MPEGTS over UDP: Service is unavailable when timeout.

[ruanshengbin]

Note: Please read FAQ before file an issue, see 2716

Note: Before asking a question, please refer to the FAQ, specifically 2716.

The video stream push on the monitored UDP port disconnects for a period of time (exceeding the timeout), and when attempting to push the video stream again, an error occurs. The management page cannot view the video stream and cannot play the video stream. It is recommended to support the reuse of UDP ports.

Please describe the issue you are facing here (请在此处描述您遇到的问题)

1. SRS Version (版本): SRS-CentOS7-x86_64-5.0.33

1. SRS Log (日志):

serve error code=1011 : service cycle : rtmp: stream service : rtmp: publish timeout 10000ms, nb_msgs=0
13.
thread [23868][1ps6r2p5]: do_cycle() [src/app/srs_app_rtmp_conn.cpp:217][errno=4]
14.
thread [23868][1ps6r2p5]: service_cycle() [src/app/srs_app_rtmp_conn.cpp:414][errno=4]
15.
thread [23868][1ps6r2p5]: do_publishing() 

[2022-07-10 15:07:11.634][Warn][20618][33868fm5][32] parse ts packet err=code=1009 : ts: handle ts message : ts: consume audio : write audio raw frame : send messages : send messages : send iovs : writev : writev
719.
thread [20618][33868fm5]: decode() [src/kernel/srs_kernel_ts.cpp:267][errno=32]
720.
thread [20618][33868fm5]: on_ts_message() [src/app/srs_app_mpegts_udp.cpp:318][errno=32]
721.
thread [20618][33868fm5]: on_ts_audio() [src/app/srs_app_mpegts_udp.cpp:530][errno=32]
722.
thread [20618][33868fm5]: rtmp_write_packet() [src/app/srs_app_mpegts_udp.cpp:584][errno=32]
723.
thread [20618][33868fm5]: send_and_free_messages() [src/protocol/srs_rtmp_stack.cpp:822][errno=32]
724.
thread [20618][33868fm5]: do_send_messages() [src/protocol/srs_rtmp_stack.cpp:528][errno=32]
725.
thread [20618][33868fm5]: srs_write_large_iovs() [src/protocol/srs_protocol_utility.cpp:357][errno=32]
726.
thread [20618][33868fm5]: writev() [src/protocol/srs_service_st.cpp:609][errno=32]
727.
�[0m[2022-07-10 15:07:11.635][Trace][20618][00209r55] RTMP client ip=127.0.0.1:33334, fd=15
728.
[2022-07-10 15:07:11.641][Trace][20618][33868fm5] complex handshake success.
729.
[2022-07-10 15:07:11.641][Trace][20618][33868fm5] protocol in.buffer=0, in.ack=0, out.ack=0, in.chunk=128, out.chunk=128



parse ts packet err=code=4019 : ts: ts packet decode : ts: demux payload : ts: PES fresh packet length=0, us=0, cc=11

parse ts packet err=code=1009 : ts: handle ts message : ts: consume audio : write audio raw frame : send messages : send messages : send iovs : writev : writev

serve error code=5011 : service cycle : rtmp: stream service : rtmp: receive thread : handle publish message : rtmp: consume message : rtmp: consume audio : bridger consume audio : aac append header : adts

ts: drop PES 13B for duplicated cc=0xb8

1. SRS Config (配置):

# main config for srs.
# @see full.conf for detail config.

listen              21935;
max_connections     1000;
srs_log_tank        console;
srs_log_file        ./objs/srs.log;
daemon              off;
http_api {
    enabled         on;
    listen          21985;
}
http_server {
    enabled         on;
    listen          28080;
    dir             ./objs/nginx/html;
}
rtc_server {
    enabled on;
    listen 28000; # UDP port
    # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
    candidate $CANDIDATE;
}

stream_caster {
    enabled         on;
    caster          mpegts_over_udp;
    output          rtmp://127.0.0.1:21935/live/line1;
    listen          8935;
}

stream_caster {
    enabled         on;
    caster          mpegts_over_udp;
    output          rtmp://127.0.0.1:21935/live/line2;
    listen          8936;
}

stream_caster {
    enabled         on;
    caster          mpegts_over_udp;
    output          rtmp://127.0.0.1:21935/live/line3;
    listen          8937;
}

stream_caster {
    enabled         on;
    caster          mpegts_over_udp;
    output          rtmp://127.0.0.1:21935/live/line4;
    listen          8938;
}

stream_caster {
    enabled         on;
    caster          mpegts_over_udp;
    output          rtmp://127.0.0.1:21935/live/line5;
    listen          8939;
}

vhost __defaultVhost__ {
    hls {
        enabled         on;
    }
    http_remux {
        enabled     on;
        mount       [vhost]/[app]/[stream].flv;
    }
    rtc {
        enabled     on;
        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
        rtmp_to_rtc on;
        # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
        rtc_to_rtmp off;
    }
	publish {
        # about MR, read https://github.com/ossrs/srs/issues/241
        # when enabled the mr, SRS will read as large as possible.
        # default: off
        mr          off;
        # the latency in ms for MR(merged-read),
        # the performance+ when latency+, and memory+,
        #       memory(buffer) = latency * kbps / 8
        # for example, latency=500ms, kbps=3000kbps, each publish connection will consume
        #       memory = 500 * 3000 / 8 = 187500B = 183KB
        # when there are 2500 publisher, the total memory of SRS at least:
        #       183KB * 2500 = 446MB
        # the recommended value is [300, 2000]
        # default: 350
        mr_latency  350;

        # the 1st packet timeout in ms for encoder.
        # default: 20000
        firstpkt_timeout    10000;
        # the normal packet timeout in ms for encoder.
        # default: 5000
        normal_timeout      10000;
        # whether parse the sps when publish stream.
        # we can got the resolution of video for stat api.
        # but we may failed to cause publish failed.
        # @remark If disabled, HLS might never update the sps/pps, it depends on this.
        # default: on
        parse_sps   on;
    }
}

Replay (重现)

Please describe how to replay the bug? (重现Bug的步骤)

1. Use UDP to push video stream
2. Stop the push service for more than 10 seconds
3. Push video stream again through the same UDP port

A UDP receiving and forwarding RTMP service that can recover from interruptions even if the timeout is exceeded, otherwise once the timeout is reached, the configured stream_caster forwarding port will become unavailable.

> Please describe your expectation (描述你期望发生的事情)

TRANS_BY_GPT3

[ruanshengbin]

Currently, the problem has been resolved by modifying the code. It is best to keep the stream_caster UDP connection active, otherwise, once it times out and breaks, it cannot be reused again.

TRANS_BY_GPT3