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discuss-webrtc
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ks75vl
, …
Björn Terelius
6
Sep 24
Invalid format .lib
The designated initializers issue seem unrelated to the rest of this thread. I expect that will be
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Invalid format .lib
The designated initializers issue seem unrelated to the rest of this thread. I expect that will be
Sep 24
Điền Đô
, …
Muhammad Usman Bashir
6
Sep 24
src\out\Default\obj\api\media_stream_interface.lib : fatal error LNK1107: invalid or corrupt file: cannot read at 0x2F4
Hi Muhammad Usman Bashir This is my Linkedin: linkedin.com/in/điền-đô-aa5224259 Can you connect with
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src\out\Default\obj\api\media_stream_interface.lib : fatal error LNK1107: invalid or corrupt file: cannot read at 0x2F4
Hi Muhammad Usman Bashir This is my Linkedin: linkedin.com/in/điền-đô-aa5224259 Can you connect with
Sep 24
Garry Jassal
, …
Harald Alvestrand
3
Sep 23
Regarding offer request.
When asking this kind of question, it is always a good idea to say what server you are trying to get
unread,
Regarding offer request.
When asking this kind of question, it is always a good idea to say what server you are trying to get
Sep 23
Taomeng tao
,
Muhammad Usman Bashir
2
Sep 23
webrtc for unity have question
Please share the console log file here. Add logging around signaling and ICE candidate exchange.
unread,
webrtc for unity have question
Please share the console log file here. Add logging around signaling and ICE candidate exchange.
Sep 23
Shiplu Kundu
,
Muhammad Usman Bashir
2
Sep 23
peerConnection localDescription is null even after setting in IOS webrtc
@Shiplu, the completionHandler is asynchronous, so accessing localDescription immediately after
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peerConnection localDescription is null even after setting in IOS webrtc
@Shiplu, the completionHandler is asynchronous, so accessing localDescription immediately after
Sep 23
Garry Jassal
,
Muhammad Usman Bashir
2
Sep 23
Regarding webrtc offer request.
@Garry, I hope this will help you resolve the issue: https://groups.google.com/g/discuss-webrtc/c/
unread,
Regarding webrtc offer request.
@Garry, I hope this will help you resolve the issue: https://groups.google.com/g/discuss-webrtc/c/
Sep 23
Eleven
,
Muhammad Usman Bashir
2
Sep 23
Does the audio stream in WebRTC support RTX yet?
> The SDP includes configuration for RTX on the audio media section, but browsers like Chrome
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Does the audio stream in WebRTC support RTX yet?
> The SDP includes configuration for RTX on the audio media section, but browsers like Chrome
Sep 23
Sebastian Kunz
, …
Muhammad Usman Bashir
5
Sep 23
Video Streaming latency is oddly high
@Robert, logs show codec negotiation is happening properly and frames are being received. Some other
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Video Streaming latency is oddly high
@Robert, logs show codec negotiation is happening properly and frames are being received. Some other
Sep 23
Philipp Hancke
,
Muhammad Usman Bashir
3
Sep 22
PSA: libWebRTC won't send RTCP BYE when disabling a layer (M119)
Am Fr., 22. Sept. 2023 um 14:56 Uhr schrieb Muhammad Usman Bashir <mail2c...@gmail.com>: @
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PSA: libWebRTC won't send RTCP BYE when disabling a layer (M119)
Am Fr., 22. Sept. 2023 um 14:56 Uhr schrieb Muhammad Usman Bashir <mail2c...@gmail.com>: @
Sep 22
gui...@webrtc.org
Sep 22
PSA: WebRtcAllowLegacyTLSProtocols enterprise policy to be removed from Chrome M121
The WebRtcAllowLegacyTLSProtocols enterprise policy in Chrome, which allows downgrading WebRTC peer
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PSA: WebRtcAllowLegacyTLSProtocols enterprise policy to be removed from Chrome M121
The WebRtcAllowLegacyTLSProtocols enterprise policy in Chrome, which allows downgrading WebRTC peer
Sep 22
Erik
, …
Muhammad Usman Bashir
4
Sep 22
Apple Vision Pro visionos target_os gn support for custom WebRTC build
Hello Erik, I've had the opportunity to work with various smart glasses platforms, including
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Apple Vision Pro visionos target_os gn support for custom WebRTC build
Hello Erik, I've had the opportunity to work with various smart glasses platforms, including
Sep 22
陈鹏越
,
shakeeb nazmus
3
Sep 21
Sending cached GOP data during the start of broadcasting cause continuous stuttering
Hi shakeeb nazmus, I store the original h264 and opus data up to the previous I-frame in the cache,
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Sending cached GOP data during the start of broadcasting cause continuous stuttering
Hi shakeeb nazmus, I store the original h264 and opus data up to the previous I-frame in the cache,
Sep 21
Robert Ayrapetyan
,
Haichao Zhu
4
Sep 21
Unexpected browser frame rendering lag (~100ms)
Hello. Replaced: static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF; with: static
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Unexpected browser frame rendering lag (~100ms)
Hello. Replaced: static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF; with: static
Sep 21
CrIcKeT
,
Henrik Andreasson
6
Sep 20
Failed to create audio device on Android
Sounds like great work. Would it be possible to upload the change for review so that others can use
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Failed to create audio device on Android
Sounds like great work. Would it be possible to upload the change for review so that others can use
Sep 20
Haichao Zhu
Sep 20
Hardware Encoding using NVDIA & AMD Graphics Card?
Do we have plan to support hardware-accelerated encoding using graphics cards, especially for screen
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Hardware Encoding using NVDIA & AMD Graphics Card?
Do we have plan to support hardware-accelerated encoding using graphics cards, especially for screen
Sep 20
Michal Śledź
Sep 20
RTC.ON - the first conference about Membrane & multimedia!
Hi everyone! Some time ago, I announced that Software Mansion and Membrane Framework Team are
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RTC.ON - the first conference about Membrane & multimedia!
Hi everyone! Some time ago, I announced that Software Mansion and Membrane Framework Team are
Sep 20
Henrik Boström
Sep 20
PSA: Callback-based getStats() is removed in M117 (the Deprecation Trial is now mandatory if you need more time)
As previously announced, the callback-based getStats() API is being deprecated and removed. A
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PSA: Callback-based getStats() is removed in M117 (the Deprecation Trial is now mandatory if you need more time)
As previously announced, the callback-based getStats() API is being deprecated and removed. A
Sep 20
Raiden LuiKang
Sep 18
Supporting more 50 webrtc::PeerConnectionInterface s
Hi! I have a question,: How to create more 50-100 webrtc::PeerConnectionInterface object in mobile
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Supporting more 50 webrtc::PeerConnectionInterface s
Hi! I have a question,: How to create more 50-100 webrtc::PeerConnectionInterface object in mobile
Sep 18
mrajkovicpta
Sep 10
webRTC iOS library crashes on setLocalDescription/setRemoteDescription if no completion handler is present
Calling setLocalDescription or setRemoteDescription in swift on an RTCPeerconnection without using
unread,
webRTC iOS library crashes on setLocalDescription/setRemoteDescription if no completion handler is present
Calling setLocalDescription or setRemoteDescription in swift on an RTCPeerconnection without using
Sep 10
Bir Garip Developer
Sep 7
Flutter media soup client error
When I enter the page with the camera and microphone turned off, when I open the camera, the camera
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Flutter media soup client error
When I enter the page with the camera and microphone turned off, when I open the camera, the camera
Sep 7
Sots Ts
, …
Harald Alvestrand
5
Sep 6
Webrtc & GPU
Have you tried flipping through the codecs available to see if some codecs are more efficient than
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Webrtc & GPU
Have you tried flipping through the codecs available to see if some codecs are more efficient than
Sep 6
Shiplu Kundu
Sep 6
IOS webRTC Video/audio/Data not working
Hi team, I need an urgent help. We have upgraded the IOS libjingle version to 114.(Previouly we were
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IOS webRTC Video/audio/Data not working
Hi team, I need an urgent help. We have upgraded the IOS libjingle version to 114.(Previouly we were
Sep 6
Zach Servies
, …
Tony Herre
4
Sep 6
getUserMedia() taking a long time to return stream
We have some stats reported in Chrome for getUserMedia latency which don't show any dramatic
unread,
getUserMedia() taking a long time to return stream
We have some stats reported in Chrome for getUserMedia latency which don't show any dramatic
Sep 6
Higuain Bushun
Sep 6
Runtime inaccessible method errors with the latest Android WebRTC
Hi, I am trying to update from org.webrtc:google-webrtc:1.0.32006. I have tried io.github.webrtc-sdk:
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Runtime inaccessible method errors with the latest Android WebRTC
Hi, I am trying to update from org.webrtc:google-webrtc:1.0.32006. I have tried io.github.webrtc-sdk:
Sep 6
Messiah Cheng
Sep 5
How to transcode webm block which doesn't has EBML header(from MediaRecorder)to m3u8 ts(transport stream))
helping! Now i'm using MediaRecorder to record my camera and screen, I want to upload webm block
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How to transcode webm block which doesn't has EBML header(from MediaRecorder)to m3u8 ts(transport stream))
helping! Now i'm using MediaRecorder to record my camera and screen, I want to upload webm block
Sep 5
Eli
Sep 4
How to fix - Several attemps before connecting?
Hi, I have this connection in: PHP FirebaseRealTime And a COTURN server -- Some times you call the
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How to fix - Several attemps before connecting?
Hi, I have this connection in: PHP FirebaseRealTime And a COTURN server -- Some times you call the
Sep 4
David P
,
guest271314
2
Sep 4
Samsung browser sends corrupted audio when using both WebAudio and WebRTC
> This is probably related to our use of a WebAudio graph to monitor the loudness before the graph
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Samsung browser sends corrupted audio when using both WebAudio and WebRTC
> This is probably related to our use of a WebAudio graph to monitor the loudness before the graph
Sep 4
paul wangota
, …
raj codonnier
3
Sep 4
How to implement Group Video call with webrtc
Hello yes tell me about your problem with mediasoup. On Wed, Aug 23, 2023 at 12:27 PM 'Harald
unread,
How to implement Group Video call with webrtc
Hello yes tell me about your problem with mediasoup. On Wed, Aug 23, 2023 at 12:27 PM 'Harald
Sep 4
Điền Đô
Sep 4
File BUILD.gn in webrtc
Hi guys. I'm working on a personal project using webrtc and Qt. When I dig through the .gn files,
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File BUILD.gn in webrtc
Hi guys. I'm working on a personal project using webrtc and Qt. When I dig through the .gn files,
Sep 4
Norbert Riedlin
Sep 4
SFU: MAX_DROPOUT not honored
Hi, we are implementing an audio SFU using the webrtc stack. When it comes to sudden drops in audio (
unread,
SFU: MAX_DROPOUT not honored
Hi, we are implementing an audio SFU using the webrtc stack. When it comes to sudden drops in audio (
Sep 4