webrtc live streaming, rtmp playback, video buffering is noticeable

[jamken]

Description'

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Currently, the rtc2rtmp function is being used, but during actual testing, there is a consistent issue of stuttering in rtmp playback. Analyzing the timestamps of the rtmp stream's audio and video frames, it is observed that SRS (Streaming Relay Server) periodically loses a portion of video frames, resulting in a period of time where only audio frames are present without any video frames. After 1 to 2 seconds, the video frames reappear.

Additional information:

1. If rtc playback is used, the video is continuous.
2. The audio frames are consistently continuous without any stuttering.

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1. SRS version: xxxxxx

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4.0.161

1. The log of SRS is as follows:

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The log is in the attachment.

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1. The configuration of SRS is as follows (Config):

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listen 1935;
pid srs.pid;
srs_log_tank file;
srs_log_level trace;
srs_log_file ./logs/srs.log;
max_connections 1000;
da
[srs.txt](https://github.com/ossrs/srs/files/7491698/srs.txt)
emon off;
asprocess on;
[srs.txt](https://github.com/ossrs/srs/files/7491591/srs.txt)

http_server {
    enabled         on;
    listen          0.0.0.0:1936;
    dir             ./html;
}

http_api {
    enabled         on;
    listen          0.0.0.0:1985;
    crossdomain     on;
}

rtc_server {
    enabled on;
    listen 8000;
    candidate *;
}

vhost __defaultVhost__ {
    rtc {
        enabled on;
        nack on;
        twcc on;
        rtc_to_rtmp on;
        pli_for_rtmp 2.0;
    }

    hls {
        enabled on;
        hls_fragment 1.5;
        hls_window 20;
        hls_td_ratio 1.5;
        hls_path /opensight/nginx/html/hls_srs/hls_srs;
        hls_m3u8_file   [app]/[stream].m3u8;
        hls_ts_file     [app]/[stream]-[seq].ts;
        hls_wait_keyframe on;
        hls_dispose     20;
    }

    http_hooks {
        enabled on;
        on_publish http://127.0.0.1:1938/api/srsproxy/v1/srscbk;
        on_unpublish http://127.0.0.1:1938/api/srsproxy/v1/srscbk;
        on_play http://127.0.0.1:1938/api/srsproxy/v1/srscbk;
    }

    http_remux {
        enabled     on;
        mount       [vhost]/flv_srs/[app]/[stream].flv;
    }

    play{
        gop_cache       on;
        queue_length    10;
        mw_latency      20;
        reduce_sequence_header  on;
        time_jitter     off;
        mix_correct     off;
    }

    min_latency     on;
    tcp_nodelay     on;
}

Replay

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How to replay bug?

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Steps to reproduce the bug:

1. Start WebRTC streaming with Chrome, using 1080P video.
2. Use RTMP to pull the stream (e.g., using VLC).
3. Observe continuous audio but video stuttering and jumping.

Expected behavior:

The video and audio should both stream smoothly without any stuttering or jumping.

RTMP pulling stream, both audio and video should be relatively smooth.

TRANS_BY_GPT3

[jamken]

srs.txt

[winlinvip]

Fixed by #2721