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Sur d’autres sites (7361)

  • There is no data in the inbound-rtp section of WebRTC. I don't know why

    13 juin 2024, par qyt

    I am a streaming media server, and I need to stream video to WebRTC in H.264 format. The SDP exchange has no errors, and Edge passes normally.

    


    These are the log debugging details from edge://webrtc-internals/. Both DTLS and STUN show normal status, and SDP exchange is also normal. I used Wireshark to capture packets and saw that data streaming has already started. The transport section (iceState=connected, dtlsState=connected, id=T01) also shows that data has been received, but there is no display of RTP video data at all.

    


    timestamp   2024/6/13 16:34:01
bytesSent   5592
[bytesSent_in_bits/s]   176.2108579387652
packetsSent 243
[packetsSent/s] 1.001198056470257
bytesReceived   69890594
[bytesReceived_in_bits/s]   0
packetsReceived 49678
[packetsReceived/s] 0
dtlsState   connected
selectedCandidatePairId CPeVYPKUmD_FoU/ff10
localCertificateId  CFE9:17:14:B4:62:C3:4C:FF:90:C0:57:50:ED:30:D3:92:BC:BB:7C:13:11:AB:07:E8:28:3B:F6:A5:C7:66:50:77
remoteCertificateId CF09:0C:ED:3E:B3:AC:33:87:2F:7E:B0:BD:76:EB:B5:66:B0:D8:60:F7:95:99:52:B5:53:DA:AC:E7:75:00:09:07
tlsVersion  FEFD
dtlsCipher  TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
dtlsRole    client
srtpCipher  AES_CM_128_HMAC_SHA1_80
selectedCandidatePairChanges    1
iceRole controlling
iceLocalUsernameFragment    R5DR
iceState    connected


    


    video recv info

    


    inbound-rtp (kind=video, mid=1, ssrc=2124085007, id=IT01V2124085007)
Statistics IT01V2124085007
timestamp   2024/6/13 16:34:49
ssrc    2124085007
kind    video
transportId T01
jitter  0
packetsLost 0
trackIdentifier 1395f18c-6ab9-4dbc-9149-edb59a81044d
mid 1
packetsReceived 0
[packetsReceived/s] 0
bytesReceived   0
[bytesReceived_in_bits/s]   0
headerBytesReceived 0
[headerBytesReceived_in_bits/s] 0
jitterBufferDelay   0
[jitterBufferDelay/jitterBufferEmittedCount_in_ms]  0
jitterBufferTargetDelay 0
[jitterBufferTargetDelay/jitterBufferEmittedCount_in_ms]    0
jitterBufferMinimumDelay    0
[jitterBufferMinimumDelay/jitterBufferEmittedCount_in_ms]   0
jitterBufferEmittedCount    0
framesReceived  0
[framesReceived/s]  0
[framesReceived-framesDecoded-framesDropped]    0
framesDecoded   0
[framesDecoded/s]   0
keyFramesDecoded    0
[keyFramesDecoded/s]    0
framesDropped   0
totalDecodeTime 0
[totalDecodeTime/framesDecoded_in_ms]   0
totalProcessingDelay    0
[totalProcessingDelay/framesDecoded_in_ms]  0
totalAssemblyTime   0
[totalAssemblyTime/framesAssembledFromMultiplePackets_in_ms]    0
framesAssembledFromMultiplePackets  0
totalInterFrameDelay    0
[totalInterFrameDelay/framesDecoded_in_ms]  0
totalSquaredInterFrameDelay 0
[interFrameDelayStDev_in_ms]    0
pauseCount  0
totalPausesDuration 0
freezeCount 0
totalFreezesDuration    0
firCount    0
pliCount    0
nackCount   0
minPlayoutDelay 0


    


    wireshark,I have verified that the SSRC in the SRTP is correct.

    


    enter image description here

    


    This player works normally when tested with other streaming servers. I don't know what the problem is. Is there any way to find out why the web browser cannot play the WebRTC stream that I'm pushing ?

    


  • avcodec/dvbsubdec : support returning exact end times

    22 juin 2014, par Anshul Maheshwari
    avcodec/dvbsubdec : support returning exact end times

    fixess part of ticket #2024
    

    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
    • [DH] libavcodec/dvbsubdec.c

  • lavu/opt : Clarify the scope of AVOptions

    24 avril 2024, par Andrew Sayers
    lavu/opt : Clarify the scope of AVOptions

    See discussion on the mailing list :
    
https://ffmpeg.org/pipermail/ffmpeg-devel/2024-April/326054.html
    

    Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] libavutil/opt.h

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