Dahua VTO support

[luzik]

Browser is asking for microphone permission, but no audio from VTO and unmute button is disabled

api:
  base_path: "/go2rtc"
streams:
  vto:
    - rtsp://admin:pass@192.168.124.30/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif

[
  {
    "media:0": "video, sendonly, 96 H264/90000",
    "media:1": "audio, sendonly, 97 L16/16000",
    "media:2": "application, sendonly, 107 VND.ONVIF.METADATA/90000",
    "media:3": "audio, recvonly, 97 L16/16000",
    "receive": 824840,
    "remote_addr": "192.168.124.30:554",
    "send": 0,
    "track:0": "96 H264/90000, sinks=1",
    "type": "RTSP client producer",
    "url": "rtsp://192.168.124.30:554/cam/realmonitor?channel=1\u0026subtype=0\u0026unicast=true\u0026proto=Onvif/"
  },
  {
    "remote_addr": "udp4 host 192.168.123.33:63426",
    "send": 826301,
    "type": "WebRTC server consumer",
    "user_agent": "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_7) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/102.0.4976.0 Safari/537.36"
  }
]

v=0
o=- 8819968399131000794 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS mUTzunk2m8gOBSbMpIqZr9ZW7amqcT2Uu8h0
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 35 36 127 122 125 107 108 109 124 121 120 119 114 37
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:cSCj
a=ice-pwd:azQ5DYFApCx3rzPQWvEqknTA
a=ice-options:trickle
a=fingerprint:sha-256 2A:0A:DC:2B:B9:B5:CC:A3:07:E0:98:E7:B3:65:7A:2B:98:5E:AA:AE:0E:76:DB:71:3F:52:3D:E8:55:2B:F8:F7
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 urn:3gpp:video-orientation
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=recvonly
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 VP9/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 profile-id=1
a=rtpmap:123 rtx/90000
a=fmtp:123 apt=102
a=rtpmap:35 AV1/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=124
a=rtpmap:120 red/90000
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=120
a=rtpmap:114 ulpfec/90000
a=rtpmap:37 flexfec-03/90000
a=rtcp-fb:37 goog-remb
a=rtcp-fb:37 transport-cc
a=fmtp:37 repair-window=10000000
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:cSCj
a=ice-pwd:azQ5DYFApCx3rzPQWvEqknTA
a=ice-options:trickle
a=fingerprint:sha-256 2A:0A:DC:2B:B9:B5:CC:A3:07:E0:98:E7:B3:65:7A:2B:98:5E:AA:AE:0E:76:DB:71:3F:52:3D:E8:55:2B:F8:F7
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:mUTzunk2m8gOBSbMpIqZr9ZW7amqcT2Uu8h0 9654a211-5d7a-4500-8034-3b17b1c6c7a7
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1443778823 cname:5sgtTCGio9Mf4sP8
a=ssrc:1443778823 msid:mUTzunk2m8gOBSbMpIqZr9ZW7amqcT2Uu8h0 9654a211-5d7a-4500-8034-3b17b1c6c7a7
a=ssrc:1443778823 mslabel:mUTzunk2m8gOBSbMpIqZr9ZW7amqcT2Uu8h0
a=ssrc:1443778823 label:9654a211-5d7a-4500-8034-3b17b1c6c7a7

6webrtc.html:1 Uncaught (in promise) DOMException: Failed to execute 'addIceCandidate' on 'RTCPeerConnection': The remote description was null

[calisro]

The camera is outputting a format that is incompatible with webrtc. Probably AAC. So transcode it to something webrtc friendly or go into your camera settings and change it to a compatible format. If its like my amcrest, you can change it to G.711 so you don't need to transcode.

Alternatively, this would work:

  vto:
    - rtsp://admin:pass@192.168.124.30/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
    - ffmpeg:rtsp://admin:pass@192.168.124.30/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif#audio=opus

https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#supported_audio_codecs

[AlexxIT]

Both send and receive tracks has L16 codec.

L16 is not AAC, but also doesn't supported by WebRTC. It's possible to receive audio with transcoding via FFmpeg.

But send audio (from microphone) with transcoding is not supported yet.

[luzik]

So Dahua VTO2211G do not allow to change audio codec via browser UI. Maybe there is a way using ONVIF, GET param or dahua api ?

Documentation show that device support
G.711Mu; G.711u; PCM audio compression
https://www.dahuasecurity.com/my/products/All-Products/Video-Intercoms/IP-Series/Villa-Door-Station/Pro-Series/VTO2211G-WP

vlc shows it as PCM S16 BE (s16b)
and ffprobe
Stream #0:1: Audio: pcm_s16be, 16000 Hz, 1 channels, s16, 256 kb/s

VTO do accept audio via API in ulaw. I believe that VTO supporting g.711 while using SIP protocol

[AlexxIT]

Are you sure you can't change codec via camera Web UI?

[luzik]

[luzik]

Hmm, interesting ..there is a way to change audio codec using Onvif

blakeblackshear/frigate#2572

[calisro]

There's likely a way to change it. I can't change it with Amcrest tools but I could change mine with this:

https://dahuawiki.com/ConfigTool

[luzik]

So yeah, it's working. I did used this Happytime onvif client and I hope this settings will stay for ever. If not I will reopen this issue, and ask for help with automating this change.

[
{
"media:0": "video, sendonly, 96 H264/90000",
"media:1": "audio, sendonly, 0 PCMU/8000",
"media:2": "application, sendonly, 107 VND.ONVIF.METADATA/90000",
"media:3": "audio, recvonly, 0 PCMU/8000",
"receive": 8279516,
"remote_addr": "192.168.124.30:554",
"send": 993960,
"track:0": "0 PCMU/8000, sinks=1",
"track:1": "0 PCMU/8000, sinks=1",
"type": "RTSP client producer",
"url": "rtsp://192.168.124.30:554/cam/realmonitor?channel=1\u0026subtype=0\u0026unicast=true\u0026proto=Onvif/"
},
{
"receive": 883680,
"remote_addr": "udp4 host 192.168.123.33:61346",
"send": 8292693,
"type": "WebRTC server consumer",
"user_agent": "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_7) AppleWebKit/605.1.15 (KHTML, like Gecko) Version/15.5 Safari/605.1.15"
}
]

[felipecrs]

@luzik is 2-way audio working for you?

[luzik]

Yeah, latency i low and we are ready to build video doorbell intercom, the only problem is to switch audio codec in programming way. I believe that this could be in scope of this project :)

[bonuzzz]

It is possible to change parameters without onvif client.
To get possible audio/video parameters:
http://admin:pass@192.168.1.110/cgi-bin/configManager.cgi?action=getConfig&name=Encode
For example to change audio bitrate to 8khz:
http://admin:pass@192.168.1.110/cgi-bin/configManager.cgi?action=setConfig&Encode[0].MainFormat[0].Audio.Depth=8

I have VTO2211G. It should work on other models.
ps: microphone in chrome doesn't work without https connection