discuss-webrtc

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ks75vl, … Björn Terelius6

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 Bashir6

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 Alvestrand3

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

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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 Bashir2

Sep 23

webrtc for unity have question

Please share the console log file here. Add logging around signaling and ICE candidate exchange.

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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 Bashir2

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 Bashir2

Sep 23

Regarding webrtc offer request.

@Garry, I hope this will help you resolve the issue: https://groups.google.com/g/discuss-webrtc/c/

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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 Bashir2

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 Bashir5

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 Bashir3

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 Bashir4

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 nazmus3

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 Zhu4

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 Andreasson6

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

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

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

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

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

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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 Alvestrand5

Sep 6

Have you tried flipping through the codecs available to see if some codecs are more efficient than

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Have you tried flipping through the codecs available to see if some codecs are more efficient than

Sep 6

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 Herre4

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

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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

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

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

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, guest2713142

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 codonnier3

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

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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

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 (

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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

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