online2311
commented
2 years ago Chrome Console Log
srs.page.js:92 ?eip=x.x.x.x to overwrite candidate. Overwrite server candidate (public IP) configuration.
srs.page.js:93 ?api=x to overwrite WebRTC API(1985).
srs.page.js:94 ?schema=http|https to overwrite WebRTC API protocol.
rtc_player.html?vhost=rtc.vhost.srs.com&app=gb28181&stream=34020000002000000001.flv&server=rtc.vhost.srs.com&port=8080&autostart=true&schema=http:106
For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements
`(Anonymous) @ rtc_player.html?vhost=rtc.vhost.srs.com&app=gb28181&stream=34020000002000000001.flv&server=rtc.vhost.srs.com&port=8080&autostart=true&schema=http:106`
(Anonymous) @ rtc_player.html?vhost=rtc.vhost.srs.com&app=gb28181&stream=34020000002000000001.flv&server=rtc.vhost.srs.com&port=8080&autostart=true&schema=http:106
srs.sdk.js:316 Generated offer: Object
srs.sdk.js:323 Got answer: Object
srs.sdk.js:316 Generated offer: Object
srs.sdk.js:323 Got answer: Object
srs.sdk.js:316 Generated offer: {api: 'http://rtc.vhost.srs.com:1985/rtc/v1/play/', tid: '13e15c2', streamurl: 'webrtc://rtc.vhost.srs.com/gb28181/34020000002000000001', clientip: null, sdp: 'v=0\r\no=- 8065349402620815713 2 IN IP4 127.0.0.1\r\ns… transport-cc\r\na=fmtp:49 repair-window=10000000\r\n'}api: "http://rtc.vhost.srs.com:1985/rtc/v1/play/"clientip: nullsdp: "v=0\r\no=- 8065349402620815713 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=extmap-allow-mixed\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:XVBp\r\na=ice-pwd:FF6zAJBKwxsHBmFEjFNIwfHO\r\na=ice-options:trickle\r\na=fingerprint:sha-256 8E:D8:8B:C9:E7:76:6B:D5:8E:08:FB:18:16:81:D4:6B:C2:D4:40:F2:03:D9:E5:2C:E5:C5:63:47:0B:96:03:2D\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na
TRANS_BY_GPT3
winlinvip
abigailcg
Description (描述)
SRS Version (版本):
SRS/5.0.89Make sure to maintain the markdown structure.SRS Log (日志): Make sure to maintain the markdown structure.
stream_caster { enabled on; caster gb28181; output rtmp://127.0.0.1/gb28181/[stream]?vhost=rtc.vhost.srs.com; listen 9000; sip { enabled on; listen 5060; candidate 172.16.250.78; } }
http_api { enabled on; listen 1985; raw_api { enabled on; allow_reload on; } }
http_server { enabled on; listen 80; dir ./objs/nginx/html; }
stats { network 0; }
rtc_server { enabled on; listen 8000; tcp { enabled on; listen 8000; } protocol all; candidate 172.16.250.78; use_auto_detect_network_ip on; ip_family ipv4; api_as_candidates on; resolve_api_domain on; keep_api_domain off; ecdsa on; encrypt on; reuseport 1; merge_nalus off; }
vhost rtc.vhost.srs.com { rtc { enabled on; nack on; nack_no_copy on; twcc on; stun_timeout 30; stun_strict_check on; dtls_role passive; dtls_version auto; drop_for_pt 0; rtmp_to_rtc on; keep_bframe off; rtc_to_rtmp off; pli_for_rtmp 6.0; } min_latency on; play { mw_latency 0; mw_msgs 0; } http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}
vhost defaultVhost { rtc {
enabled on;
nack on;
nack_no_copy on;
twcc on; stun_timeout 30; stun_strict_check on; dtls_role passive; dtls_version auto; drop_for_pt 0; rtmp_to_rtc on; keep_bframe off; rtc_to_rtmp off; pli_for_rtmp 6.0; }
min_latency on;
play {
mw_latency 0;
mw_msgs 0;
}
}