, I hear the same scratching sound.

I’m struggling to get the correct configuration, although this guy made it work on freepbx on first try: https://www.youtube.com/watch?v=6eN4Kn1BX3A !

Here’s the log from the last call scenario (Jitsi → VTH):

<--- Received SIP request (1040 bytes) from UDP:10.0.0.253:53884 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjb-BAUg8JzJZkYYYD95uNP3RFIRcKkmRr
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: sip:[email protected]
Contact: <sip:[email protected]:53884;ob>
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
CSeq: 5270 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 3935852248 3935852248 IN IP4 10.0.0.253
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4052 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.0.253
b=TIAS:96000
a=rtcp:4053 IN IP4 10.0.0.253
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:10762402 cname:35c77fdc3c9b6019

<--- Transmitting SIP response (557 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjb-BAUg8JzJZkYYYD95uNP3RFIRcKkmRr
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>;tag=z9hG4bKPjb-BAUg8JzJZkYYYD95uNP3RFIRcKkmRr
CSeq: 5270 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1726863449/db6eedf0dd9779aebf54ac6d5d1a171a",opaque="4e1e83ed37cb6a07",algorithm=MD5,qop="auth"
Server: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


<--- Received SIP request (371 bytes) from UDP:10.0.0.253:53884 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjb-BAUg8JzJZkYYYD95uNP3RFIRcKkmRr
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: sip:[email protected];tag=z9hG4bKPjb-BAUg8JzJZkYYYD95uNP3RFIRcKkmRr
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
CSeq: 5270 ACK
Content-Length:  0


<--- Received SIP request (1326 bytes) from UDP:10.0.0.253:53884 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: sip:[email protected]
Contact: <sip:[email protected]:53884;ob>
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
CSeq: 5271 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Authorization: Digest username="100", realm="asterisk", nonce="1726863449/db6eedf0dd9779aebf54ac6d5d1a171a", uri="sip:[email protected]", response="d1308a04ff378eaeeae749282be725b3", algorithm=MD5, cnonce="ywsOnCfhqTslIQCmqrGNbbfYGW7JBo", opaque="4e1e83ed37cb6a07", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 3935852248 3935852248 IN IP4 10.0.0.253
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4052 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.0.253
b=TIAS:96000
a=rtcp:4053 IN IP4 10.0.0.253
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:10762402 cname:35c77fdc3c9b6019

<--- Transmitting SIP response (359 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>
CSeq: 5271 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [8011@from-internal:1] GotoIf("PJSIP/100-0000004f", "0?ext-local,*8011,1") in new stack
    -- Executing [8011@from-internal:2] GotoIf("PJSIP/100-0000004f", "1?ext-local,8011,1:followme-check,8011,1") in new stack
    -- Goto (ext-local,8011,1)
    -- Executing [8011@ext-local:1] Set("PJSIP/100-0000004f", "__RINGTIMER=15") in new stack
    -- Executing [8011@ext-local:2] ExecIf("PJSIP/100-0000004f", "0?Set(__CWIGNORE=)") in new stack
    -- Executing [8011@ext-local:3] Gosub("PJSIP/100-0000004f", "macro-exten-vm,s,1(novm,8011,0,0,0)") in new stack
    -- Executing [s@macro-exten-vm:1] Gosub("PJSIP/100-0000004f", "macro-user-callerid,s,1()") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/100-0000004f", "TOUCH_MONITOR=1726863449.113") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/100-0000004f", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:3] Progress("PJSIP/100-0000004f", "") in new stack
<--- Transmitting SIP response (823 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>;tag=3283a8f0-2ca6-44a5-bc7c-1e4afeedcfd3
CSeq: 5271 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Contact: <sip:10.0.2.16:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3935852248 3935852250 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15810 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Executing [s@macro-user-callerid:4] Set("PJSIP/100-0000004f", "CHANCONTEXT=") in new stack
    ... stripped for brevity ...
    -- Executing [s@func-apply-sipheaders:9] While("PJSIP/8011-00000050", "0") in new stack
    -- Jumping to priority 15
    -- Executing [s@func-apply-sipheaders:16] Return("PJSIP/8011-00000050", "") in new stack
  == Spawn extension (from-internal, 8011, 1) exited non-zero on 'PJSIP/8011-00000050'
    -- PJSIP/8011-00000050 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
<--- Transmitting SIP request (1164 bytes) to UDP:10.0.2.98:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.16:5060;rport;branch=z9hG4bKPj2de2138d-8dc4-497e-adee-4a3e44638b36
From: "Sebastien CEF (laptop)" <sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
CSeq: 27609 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Sebastien CEF (laptop)" <sip:[email protected]>
Max-Forwards: 70
User-Agent: FPBX-17.0.19.11(21.4.3)
Content-Type: application/sdp
Content-Length:   418

v=0
o=- 1478656090 1478656090 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15562 RTP/AVP 0 10 111 4 3 118 101 102
a=rtpmap:0 PCMU/8000
a=rtpmap:10 L16/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/16000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

    -- Called PJSIP/8011/sip:[email protected]:5060
<--- Transmitting SIP response (904 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>;tag=3283a8f0-2ca6-44a5-bc7c-1e4afeedcfd3
CSeq: 5271 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
P-Asserted-Identity: "First floor ring screen (Available)" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3935852248 3935852250 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15810 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (376 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 100 Trying
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
Content-Length: 0
CSeq: 27609 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
To: <sip:[email protected]>
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPj2de2138d-8dc4-497e-adee-4a3e44638b36


<--- Received SIP response (463 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 101 Dialog Establishment
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 27609 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
To: <sip:[email protected]>;tag=bc4f9ca0194134ae5d8ca5964888c7ed
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPj2de2138d-8dc4-497e-adee-4a3e44638b36


<--- Received SIP response (601 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 180 Ringing
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 27609 INVITE
DependentInfo: 10.0.2.99
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
LeaveType: FTP
MaxConnectingTime: 300
MaxLeaveWordTime: 30
MaxRingingTime: 45
ShortNumber: 8011
To: <sip:[email protected]>;tag=bc4f9ca0194134ae5d8ca5964888c7ed
TransMode: SupportRTSP
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPj2de2138d-8dc4-497e-adee-4a3e44638b36


    -- PJSIP/8011-00000050 is ringing
<--- Transmitting SIP response (892 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>;tag=3283a8f0-2ca6-44a5-bc7c-1e4afeedcfd3
CSeq: 5271 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
P-Asserted-Identity: "First floor ring screen" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3935852248 3935852250 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15810 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP request (414 bytes) to UDP:10.0.0.253:53884 --->
OPTIONS sip:[email protected]:53884;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.2.16:5060;rport;branch=z9hG4bKPjc1bd9b8f-b47a-4358-aed1-22c492622389
From: <sip:[email protected]>;tag=199e5bb5-2b6b-4dfc-a174-f30d5d66becd
To: <sip:[email protected];ob>
Contact: <sip:[email protected]:5060>
Call-ID: 55b6b31c-9b31-4d90-bbff-c66e2bf13d7d
CSeq: 36822 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


<--- Received SIP response (769 bytes) from UDP:10.0.0.253:53884 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;received=10.0.2.16;branch=z9hG4bKPjc1bd9b8f-b47a-4358-aed1-22c492622389
Call-ID: 55b6b31c-9b31-4d90-bbff-c66e2bf13d7d
From: <sip:[email protected]>;tag=199e5bb5-2b6b-4dfc-a174-f30d5d66becd
To: <sip:[email protected];ob>;tag=z9hG4bKPjc1bd9b8f-b47a-4358-aed1-22c492622389
CSeq: 36822 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Telephone 1.6
Content-Length:  0


<--- Received SIP response (787 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 200 OK
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
Contact: <sip:[email protected]:5060>
Content-Length: 309
Content-Type: application/sdp
CSeq: 27609 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
To: <sip:[email protected]>;tag=bc4f9ca0194134ae5d8ca5964888c7ed
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPj2de2138d-8dc4-497e-adee-4a3e44638b36

v=0
o=- 1726863458 3 IN IP4 10.0.2.98
s=Dahua VT 1.5
c=IN IP4 10.0.2.98
t=0 0
m=audio 20000 RTP/AVP 101 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 PCM/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=recvonly

<--- Transmitting SIP request (426 bytes) to UDP:10.0.2.98:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.16:5060;rport;branch=z9hG4bKPj05601982-e7f8-466a-93cd-fe5acf394c62
From: "Sebastien CEF (laptop)" <sip:[email protected]>;tag=10284737-3064-4b37-a61a-af0dcfc96a4f
To: <sip:[email protected]>;tag=bc4f9ca0194134ae5d8ca5964888c7ed
Call-ID: dc6b9069-2c2e-44c0-a49b-77476a1f9765
CSeq: 27609 ACK
Max-Forwards: 70
User-Agent: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


    -- PJSIP/8011-00000050 answered PJSIP/100-0000004f
<--- Transmitting SIP response (926 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjx7T5T35wmgKdF4lwWswET7OpMfWAGKPD
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: <sip:[email protected]>;tag=3283a8f0-2ca6-44a5-bc7c-1e4afeedcfd3
CSeq: 5271 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "First floor ring screen" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 3935852248 3935852250 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15810 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Channel PJSIP/8011-00000050 joined 'simple_bridge' basic-bridge <1ae6dc27-a122-45e9-b4c3-995a96b047de>
    -- Channel PJSIP/100-0000004f joined 'simple_bridge' basic-bridge <1ae6dc27-a122-45e9-b4c3-995a96b047de>
<--- Received SIP request (366 bytes) from UDP:10.0.0.253:53884 --->
ACK sip:10.0.2.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjGHFTMrZR.XOhKWP4-0Sj7kjjtaoFz0Ov
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=.1S3JDxTk1Q1RHEUBJZAfDbkV7VqXOoC
To: sip:[email protected];tag=3283a8f0-2ca6-44a5-bc7c-1e4afeedcfd3
Call-ID: tvUORbr78K4JRKVi05jCwFtgJQAectAh
CSeq: 5271 ACK
Content-Length:  0


<--- Received SIP request (393 bytes) from UDP:10.0.0.253:53884 --->
BYE sip:10.0.2.16:5060 SIP/2.0
... stripped for brevity ...

dicko · September 20, 2024, 10:49pm

I suggest you limit your codecs to ulaw or alaw ONLY to test.

fourat · September 21, 2024, 9:42am

Ok, so kept ulaw/alaw for audio and h264 for video, still scratching sound:

<--- Received SIP request (1042 bytes) from UDP:10.0.0.253:53884 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjj4GY7Uus4e1oSmEFAnszkLLX0..pnqjl
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: sip:[email protected]
Contact: <sip:[email protected]:53884;ob>
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
CSeq: 5233 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Content-Type: application/sdp
Content-Length:   471

v=0
o=- 3935900140 3935900140 IN IP4 10.0.0.253
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4058 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.0.253
b=TIAS:96000
a=rtcp:4059 IN IP4 10.0.0.253
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:1539216976 cname:62cfb9b933aa535d

<--- Transmitting SIP response (557 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjj4GY7Uus4e1oSmEFAnszkLLX0..pnqjl
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>;tag=z9hG4bKPjj4GY7Uus4e1oSmEFAnszkLLX0..pnqjl
CSeq: 5233 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1726911340/5cac45de328ef71d460eb4fde41d342b",opaque="5c3d32d4062fcc68",algorithm=MD5,qop="auth"
Server: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


<--- Received SIP request (371 bytes) from UDP:10.0.0.253:53884 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjj4GY7Uus4e1oSmEFAnszkLLX0..pnqjl
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: sip:[email protected];tag=z9hG4bKPjj4GY7Uus4e1oSmEFAnszkLLX0..pnqjl
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
CSeq: 5233 ACK
Content-Length:  0


<--- Received SIP request (1329 bytes) from UDP:10.0.0.253:53884 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: sip:[email protected]
Contact: <sip:[email protected]:53884;ob>
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
CSeq: 5234 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Authorization: Digest username="100", realm="asterisk", nonce="1726911340/5cac45de328ef71d460eb4fde41d342b", uri="sip:[email protected]", response="ef8751099a1f4d35694eb7b777ecbb22", algorithm=MD5, cnonce="qd4guqDx0PJXlZ2JWHVAm3FSfSDbdPC", opaque="5c3d32d4062fcc68", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   471

v=0
o=- 3935900140 3935900140 IN IP4 10.0.0.253
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4058 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.0.253
b=TIAS:96000
a=rtcp:4059 IN IP4 10.0.0.253
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:1539216976 cname:62cfb9b933aa535d

<--- Transmitting SIP response (359 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>
CSeq: 5234 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [8011@from-internal:1] GotoIf("PJSIP/100-00000055", "0?ext-local,*8011,1") in new stack
    -- Executing [8011@from-internal:2] GotoIf("PJSIP/100-00000055", "1?ext-local,8011,1:followme-check,8011,1") in new stack
    -- Goto (ext-local,8011,1)
    -- Executing [8011@ext-local:1] Set("PJSIP/100-00000055", "__RINGTIMER=15") in new stack
    -- Executing [8011@ext-local:2] ExecIf("PJSIP/100-00000055", "0?Set(__CWIGNORE=)") in new stack
    -- Executing [8011@ext-local:3] Gosub("PJSIP/100-00000055", "macro-exten-vm,s,1(novm,8011,0,0,0)") in new stack
    -- Executing [s@macro-exten-vm:1] Gosub("PJSIP/100-00000055", "macro-user-callerid,s,1()") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/100-00000055", "TOUCH_MONITOR=1726911340.122") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/100-00000055", "CHANCONTEXT=") in new stack
    -- Executing [s@macro-user-callerid:3] Progress("PJSIP/100-00000055", "") in new stack
<--- Transmitting SIP response (847 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>;tag=462e3300-b4ab-4853-b54c-ddf4ec25c1c1
CSeq: 5234 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Contact: <sip:10.0.2.16:5060>
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3935900140 3935900142 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15570 RTP/AVP 0 8 102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Executing [s@macro-user-callerid:4] Set("PJSIP/100-00000055", "CHANCONTEXT=") in new stack
    ... stripped for brevity ...
    -- Executing [s@func-apply-sipheaders:16] Return("PJSIP/8011-00000056", "") in new stack
  == Spawn extension (from-internal, 8011, 1) exited non-zero on 'PJSIP/8011-00000056'
    -- PJSIP/8011-00000056 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
<--- Transmitting SIP request (999 bytes) to UDP:10.0.2.98:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.16:5060;rport;branch=z9hG4bKPjd6064420-0558-42cc-8bbf-2ec5740eb91a
From: "Sebastien CEF (laptop)" <sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
CSeq: 14302 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Sebastien CEF (laptop)" <sip:[email protected]>
Max-Forwards: 70
User-Agent: FPBX-17.0.19.11(21.4.3)
Content-Type: application/sdp
Content-Length:   253

v=0
o=- 408857039 408857039 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15234 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Called PJSIP/8011/sip:[email protected]:5060
<--- Transmitting SIP response (928 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>;tag=462e3300-b4ab-4853-b54c-ddf4ec25c1c1
CSeq: 5234 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
P-Asserted-Identity: "First floor ring screen (Available)" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3935900140 3935900142 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15570 RTP/AVP 0 8 102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (376 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 100 Trying
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
Content-Length: 0
CSeq: 14302 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
To: <sip:[email protected]>
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPjd6064420-0558-42cc-8bbf-2ec5740eb91a


<--- Received SIP response (463 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 101 Dialog Establishment
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 14302 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
To: <sip:[email protected]>;tag=29d8da1348363cc861d421378158b64f
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPjd6064420-0558-42cc-8bbf-2ec5740eb91a


<--- Received SIP response (601 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 180 Ringing
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
Contact: <sip:[email protected]:5060>
Content-Length: 0
CSeq: 14302 INVITE
DependentInfo: 10.0.2.99
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
LeaveType: FTP
MaxConnectingTime: 300
MaxLeaveWordTime: 30
MaxRingingTime: 45
ShortNumber: 8011
To: <sip:[email protected]>;tag=29d8da1348363cc861d421378158b64f
TransMode: SupportRTSP
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPjd6064420-0558-42cc-8bbf-2ec5740eb91a


    -- PJSIP/8011-00000056 is ringing
<--- Transmitting SIP response (916 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>;tag=462e3300-b4ab-4853-b54c-ddf4ec25c1c1
CSeq: 5234 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
P-Asserted-Identity: "First floor ring screen" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3935900140 3935900142 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15570 RTP/AVP 0 8 102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (787 bytes) from UDP:10.0.2.98:5060 --->
SIP/2.0 200 OK
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
Contact: <sip:[email protected]:5060>
Content-Length: 309
Content-Type: application/sdp
CSeq: 14302 INVITE
From: "Sebastien CEF (laptop)"<sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
To: <sip:[email protected]>;tag=29d8da1348363cc861d421378158b64f
User-Agent: Dahua UAC/3.0 DHI-VTH5321G-W V4.400.0.6
Via: SIP/2.0/UDP 10.0.2.16:5060;rport=5060;branch=z9hG4bKPjd6064420-0558-42cc-8bbf-2ec5740eb91a

v=0
o=- 1726911344 3 IN IP4 10.0.2.98
s=Dahua VT 1.5
c=IN IP4 10.0.2.98
t=0 0
m=audio 20000 RTP/AVP 101 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 PCM/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 20001 RTP/AVP 96
a=framerate:25.000000
a=rtpmap:96 H264/90000
a=recvonly

<--- Transmitting SIP request (426 bytes) to UDP:10.0.2.98:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.16:5060;rport;branch=z9hG4bKPj6e894f41-7037-444a-b363-f4be47d6297c
From: "Sebastien CEF (laptop)" <sip:[email protected]>;tag=39e8999a-ca72-4f33-a2b7-3bb58bda9612
To: <sip:[email protected]>;tag=29d8da1348363cc861d421378158b64f
Call-ID: ab37d838-a511-40d6-904f-d2b65863d41a
CSeq: 14302 ACK
Max-Forwards: 70
User-Agent: FPBX-17.0.19.11(21.4.3)
Content-Length:  0


    -- PJSIP/8011-00000056 answered PJSIP/100-00000055
<--- Transmitting SIP response (950 bytes) to UDP:10.0.0.253:53884 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.253:53884;rport=53884;received=10.0.0.253;branch=z9hG4bKPjy8WFb3ZMIXsv3wO3V-z07qh6uultlqPm
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: <sip:[email protected]>;tag=462e3300-b4ab-4853-b54c-ddf4ec25c1c1
CSeq: 5234 INVITE
Server: FPBX-17.0.19.11(21.4.3)
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER
Contact: <sip:10.0.2.16:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "First floor ring screen" <sip:[email protected]>
Content-Type: application/sdp
Content-Length:   255

v=0
o=- 3935900140 3935900142 IN IP4 10.0.2.16
s=Asterisk
c=IN IP4 10.0.2.16
t=0 0
m=audio 15570 RTP/AVP 0 8 102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Channel PJSIP/8011-00000056 joined 'simple_bridge' basic-bridge <c27bd2b4-8c23-4d9f-b6c3-1a3a5dff8b9f>
    -- Channel PJSIP/100-00000055 joined 'simple_bridge' basic-bridge <c27bd2b4-8c23-4d9f-b6c3-1a3a5dff8b9f>
<--- Received SIP request (366 bytes) from UDP:10.0.0.253:53884 --->
ACK sip:10.0.2.16:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.253:53884;rport;branch=z9hG4bKPjCnFAbxlaovJn4T.3zJjOUVBWmGZK7qWf
Max-Forwards: 70
From: "Sebastien C" <sip:[email protected]>;tag=6C50m7pHAaBd9JP2SHbESoUMaktU7z1B
To: sip:[email protected];tag=462e3300-b4ab-4853-b54c-ddf4ec25c1c1
Call-ID: .9SHxuX8m.XtQqUuLO.l7gZKX9el4WQZ
CSeq: 5234 ACK
Content-Length:  0


<--- Received SIP request (393 bytes) from UDP:10.0.0.253:53884 --->
BYE sip:10.0.2.16:5060 SIP/2.0
... stripped for brevity ...

dicko · September 21, 2024, 11:33am

Can we assume that the transport is jitsi → jigasi → asterisk → videophone ? Asterisk is a b2bua and jigasi → jitsi a bridge via xmpp , so at least 4 channels involved all advertising different codecs.

I would first reduce complexity while testing by eliminating the jitsi → jigasi component, does vto → asterisk → vth work?

if so reduce the codecs that jigasi has configured to ulaw/alaw/video also and test again

fourat · September 21, 2024, 7:43pm

I’m using Jitsi to connect directly to asterisk, but anyway, just switched to “Telephone” on macos, a bare simple sip client; and still having identical logs (and codec negociation details).

dicko · September 21, 2024, 10:56pm

I don’t really understand, I was not aware that Jitsi spoke SIP, and “Telephone” on macos doesn’t mean anything to me, sorry, maybe someone else ?

fourat · September 22, 2024, 8:37am

Sorry for the confusion, although I’ve made my best to expose the issue in a clear simple manner …

So getting back to the issue description, let me clarify:

Test scenarios:

When I call VTO from VTH I hear scratching sound, It’s like a codec negociation issue.
When I call VTO from a Sip phone (Telephone on macos or portsip on android) (extension 100), sound and video is good !
When I call VTH from the Jitsi, I hear the same scratching sound.

dicko · September 22, 2024, 2:47pm

Given

As meither Asterisk nor Jigasi (Jitsi doesn’t do SIP) are involved I think your resolution must begin with Dahua

fourat · September 22, 2024, 9:29pm

Hello @dicko, I appreciate your support !

I’ve gone tcpdumping the traffic and comparing a successful videocall to a failed videocall, for the context:

From X to Z: Caller is my sip phone, Called is Dahua’s VTO: Video/Audio are okay,
From Y to Z: Caller is Dahua’s VTH, Called is Dahua’s VTO: No video, audio is scratchy,

And there goes the comparison:

1. INVITE:

2. INVITE OK:

3. Streaming audio/video:

Does these shots help identifying the issue ?

dicko · September 22, 2024, 9:33pm

Not to me, I don’t have any Dahua devices, but if yours can’t communicate successfully when neither Asterisk nor jigasi are involved, surely you should be asking Dahua for help as to how to configure them