Recent questions tagged webrtc

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webrtc - ffmpeg fails to output rtp when setting start time (-ss)

I'm using ffmpeg to stream output to an RTP endpoint. This works fine with audio and video, but fails when using ... /65850430/ffmpeg-fails-to-output-rtp-when-setting-start-time-ss...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.2k views

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webrtc - How can I set up a reverse proxy for the Janus REST api and socket api in Nginx?

I have a Janus Gateway which exposes a REST api on port 8088. The web socket transport is also enabled on my janus ... reverse-proxy-for-the-janus-rest-api-and-socket-api-in-nginx...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.1k views

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webrtc - currently in the P2P field, only using STUN (not use Turn Server) hole punching success rate is what?

I would appreciate you did a great job here, if you could help me answer the following questions: In the case of ... the-p2p-field-only-using-stun-not-use-turn-server-hole-punching...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.1k views

1 answer

webrtc Switching selected connection due to candidate pair state changed

dtls handshake completed over 73.170.65.x:55876 already and then...after few seconds, (p2p_transport_channel. ... -switching-selected-connection-due-to-candidate-pair-state-changed...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.0k views

1 answer

webrtc - How to send stream of objects in Typescript?

I am trying to set up the GRPC + WebRtc code in my local from this GitHub project https://github.com/ ... .com/questions/65902635/how-to-send-stream-of-objects-in-typescript...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.0k views

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webrtc - Publish video to antmedia server from custom camera

I am not able to compile the WebRTCTest application in the antmedia github repository. This is the application ... /65913070/publish-video-to-antmedia-server-from-custom-camera...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.4k views

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webrtc - Use Single PeerConnection for two different plugins of janus (VideoRoom & AudioBridge) in one Android Activity - Android

Prerequisite: I have added VideoRoom functionality in one activity and it is working fine Then i added ... -peerconnection-for-two-different-plugins-of-janus-videoroom-audiob...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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969 views

1 answer

webrtc - When is an endpoint bundle-aware and when not?

From Link: www.w3.org/TR/webrtc/#dom-rtcbundlepolicy Content: 4.2.5 RTCBundlePolicy Enum "If the remote endpoint ... /65926913/when-is-an-endpoint-bundle-aware-and-when-not...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.1k views

1 answer

webrtc - Facing issue with the WebSocket handshake in reSIProcate library

I was trying to use reSIProcate as a SIP client on an iOS device. I m able to compile it for ... questions/65931833/facing-issue-with-the-websocket-handshake-in-resiprocate-library...

asked Oct 7, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.1k views

1 answer

webrtc - Stand alone voice broadcast with RPI4 and WiFi

I want to use RPI4 as a server in a wifi network without internet access. The RPI will have a USB microphone attached to ... What would you use to make this work? Thank you, Roman....

asked Feb 19, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.1k views

1 answer

webrtc 发起方可以收到对方视频流，接收方收不到发起方的视频流

<%@ page contentType="text/html;charset=UTF-8" language="java" %> <!doctype html> <html lang="zh-CN"> <head> <meta charset="UTF-8"> < ... }); </script> </html> 接收方的收到信息发起方收到的信息...

asked Feb 5, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.3k views

1 answer

webrtc 发起方可以收到对方视频流，接收方收不到发起方的视频流

<%@ page contentType="text/html;charset=UTF-8" language="java" %> <!doctype html> <html lang="zh-CN"> <head> <meta charset="UTF-8"> < ... }); </script> </html> 接收方的收到信息发起方收到的信息...

asked Feb 5, 2021 in Technique[技术] by 深蓝 (71.8m points)

0 votes

1.3k views

1 answer

webrtc gclient sync

操作系统是乌班图，下载的是webrtc，使用的是声网的镜像 $ gclient sync line 32: bootstrap-3.8.0.chromium.8_bin/python3/bin/ ... workspace -ensure-file /tmp/tmp_VMpuD.ensure' returned non-zero exit status 4...

asked Jan 27, 2021 in Technique[技术] by 深蓝 (71.8m points)

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1.4k views

1 answer

webrtc gclient sync

操作系统是乌班图，下载的是webrtc，使用的是声网的镜像 $ gclient sync line 32: bootstrap-3.8.0.chromium.8_bin/python3/bin/ ... workspace -ensure-file /tmp/tmp_VMpuD.ensure' returned non-zero exit status 4...

asked Jan 27, 2021 in Technique[技术] by 深蓝 (71.8m points)

0 votes

3.4k views

1 answer

webrtc - currently in the open source code field, only using STUN (not use Turn Server) punching success rate is what?

I would appreciate if you could help me answer the following questions: In the case of large scale and various NAT type ... rate of the hole punching reach to 99.99% above? Thanks...

asked Jan 24, 2021 in Technique[技术] by 深蓝 (71.8m points)

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11.2k views

1 answer

webrtc Switching selected connection due to candidate pair state changed

dtls handshake completed over 73.170.65.x:55876 already and then...after few seconds, (p2p_transport_channel.cc:286): Switching ... , its webrtc m76 stack, we are using for ios,...

asked Jan 23, 2021 in Technique[技术] by 深蓝 (71.8m points)

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