Groups
Conversations
All groups and messages
Send feedback to Google
Help
Training
Sign in
Groups
discuss-webrtc
Conversations
About
discuss-webrtc
Contact owners and managers
1–30 of 11201
Welcome to
WebRTC
!
Note: If it's your first time posting your message will be moderated and will not appear straight away, please be patient, it's not necessary to post/send emails several times.
This list falls under the WebRTC Code of Conduct -
https://webrtc.org/support/c
ode-of-conduct
.
Mark all as read
Report group
0 selected
Christoph Müller
Oct 11
WebRTC under Apple Vision Pro/visionOS
Hello, We currently have a native app for iOS that supports iPhone and iPad. Soon we want to support
unread,
WebRTC under Apple Vision Pro/visionOS
Hello, We currently have a native app for iOS that supports iPhone and iPad. Soon we want to support
Oct 11
Alessio Bazzica
Oct 9
PSA: the iLBC audio codec will be removed
The iLBC audio codec will be removed from WebRTC. The codec is not used in Chrome (`RTCRtpSender.
unread,
PSA: the iLBC audio codec will be removed
The iLBC audio codec will be removed from WebRTC. The codec is not used in Chrome (`RTCRtpSender.
Oct 9
Byoungchan Lee
,
Muhammad Usman Bashir
2
Oct 7
WebRTC with B-frames
The draft, identified as draft-deping-avtcore-video-bframe-01, is an expired Internet-Draft (
unread,
WebRTC with B-frames
The draft, identified as draft-deping-avtcore-video-bframe-01, is an expired Internet-Draft (
Oct 7
guest271314
,
Muhammad Usman Bashir
5
Oct 6
Status of MediaStreamTrackGenerator for audio
> regarding the capability of generating a MediaStreamTrack of kind "audio." The
unread,
Status of MediaStreamTrackGenerator for audio
> regarding the capability of generating a MediaStreamTrack of kind "audio." The
Oct 6
Mojtaba Esfandiari
,
Muhammad Usman Bashir
6
Oct 6
Use sample code in different PCs with different IP Address
First, you have to establish P2P communication between the servers (EC2 debian machines) by
unread,
Use sample code in different PCs with different IP Address
First, you have to establish P2P communication between the servers (EC2 debian machines) by
Oct 6
Kelly Kinyama
Oct 4
Implementing webrtc in pure dart
I have start implementing webrtc protocols in pure dart. I have started with Datagram Transport layer
unread,
Implementing webrtc in pure dart
I have start implementing webrtc protocols in pure dart. I have started with Datagram Transport layer
Oct 4
SangMin (Hassan) Han
Sep 23
iOS Audio output sound Accelerate(fastforwarded) issue when Speaker <-> BLE
is there anyone having trouble with audio sound being accelerated(fastforwared) for few seconds to
unread,
iOS Audio output sound Accelerate(fastforwarded) issue when Speaker <-> BLE
is there anyone having trouble with audio sound being accelerated(fastforwared) for few seconds to
Sep 23
Devashish Sharma
Sep 22
Issue Facing iceCandidateState is showing disconnected but on reverse connection it is connected
I'm getting audio transmitting issue on call when sender and reciever connecting to each other on
unread,
Issue Facing iceCandidateState is showing disconnected but on reverse connection it is connected
I'm getting audio transmitting issue on call when sender and reciever connecting to each other on
Sep 22
Jeremy Mao
, …
shunbo li
15
Sep 19
H264 Video Freeze on Chrome M128 when using OBS for streaming
Thank you very much. Your reply and the related posts have been very helpful and have pointed the
unread,
H264 Video Freeze on Chrome M128 when using OBS for streaming
Thank you very much. Your reply and the related posts have been very helpful and have pointed the
Sep 19
asplin...@gmail.com
Sep 17
FFmpeg can not play dumped ivf file(generated by FrameDumpingDecoder) with codec H265
Hi,buddies, I'm using FrameDumpingDecoder in WebRTC to dump encoded video stream with codec of
unread,
FFmpeg can not play dumped ivf file(generated by FrameDumpingDecoder) with codec H265
Hi,buddies, I'm using FrameDumpingDecoder in WebRTC to dump encoded video stream with codec of
Sep 17
Mohammed Thanweer
,
Philipp Hancke
2
Sep 17
How can I integrate libwebrtc with chromium build and what are the header files which should be included
Chromium builds libwebrtc already, are you trying to link against a different version? Modifying the
unread,
How can I integrate libwebrtc with chromium build and what are the header files which should be included
Chromium builds libwebrtc already, are you trying to link against a different version? Modifying the
Sep 17
Matt Knowles
, …
Philipp Hancke
3
Sep 16
How to Desync Audio / Video
Am Mo., 16. Sept. 2024 um 09:29 Uhr schrieb Matt Knowles <matt.k...@level-ex.com>:
unread,
How to Desync Audio / Video
Am Mo., 16. Sept. 2024 um 09:29 Uhr schrieb Matt Knowles <matt.k...@level-ex.com>:
Sep 16
Christoph Müller
,
Philipp Hancke
2
Sep 16
How can I set or read the RTP timestamp in the iOS objc api?
The random start of the RTP timestamp is intentional. If you want the clients local clock (which may
unread,
How can I set or read the RTP timestamp in the iOS objc api?
The random start of the RTP timestamp is intentional. If you want the clients local clock (which may
Sep 16
Ridhima Dande
, …
Muhammad Usman Bashir
3
Sep 16
WebRTC integration in a voice bot for real-time communication for overcoming Voice Isolation Challenges for Our AI Voice Bot
To improve voice bot performance, I would recommend using RNNoise for audio processing before you
unread,
WebRTC integration in a voice bot for real-time communication for overcoming Voice Isolation Challenges for Our AI Voice Bot
To improve voice bot performance, I would recommend using RNNoise for audio processing before you
Sep 16
aarthi saravanan
, …
Muhammad Usman Bashir
6
Sep 16
WebRTC WiFi-to-WiFi Video Call Not Connecting Despite Using TURN Server
You have to provide ICE Connection and PeerConnection States along with detailed webrtc logs from
unread,
WebRTC WiFi-to-WiFi Video Call Not Connecting Despite Using TURN Server
You have to provide ICE Connection and PeerConnection States along with detailed webrtc logs from
Sep 16
Patrick O'Neil
,
Muhammad Usman Bashir
2
Sep 16
Ice candidates received before offer
You can follow this repo for better understanding of the signaling process: WebRTC Signaling and
unread,
Ice candidates received before offer
You can follow this repo for better understanding of the signaling process: WebRTC Signaling and
Sep 16
Yurii Kyrylchuk
, …
Muhammad Usman Bashir
6
Sep 16
Upcoming Android 15 requirement - 16KB page size
You can follow this repo for better understanding how things are working: WebRTC Android Example On
unread,
Upcoming Android 15 requirement - 16KB page size
You can follow this repo for better understanding how things are working: WebRTC Android Example On
Sep 16
Vijay Kumar
,
Muhammad Usman Bashir
2
Sep 16
sometimes client is not sending RTP after DTLS handshake
Please share details WebRTC logging and share ICE connection states ie webrtc-internals dump reports.
unread,
sometimes client is not sending RTP after DTLS handshake
Please share details WebRTC logging and share ICE connection states ie webrtc-internals dump reports.
Sep 16
Nitin
Sep 13
Answering the call from firefox not going into ice connected state
Hi Everyone, I am facing the issue, when I aswers the audio call over firefox browser the ice
unread,
Answering the call from firefox not going into ice connected state
Hi Everyone, I am facing the issue, when I aswers the audio call over firefox browser the ice
Sep 13
Roman Yuan
Sep 13
[PC] SDP-ApplyLocalDescription SetSsrc(0) crash
We encountered a crash when applying LocalDescription, and found that the call stack was like this:
unread,
[PC] SDP-ApplyLocalDescription SetSsrc(0) crash
We encountered a crash when applying LocalDescription, and found that the call stack was like this:
Sep 13
Beta Wolf
Sep 13
Callee is unable to see the video of the caller but the caller can see callee's video
I'm trying to build a video call app using simple-peer. When there are 2 users in a room, the
unread,
Callee is unable to see the video of the caller but the caller can see callee's video
I'm trying to build a video call app using simple-peer. When there are 2 users in a room, the
Sep 13
Hithesh Jayawardana
Sep 13
change src of a webrtc pipeline dynamically
I'm using the webRTC bin to stream some video (mp4) files. I'm using a pipeline like this.
unread,
change src of a webrtc pipeline dynamically
I'm using the webRTC bin to stream some video (mp4) files. I'm using a pipeline like this.
Sep 13
Leonid Krashanoff
Sep 13
Recommendations in adding video streams via C++ native API on Mac
Greetings friends, family, and enthusiasts of WebRTC. Forgive the long-winded post. tl;dr: Why would
unread,
Recommendations in adding video streams via C++ native API on Mac
Greetings friends, family, and enthusiasts of WebRTC. Forgive the long-winded post. tl;dr: Why would
Sep 13
Jason
Sep 13
webrtc datachannel id
it seems webrtc's DataChannel' id not work properly for example, DataChannel.Init init = new
unread,
webrtc datachannel id
it seems webrtc's DataChannel' id not work properly for example, DataChannel.Init init = new
Sep 13
tomwa...@gmail.com
,
Muhammad Usman Bashir
3
Sep 3
Why hasn't a new version update been released for a long time?
Oh, My bad, I meant the release notes just like as below link https://groups.google.com/u/1/g/discuss
unread,
Why hasn't a new version update been released for a long time?
Oh, My bad, I meant the release notes just like as below link https://groups.google.com/u/1/g/discuss
Sep 3
Afonso Vilalonga
,
Philipp Hancke
3
Aug 26
Maximum RTP packet size
Thank you for your answer! Is RTX negotiated by default? A sexta-feira, 23 de agosto de 2024 à(s) 15:
unread,
Maximum RTP packet size
Thank you for your answer! Is RTX negotiated by default? A sexta-feira, 23 de agosto de 2024 à(s) 15:
Aug 26
Elise Monaghan
, …
Long Enertee
4
Aug 21
Custom ADM for Mac OS X
Hi Elise Monaghan, I'm encountering a similar issue. Could you please share your code or describe
unread,
Custom ADM for Mac OS X
Hi Elise Monaghan, I'm encountering a similar issue. Could you please share your code or describe
Aug 21
ht l
Aug 21
Failed to fetch webrtc_android due to something wrong with third_party/test_fonts
Hi everyone, I ran the command fetch --nohooks webrtc_android to fetch the source code, but I got
unread,
Failed to fetch webrtc_android due to something wrong with third_party/test_fonts
Hi everyone, I ran the command fetch --nohooks webrtc_android to fetch the source code, but I got
Aug 21
Long Enertee
,
Henrik Andreasson
2
Aug 20
How to Change Audio Input and Output Devices in WebRTC on macOS with SwiftUI?
Not sure if it is exactly what you need but here is an example of an externally injected ADM https://
unread,
How to Change Audio Input and Output Devices in WebRTC on macOS with SwiftUI?
Not sure if it is exactly what you need but here is an example of an externally injected ADM https://
Aug 20
Matt Knowles
, …
lidedongsn
5
Aug 19
Jitter and Packet Loss
Have you tested this parameter - jitterBufferTarget? https://chromestatus.com/feature/
unread,
Jitter and Packet Loss
Have you tested this parameter - jitterBufferTarget? https://chromestatus.com/feature/
Aug 19