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ramon .Gaba
2/24/23
How send rtp_forward packets to decklink
errors in
gstreamer
but at the output of the blackmagic I get a black signal. If I capture the same rtp stream to disk in mkv format the recording is perfect. Does anyone know how to play
unread,
How send rtp_forward packets to decklink
errors in
gstreamer
but at the output of the blackmagic I get a black signal. If I capture the same rtp stream to disk in mkv format the recording is perfect. Does anyone know how to play
2/24/23
Jaswanth K
, …
Tristan Matthews
8
2/23/23
JANUS-GATEWAY plugin streaming for AV1 codac
of any
gstreamer
package I have on Fedora 37, for instance. > Indeed it's part of the rust plugin set: https://gitlab.freedesktop.org/
gstreamer
/gst-plugins-rs which I
unread,
JANUS-GATEWAY plugin streaming for AV1 codac
of any
gstreamer
package I have on Fedora 37, for instance. > Indeed it's part of the rust plugin set: https://gitlab.freedesktop.org/
gstreamer
/gst-plugins-rs which I
2/23/23
Divyansh Jain
,
Alessandro Toppi
2
2/7/23
How to bring video from janus server to browser
, your
gstreamer
app must send a RTP stream to a Janus streaming mountpoint. If you want to use janode (and the Janus API directly) then take a look at janode "streaming" sample
unread,
How to bring video from janus server to browser
, your
gstreamer
app must send a RTP stream to a Janus streaming mountpoint. If you want to use janode (and the Janus API directly) then take a look at janode "streaming" sample
2/7/23
Eran Braun
1/31/23
Media Detection
weather the
video
from a client in Videoroom is publishing and the
video
is being consumed, and as such for audio, is there a regulated way to do it, i thought i can consume the stream with
unread,
Media Detection
weather the
video
from a client in Videoroom is publishing and the
video
is being consumed, and as such for audio, is there a regulated way to do it, i thought i can consume the stream with
1/31/23
Avital Yachin
,
Lorenzo Miniero
4
1/24/23
Multi-participants and RTP forwarding
with the
video
stream > at server side (using ffmpeg), and the flexibility is quite limited. > > Did you encounter similar issues? Or, are you using a fixed layout for the >
unread,
Multi-participants and RTP forwarding
with the
video
stream > at server side (using ffmpeg), and the flexibility is quite limited. > > Did you encounter similar issues? Or, are you using a fixed layout for the >
1/24/23
Benjamin Ionescu
, …
Lorenzo Miniero
11
12/13/22
Trouble with RTP forwarding
forward a
video
stream? (in which case you should have used video_pt > instead). But using the new APIs is the correct way to do it anyway. > > L. > > Il giorno venerdì 9 dicembre
unread,
Trouble with RTP forwarding
forward a
video
stream? (in which case you should have used video_pt > instead). But using the new APIs is the correct way to do it anyway. > > L. > > Il giorno venerdì 9 dicembre
12/13/22
adna...@gmail.com
,
Lorenzo Miniero
2
11/8/22
Snapshot of video streams?
, a
GStreamer
pipelime that receives RTP traffic, decodes
video
, and saves something every tot seconds. I showed an example of that in a workshop some time ago: https://www.youtube
unread,
Snapshot of video streams?
, a
GStreamer
pipelime that receives RTP traffic, decodes
video
, and saves something every tot seconds. I showed an example of that in a workshop some time ago: https://www.youtube
11/8/22
Ramon Blanquer
,
Lorenzo Miniero
6
11/8/22
VideoRoom RTP-Forward: Big visual artifacts when low bandwidth publishers change sent resolution
FFmpeg/
GStreamer
as they are, they should > be able to cope with varying resolutions now, but of course that depends on > the version and how they're used. The videomixer
unread,
VideoRoom RTP-Forward: Big visual artifacts when low bandwidth publishers change sent resolution
FFmpeg/
GStreamer
as they are, they should > be able to cope with varying resolutions now, but of course that depends on > the version and how they're used. The videomixer
11/8/22
jeff carman
, …
Lorenzo Miniero
4
11/7/22
AudioBridge Plain RTP
In the
GStreamer
example, though, you're setting 8111 as the SINK >> port, the one you'll send media to, which is clearly wrong. The sink port >> should be the one
unread,
AudioBridge Plain RTP
In the
GStreamer
example, though, you're setting 8111 as the SINK >> port, the one you'll send media to, which is clearly wrong. The sink port >> should be the one
11/7/22
Venelin Spiridonov
,
Lorenzo Miniero
4
10/21/22
Streaming from opencv with gstreamer
for the
GStreamer
guys. Apologies. > On Wednesday, September 28, 2022 at 11:45:25 AM UTC+3 lmin...@gmail.com > wrote: > >> That may be more of a question for a
GStreamer
unread,
Streaming from opencv with gstreamer
for the
GStreamer
guys. Apologies. > On Wednesday, September 28, 2022 at 11:45:25 AM UTC+3 lmin...@gmail.com > wrote: > >> That may be more of a question for a
GStreamer
10/21/22
Mainul Hassan
,
Lorenzo Miniero
3
10/5/22
Video Stream using Gstreamer to Janus Server
for creating
video
feed from
Gstreamer
. >> >> gst-launch-1.0 -v videotestsrc !
video
/x-raw,framerate=20/1 ! videoscale >> ! videoconvert ! x264enc tune=zerolatency
unread,
Video Stream using Gstreamer to Janus Server
for creating
video
feed from
Gstreamer
. >> >> gst-launch-1.0 -v videotestsrc !
video
/x-raw,framerate=20/1 ! videoscale >> ! videoconvert ! x264enc tune=zerolatency
10/5/22
AbdulSamad
,
Lorenzo Miniero
3
9/23/22
How to resolve "Unknown SSRC, dropping packet" warning?
the default
gstreamer
pipeline sending this as part of its attempt to transmit
video
. Once I fixed that part of the pipeline (replaced with just a 'queue,' really), the warning
unread,
How to resolve "Unknown SSRC, dropping packet" warning?
the default
gstreamer
pipeline sending this as part of its attempt to transmit
video
. Once I fixed that part of the pipeline (replaced with just a 'queue,' really), the warning
9/23/22
Hokoha Team
, …
Lorenzo Miniero
5
8/18/22
High latency (10-15s) on Android web browser when viewing RTSP streams
my generated
video
stream had >>> non-perfect
video
frame timestamps. Do you have a component like >>> gstreamer's videorate for your camera's
video
unread,
High latency (10-15s) on Android web browser when viewing RTSP streams
my generated
video
stream had >>> non-perfect
video
frame timestamps. Do you have a component like >>> gstreamer's videorate for your camera's
video
8/18/22
Tobias Hnyk
,
Lorenzo Miniero
8
6/29/22
WHIP H264 no image
know what
gstreamer
plugin is best > to use, if one wants a hardware accelerated H.264 encoder? > > lmin...@gmail.com schrieb am Dienstag, 28. Juni 2022 um 10:10:09 UTC+2
unread,
WHIP H264 no image
know what
gstreamer
plugin is best > to use, if one wants a hardware accelerated H.264 encoder? > > lmin...@gmail.com schrieb am Dienstag, 28. Juni 2022 um 10:10:09 UTC+2
6/29/22
Sebastian Sprafke
,
Lorenzo Miniero
2
6/6/22
Video with Audio and Gstreamer script
Hello, > > does anyone has a
Video
with Audio and related to that a working
GStreamer
> Script for thepurpose of the Streaming Plugin ? > > Regards > Sebastian >
unread,
Video with Audio and Gstreamer script
Hello, > > does anyone has a
Video
with Audio and related to that a working
GStreamer
> Script for thepurpose of the Streaming Plugin ? > > Regards > Sebastian >
6/6/22
trunki 3
, …
Lorenzo Miniero
10
5/31/22
Streaming with simultcast problems.
browser and
GStreamer
very likely the PictureID was always 15-bit, which is why this went unnoticed for a while. I've pushed the fix on both the 0.x branch (the one you'll probably
unread,
Streaming with simultcast problems.
browser and
GStreamer
very likely the PictureID was always 15-bit, which is why this went unnoticed for a while. I've pushed the fix on both the 0.x branch (the one you'll probably
5/31/22
trunki 3
,
Lorenzo Miniero
3
5/9/22
Its posible Videocall from and to capture card?
>>
video
from the client and forward to the decklink. >> >> It would be similar to what VmixCall does. A videocall but from and to >> SDI Decklink using ffmpeg
unread,
Its posible Videocall from and to capture card?
>>
video
from the client and forward to the decklink. >> >> It would be similar to what VmixCall does. A videocall but from and to >> SDI Decklink using ffmpeg
5/9/22
Ben Parham
,
Lorenzo Miniero
2
4/21/22
RTSP to WebRTC with AV1 codec
ffmpeg or
gstreamer
. Once that's done, you can send the output via RTP to a mountpoint in the Janus Streaming plugin to turn it into a WebRTC broadcast. L. Il giorno mercoledì 20 aprile
unread,
RTSP to WebRTC with AV1 codec
ffmpeg or
gstreamer
. Once that's done, you can send the output via RTP to a mountpoint in the Janus Streaming plugin to turn it into a WebRTC broadcast. L. Il giorno mercoledì 20 aprile
4/21/22
Angus Margerison
,
Alessandro Toppi
3
4/15/22
Client sync on Audio only streaming on Wifi LAN
>>
gstreamer
-sample: { >> type="rtp" >> id = 1 >> desription = "local audio stream" >> audio = true >>
video
= false >
unread,
Client sync on Audio only streaming on Wifi LAN
>>
gstreamer
-sample: { >> type="rtp" >> id = 1 >> desription = "local audio stream" >> audio = true >>
video
= false >
4/15/22
Rani Yaroshinski
,
Lorenzo Miniero
2
3/17/22
webrtcbin and janus issues
client via
gstreamer
webrtcbin java implementation, as Janus publisher. For some reason janus gets a good SDP, and finds a good candidate, succeeds with it, and then passes to other
unread,
webrtcbin and janus issues
client via
gstreamer
webrtcbin java implementation, as Janus publisher. For some reason janus gets a good SDP, and finds a good candidate, succeeds with it, and then passes to other
3/17/22
lizardpeter
, …
Ravi Srivastava
11
3/1/22
Error: only sent -1 bytes
NACK (
video
stream #0) > [Tue Mar 1 13:06:55 2022] [4579198889511125] Retransmitted 9 packets due > to NACK (
video
stream #0) > [Tue Mar 1 13:07:02 2022] [4579198889511125
unread,
Error: only sent -1 bytes
NACK (
video
stream #0) > [Tue Mar 1 13:06:55 2022] [4579198889511125] Retransmitted 9 packets due > to NACK (
video
stream #0) > [Tue Mar 1 13:07:02 2022] [4579198889511125
3/1/22
barak pahima
, …
Sarsaparilla Sunset
5
2/23/22
Videoroom bitrate
of the
video
stream to fit that bandwidth. The REMB makes more sense in the traditional p2p context. Each peer uses it to tell the other peer how much bandwidth it is capable of handling
unread,
Videoroom bitrate
of the
video
stream to fit that bandwidth. The REMB makes more sense in the traditional p2p context. Each peer uses it to tell the other peer how much bandwidth it is capable of handling
2/23/22
Beka Iglesias López
, …
Kevin Artesh
8
2/10/22
Webcam still on in browser after destroying session
>>
video
/demo websites do they yield the expected light off when closed? >>>>>>> >>>>>>> On Wed, Jul 29, 2020 at 7:09 AM Beka
unread,
Webcam still on in browser after destroying session
>>
video
/demo websites do they yield the expected light off when closed? >>>>>>> >>>>>>> On Wed, Jul 29, 2020 at 7:09 AM Beka
2/10/22
kaiduan xie
,
Alessandro Toppi
5
2/9/22
SRTP error on latest master with latest chrome
particulat having
gstreamer
and janus installed on the same machine (as you discovered) is often a bad idea. Il giorno mercoledì 9 febbraio 2022 alle 03:46:54 UTC+1 kaid...@goodstartsoft
unread,
SRTP error on latest master with latest chrome
particulat having
gstreamer
and janus installed on the same machine (as you discovered) is often a bad idea. Il giorno mercoledì 9 febbraio 2022 alle 03:46:54 UTC+1 kaid...@goodstartsoft
2/9/22
Dini
2
2/5/22
Janus streaming - family DIY
audio &
video
via browser (single FHD > webcam). > - During the time of no connection to webserver (99% of time), system > should be in standby mode, meaning no processing
unread,
Janus streaming - family DIY
audio &
video
via browser (single FHD > webcam). > - During the time of no connection to webserver (99% of time), system > should be in standby mode, meaning no processing
2/5/22
Greer Viau
, …
Alessandro Toppi
7
1/10/22
Cant play stream on specific WiFi network
(eg
gstreamer
/ffmpeg pipe or similar). > > Il giorno venerdì 26 novembre 2021 alle 17:15:24 UTC+1 gvi...@gmail.com > ha scritto: > >> correction, I realize i shouldnt
unread,
Cant play stream on specific WiFi network
(eg
gstreamer
/ffmpeg pipe or similar). > > Il giorno venerdì 26 novembre 2021 alle 17:15:24 UTC+1 gvi...@gmail.com > ha scritto: > >> correction, I realize i shouldnt
1/10/22
David Müller
1/4/22
Streaming Plugin - remote stream is muted for some videos
some small
video
files. For some videos I can't display them in the client despite the server receiving the rtcp packets. After establishing the webrtc peerconnection it tells
unread,
Streaming Plugin - remote stream is muted for some videos
some small
video
files. For some videos I can't display them in the client despite the server receiving the rtcp packets. After establishing the webrtc peerconnection it tells
1/4/22
Jim O'Carroll
, …
Lorenzo Miniero
5
12/30/21
Multistream Help
3 concurrent
video
>> streams in separate iframes. Streams originate from 3 cameras on a locally >> connected raspberry pi using
Gstreamer
. Streaming config file has
unread,
Multistream Help
3 concurrent
video
>> streams in separate iframes. Streams originate from 3 cameras on a locally >> connected raspberry pi using
Gstreamer
. Streaming config file has
12/30/21
Sol
,
Lorenzo Miniero
4
12/30/21
Re-stream an incoming video stream to another server
the following
gstreamer
command: gst-launch-1.0 -v udpsrc port=10001 caps=application/x-rtp,media=
video
,clock-rate=90000,encoding-name=H264,payload=96 ! rtph264depay
unread,
Re-stream an incoming video stream to another server
the following
gstreamer
command: gst-launch-1.0 -v udpsrc port=10001 caps=application/x-rtp,media=
video
,clock-rate=90000,encoding-name=H264,payload=96 ! rtph264depay
12/30/21
Nigam Patel
, …
Meonardo
9
2/18/22
Streaming Plugin :: Only video packet loss
FFmpeg and
GStreamer
have ways to set it, if it's a hardware camera >>>> maybe there's a setting somewhere. >>>> >>>> L. >>
unread,
Streaming Plugin :: Only video packet loss
FFmpeg and
GStreamer
have ways to set it, if it's a hardware camera >>>> maybe there's a setting somewhere. >>>> >>>> L. >>
2/18/22