WebRTC AV1 streaming failed, no valid AV1 payload type found. #2760

trswygg · 2021-11-29T12:17:57Z

Note: Before asking a question, please read the FAQ (Please read FAQ before filing an issue) #2716

WebRTC AV1 streaming, playback failed, using SRS returned 400, checking SDP found that AV1X has changed to a=rtpmap:35 AV1/90000.
Using a self-built server (Linux on a local network).
chrome:
Google Chrome | 96.0.4664.55 (Official Build) (x86_64)
Operating System | macOS Version 12.0.1 (Build 21A559)
JavaScript | V8 9.6.180.12
User Agent | Mozilla/5.0 (Macintosh; Intel Mac OS X 10_15_7) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/96.0.4664.55 Safari/537.36

Please describe the issue you encountered.

SRS version: SRS/5.0.19(Leo)
SRS log:

[2021-11-29 11:19:10.223][Trace][274637][373v3df6] Hybrid cpu=1.00%,161MB, cid=1,0, timer=62,0,0, clock=0,49,0,0,0,0,0,0,0
[2021-11-29 11:19:13.138][Trace][274637][67f446i5] HTTP #0 172.17.0.3:34462 POST http://ubuntu.pve/rtc/v1/publish/?codec=av1, content-length=6179
[2021-11-29 11:19:13.138][Trace][274637][67f446i5] RTC publish webrtc://ubuntu.pve/live/livestream?codec=av1, api=https://ubuntu.pve:443/rtc/v1/publish/?codec=av1, tid=2ba78c4, clientip=192.1.1.50, app=live, stream=livestream, offer=5678B, eip=, codec=av1
[2021-11-29 11:19:13.138][Trace][274637][67f446i5] ignore attribute=, value=
[2021-11-29 11:19:13.138][Warn][274637][67f446i5][11] RTC error code=5018 : create session : create session : add publisher : publish negotiate : no found valid AV1 payload type
thread [274637][67f446i5]: do_serve_http() [src/app/srs_app_rtc_api.cpp:443][errno=11]
thread [274637][67f446i5]: create_session() [src/app/srs_app_rtc_server.cpp:480][errno=11]
thread [274637][67f446i5]: do_create_session() [src/app/srs_app_rtc_server.cpp:497][errno=11]
thread [274637][67f446i5]: add_publisher() [src/app/srs_app_rtc_conn.cpp:1981][errno=11]
thread [274637][67f446i5]: negotiate_publish_capability() [src/app/srs_app_rtc_conn.cpp:2882][errno=11]
[2021-11-29 11:19:13.138][Trace][274637][67f446i5] TCP: before dispose resource(HttpConn)(0x5592d7e65760), conns=1, zombies=0, ign=0, inz=0, ind=0
[2021-11-29 11:19:13.138][Warn][274637][67f446i5][104] client disconnect peer. ret=1007
[2021-11-29 11:19:13.138][Trace][274637][01t53653] TCP: clear zombies=1 resources, conns=1, removing=0, unsubs=0
[2021-11-29 11:19:13.138][Trace][274637][67f446i5] TCP: disposing #0 resource(HttpConn)(0x5592d7e65760), conns=1, disposing=1, zombies=0
[2021-11-29 11:19:15.224][Trace][274637][373v3df6] Hybrid cpu=0.00%,161MB, cid=1,0, timer=62,0,0, clock=0,49,0,0,0,0,0,0,0

SRS configuration:

listen 1935;
max_connections 1000;
daemon on;
srs_log_tank file;
srs_log_level trace;
srs_log_file ./objs/srs.log;

http_api {
    enabled on;
    listen 1985;
    crossdomain on;
    raw_api {
        enabled on;
        allow_reload on;
        allow_query on;
        allow_update off;
    }
    https {
        enabled on;
        listen 1986;
        key ./conf/server.key;
        cert ./conf/server.crt;
    }
}

http_server {
    enabled on;
    listen 8080;
}

# WebRTC
rtc_server {
    enabled on;
    listen 8000;
    perf_stat       on;
    ip_family ipv4;
    candidate $CANDIDATE;
}

vhost __defaultVhost__ {
    rtc {
        enabled on;
        dtls_version auto;

        rtmp_to_rtc on;
        rtc_to_rtmp on;
        # The PLI interval in seconds, for RTC to RTMP.
        # Note the available range is [0.5, 30]
        pli_for_rtmp 6.0;
    }
    min_latency on;
    play {
            mw_latency      0;
            mw_msgs         0;
            queue_length    10;
        }
    http_remux {
        enabled on;
        mount [vhost]/[app]/[stream].flv;
    }
}

Replay

How to replay bug?

Steps to reproduce the bug

1. Visit https://ubuntu.pve/players/rtc_publisher.html?autostart=false

webrtc://ubuntu.pve/live/aabbcc?codec=av1
1. The player is destroyed, and the console prints Got answer: {code: 400}.

推流端sdp (Streaming end SDP)

v=0
o=- 583941141996781899 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS
m=audio 9 UDP/TLS/RTP/SAVPF 111 63 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Vf9F
a=ice-pwd:7RzzXxhY18LmUszopMe8KaUW
a=ice-options:trickle
a=fingerprint:sha-256 F3:9F:40:41:16:79:4B:07:90:D6:EC:40:77:CD:3D:A2:30:95:47:5A:95:A6:3D:70:A1:BF:B0:30:1B:80:96:F2
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:- 1fd1f971-b674-44c0-802c-3fc5e2c6bfd3
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:331486039 cname:nHAPEsqENL+Z6CK2
a=ssrc:331486039 msid:- 1fd1f971-b674-44c0-802c-3fc5e2c6bfd3
a=ssrc:331486039 mslabel:-
a=ssrc:331486039 label:1fd1f971-b674-44c0-802c-3fc5e2c6bfd3
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Vf9F
a=ice-pwd:7RzzXxhY18LmUszopMe8KaUW
a=ice-options:trickle
a=fingerprint:sha-256 F3:9F:40:41:16:79:4B:07:90:D6:EC:40:77:CD:3D:A2:30:95:47:5A:95:A6:3D:70:A1:BF:B0:30:1B:80:96:F2
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:- 9a54aa56-2115-4a30-8cf7-db9aa4fef522
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:35 AV1/90000
a=rtcp-fb:35 goog-remb
a=rtcp-fb:35 transport-cc
a=rtcp-fb:35 ccm fir
a=rtcp-fb:35 nack
a=rtcp-fb:35 nack pli
a=rtpmap:36 rtx/90000
a=fmtp:36 apt=35
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032
a=rtpmap:118 rtx/90000
a=fmtp:118 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 582915416 3168979914
a=ssrc:582915416 cname:nHAPEsqENL+Z6CK2
a=ssrc:582915416 msid:- 9a54aa56-2115-4a30-8cf7-db9aa4fef522
a=ssrc:582915416 mslabel:-
a=ssrc:582915416 label:9a54aa56-2115-4a30-8cf7-db9aa4fef522
a=ssrc:3168979914 cname:nHAPEsqENL+Z6CK2
a=ssrc:3168979914 msid:- 9a54aa56-2115-4a30-8cf7-db9aa4fef522
a=ssrc:3168979914 mslabel:-
a=ssrc:3168979914 label:9a54aa56-2115-4a30-8cf7-db9aa4fef522

I don't know why

TRANS_BY_GPT3

The text was updated successfully, but these errors were encountered:

winlinvip · 2021-11-30T00:55:30Z

You should have failed to push the stream, it didn't go up successfully, it's not a playback failure.

no found valid AV1 payload type

TRANS_BY_GPT3

trswygg · 2021-11-30T02:32:47Z

Before and after exchanging SDP on the client side, you can solve it by using replaceAll("AV1","AV1X") and replaceAll("AV1X","AV1"), but I don't know if I should do this, and currently I don't have any other computers and browsers available for testing.

setLocalDescription of SrsRtcPublisherAsync.publish can accept AV1X, but the player will report "Failed to execute 'setLocalDescription' on 'RTCPeerConnection': Failed to set local offer sdp: Failed to set local video description recv parameters for m-section with mid='1'."

It feels very strange that this problem only appeared after upgrading Chrome.

TRANS_BY_GPT3

jocover · 2021-12-03T01:46:32Z

https://bugs.chromium.org/p/webrtc/issues/detail?id=13166
AV1X has been renamed to AV1.

TRANS_BY_GPT3

winlinvip · 2021-12-04T03:13:34Z

@jocover 👍 Can you conveniently submit a PR?

TRANS_BY_GPT3

Huachao · 2021-12-10T17:13:23Z

@winlinvip @jocover I submitted a fix PR #2784, please help take a look. Make sure to maintain the markdown structure.

TRANS_BY_GPT3

* RTC: Replace payload name AV1X with AV1 for WebRTC. (#2760) * Add compatibility check code for old versions of Chrome * 新增获取track_desc的AV1X编码兼容性检查

winlinvip · 2023-01-02T13:01:38Z

Fixed in SRS 4

winlinvip changed the title ~~WebRTC AV1 推流，播放失败~~ WebRTC AV1 推流失败 no found valid AV1 payload type Nov 30, 2021

winlinvip self-assigned this Nov 30, 2021

winlinvip added the WebRTC WebRTC, RTC2RTMP or RTMP2RTC. label Nov 30, 2021

winlinvip added this to the 5.0 milestone Nov 30, 2021

Huachao added a commit to Huachao/srs that referenced this issue Dec 10, 2021

RTC: Replace payload name AV1X with AV1 for WebRTC. (ossrs#2760)

1cbfd15

winlinvip added a commit that referenced this issue Dec 20, 2021

RTC: Support payload name AV1X/AV1. (#2784)(#2760). v4.0.207

ee24bb4

winlinvip closed this as completed Jan 2, 2023

winlinvip modified the milestones: 5.0, 4.0 Jan 2, 2023

winlinvip changed the title ~~WebRTC AV1 推流失败 no found valid AV1 payload type~~ WebRTC AV1 streaming failed, no valid AV1 payload type found. Jul 28, 2023

winlinvip added the TransByAI Translated by AI/GPT. label Jul 28, 2023

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WebRTC AV1 streaming failed, no valid AV1 payload type found. #2760

WebRTC AV1 streaming failed, no valid AV1 payload type found. #2760

trswygg commented Nov 29, 2021 •

edited by winlinvip

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winlinvip commented Nov 30, 2021 •

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Huachao commented Dec 10, 2021 •

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winlinvip commented Jan 2, 2023

WebRTC AV1 streaming failed, no valid AV1 payload type found. #2760

WebRTC AV1 streaming failed, no valid AV1 payload type found. #2760

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