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When using rtc2rtmp, an error occurs when streaming with getDisplayMedia and playing flv. #2403

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bywwcnll opened this issue Jun 8, 2021 · 4 comments · Fixed by #2461
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When using rtc2rtmp, an error occurs when streaming with getDisplayMedia and playing flv. #2403

bywwcnll opened this issue Jun 8, 2021 · 4 comments · Fixed by #2461
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@duiniuluantanqin
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TransByAI Translated by AI/GPT. WebRTC WebRTC, RTC2RTMP or RTMP2RTC.
Milestone
4.0

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@bywwcnll
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bywwcnll commented Jun 8, 2021 •
edited by winlinvip
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Description'

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Please describe the issue you encountered here.
'
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  1. SRS version: 4.0release latest version compiled after git clone
  2. SRS log is as follows:
[2021-06-08 17:55:18.472][Trace][9615][spx42895] NACK: update seq=20481, nack range [20480, 20481]
[2021-06-08 17:55:18.506][Trace][9615][spx42895] NACK: update seq=20506, nack range [20505, 20506]
[2021-06-08 17:55:19.439][Trace][9615][16y1uv31] Hybrid cpu=0.00%,0MB, cid=19,10, timer=59,4,20, clock=0,25,20,1,0,0,0,0,0, free=1, objs=(pkt:103,raw:7,fua:95,msg:104,oth:1,buf:51)
[2021-06-08 17:55:19.439][Trace][9615][16y1uv31] RTC: Server conns=1, rpkts=(89,rtp:86,stun:1,rtcp:1), spkts=(12,rtp:0,stun:1,rtcp:22), rtcp=(pli:1,twcc:9,rr:1), snk=(47,a:0,v:47,h:1), fid=(id:0,fid:0,ffid:89,addr:1,faddr:89)
[2021-06-08 17:55:19.518][Trace][9615][23q83814] <- RTC RECV #10, udp 888, pps 52/88, schedule 888
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] HTTP #0 127.0.0.1:52354 GET http://localhost:8080/live/show.flv, content-length=-1
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] dispatch cached gop success. count=0, duration=9
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] create consumer, active=1, queue_size=0.00, jitter=30000000
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] set fd=11, SO_SNDBUF=146988=>87500, buffer=350ms
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] FLV /live/show.flv, encoder=FLV, nodelay=0, mw_sleep=350ms, cache=0, msgs=128
[2021-06-08 17:55:19.754][Trace][9615][i128tn42] FLV: write header audio=0, video=1
[2021-06-08 17:55:20.031][Trace][9615][spx42895] 36B video sh,  codec(7, profile=Baseline, level=Other, 1728x1088, 0kbps, 0.0fps, 0.0s)
[2021-06-08 17:55:20.031][Trace][9615][spx42895] set ts=505698570, header=20615, lost=20616
[2021-06-08 17:55:24.439][Trace][9615][16y1uv31] Hybrid cpu=0.00%,0MB, cid=19,10, timer=59,4,20, clock=0,25,20,1,0,0,0,0,0, free=1, objs=(pkt:103,raw:7,fua:95,msg:104,oth:1,buf:51)
[2021-06-08 17:55:24.439][Trace][9615][16y1uv31] RTC: Server conns=1, rpkts=(89,rtp:86,stun:1,rtcp:1), spkts=(12,rtp:0,stun:1,rtcp:22), rtcp=(pli:1,twcc:9,rr:1), snk=(47,a:0,v:47,h:1), fid=(id:0,fid:0,ffid:89,addr:1,faddr:89)
[2021-06-08 17:55:24.523][Trace][9615][spx42895] NACK: update seq=20848, nack range [20844, 20848]
[2021-06-08 17:55:24.551][Trace][9615][spx42895] NACK: update seq=20871, nack range [20870, 20871]
[2021-06-08 17:55:26.050][Trace][9615][16y1uv31] RTC: Need PLI ssrc=3968468662, play=[16y1uv31], publish=[spx42895], count=3/3
[2021-06-08 17:55:26.050][Trace][9615][spx42895] RTC: Request PLI ssrc=3968468662, play=[16y1uv31], count=3/3, bytes=12B
[2021-06-08 17:55:26.072][Trace][9615][spx42895] 36B video sh,  codec(7, profile=Baseline, level=Other, 1728x1088, 0kbps, 0.0fps, 0.0s)
[2021-06-08 17:55:26.072][Trace][9615][spx42895] set ts=506242350, header=20916, lost=20917
[2021-06-08 17:55:28.543][Trace][9615][spx42895] NACK: update seq=21120, nack range [21117, 21120]
^C[2021-06-08 17:55:29.268][Trace][9615][68364m49] sig=2, user terminate program, fast quit
[2021-06-08 17:55:29.440][Trace][9615][16y1uv31] Hybrid cpu=0.00%,0MB, cid=2,1, timer=59,10,46, clock=0,20,25,1,0,0,0,0,0, objs=(pkt:110,raw:62,fua:47,msg:157,oth:1,buf:55)
[2021-06-08 17:55:29.440][Trace][9615][16y1uv31] RTC: Server conns=1, rpkts=(89,rtp:86,stun:1,rtcp:1), spkts=(12,rtp:0,stun:1,rtcp:22), rtcp=(pli:1,twcc:9,rr:1), snk=(47,a:0,v:47,h:1), fid=(id:0,fid:0,ffid:89,addr:1,faddr:89)
[2021-06-08 17:55:29.475][Trace][9615][68364m49] cleanup for quit signal fast=1, grace=0
  1. SRS configuration is as follows:
listen              1935;
max_connections     1000;
daemon              off;
srs_log_tank        console;

http_server {
    enabled         on;
    listen          8080;
    dir             ./objs/nginx/html;
}

http_api {
    enabled         on;
    listen          1985;
}
stats {
    network         0;
}
rtc_server {
    enabled on;
    # Listen at udp://8000
    listen 8000;
    #
    # The $CANDIDATE means fetch from env, if not configed, use * as default.
    #
    # The * means retrieving server IP automatically, from all network interfaces,
    # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate
    candidate $CANDIDATE;
}

vhost __defaultVhost__ {
    rtc {
        enabled     on;
        rtc_to_rtmp on;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
    }
}

It is the default rtc2rtmp.conf.

Replay

How to replay bug?

Steps to reproduce the bug

Steps to replay the bug:

  1. Start SRS: ./objs/srs -c conf/rtc2rtmp.conf
  2. Modify line 50 in /path/to/srs/trunk/objs/nginx/html/players/js/srs.sdk.js from navigator.mediaDevices.getUserMedia to navigator.mediaDevices.getDisplayMedia
  3. Access http://localhost:8080/players/rtc_publisher.html?autostart=true to perform RTC screen sharing streaming
  4. Access http://localhost:8080/players/srs_player.html?autostart=true to view the FLV stream video. After watching for a while, an error occurs.
    image

Expected behavior (Expect)

> Hope that getDisplayMedia can work properly like getUserMedia.

TRANS_BY_GPT3
@bywwcnll
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bywwcnll commented Jun 9, 2021 •
edited by winlinvip
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After testing, it was found that disabling the twcc function of rtc resolved the issue. I hope the experts can pay attention to it.
image

TRANS_BY_GPT3

@winlinvip winlinvip added the WebRTC WebRTC, RTC2RTMP or RTMP2RTC. label Jun 16, 2021
@duiniuluantanqin
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duiniuluantanqin commented Jul 5, 2021 •
edited by winlinvip
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Verified, Chrome does indeed have this issue. But Firefox runs well.

TRANS_BY_GPT3

@duiniuluantanqin duiniuluantanqin linked a pull request Jul 6, 2021 that will close this issue
Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy #2461
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@duiniuluantanqin duiniuluantanqin self-assigned this Jul 6, 2021
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#2461 (comment)

winlinvip pushed a commit that referenced this issue Jul 8, 2021
@duiniuluantanqin @winlinvip
For #2403, fix padding packets for RTMP2RTC. 4.0.140.
90b7933 
Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy (#2461)

* Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy

* make clear for magic number

make clear for magic number

* Update srs_app_rtc_source.cpp
@bywwcnll
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bywwcnll commented Jul 9, 2021 •
edited by winlinvip
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Verified

TRANS_BY_GPT3

@bywwcnll bywwcnll closed this as completed Jul 9, 2021
@winlinvip winlinvip added this to the 4.0 milestone Sep 4, 2021
@winlinvip winlinvip changed the title 使用rtc2rtmp时，用getDisplayMedia推流，播放flv出错 When using rtc2rtmp, an error occurs when streaming with getDisplayMedia and playing flv. Jul 28, 2023
@winlinvip winlinvip added the TransByAI Translated by AI/GPT. label Jul 28, 2023
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Labels
TransByAI Translated by AI/GPT. WebRTC WebRTC, RTC2RTMP or RTMP2RTC.
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Milestone
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@winlinvip @duiniuluantanqin @bywwcnll