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WebRTC is unable to play video streams in the H.265 format. #4259

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cxkjnszstm opened this issue Dec 18, 2024 · 2 comments
Open

WebRTC is unable to play video streams in the H.265 format. #4259

cxkjnszstm opened this issue Dec 18, 2024 · 2 comments
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TransByAI Translated by AI/GPT.

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@cxkjnszstm
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cxkjnszstm commented Dec 18, 2024 •
edited by winlinvip
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Describe the bug
WebRTC is unable to play video streams in the H265 format.

Version
The version of SRS is 6.0.48.

To Reproduce
Steps to reproduce the behavior:

  1. ./configure --gb28181=on --h265=on
  2. make
  3. /objs/srs -c conf/push.gb28181.conf

Expected behavior

The Safire browser is capable of playing H.265 format videos.

Screenshots
image
image

Additional context
Viewing the printed data in the console, the generated offer in the Safire browser's SDP includes H.265 format within the 'm=video' section. However, after receiving the answer, the selected format is still H.264.

Screenshots
image
image
image

Additional context
Add any other context about the problem here.

TRANS_BY_GPT4
@winlinvip winlinvip changed the title WebRtc无法播放视频流格式为H265的视频 WebRTC is unable to play video streams in the H.265 format. Dec 18, 2024
@winlinvip winlinvip added the TransByAI Translated by AI/GPT. label Dec 18, 2024
@cxkjnszstm
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cxkjnszstm commented Dec 18, 2024 •
edited by winlinvip
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The interface method corresponding to 'got answer' appears to be '/rtc/v1/play'. I have not previously worked with C++ or Go languages. In the directory '\trunk\3rdparty\srs-bench\srs', does the file 'player.go' correspond to the '/rtc/v1/play' method? If so, regarding the 'import' section...
"github.com/pion/webrtc/v3/pkg/media/h264writer"
"github.com/pion/webrtc/v3/pkg/media/ivfwriter"
"github.com/pion/webrtc/v3/pkg/media/oggwriter"
Is it missing H.265 support?

TRANS_BY_GPT4

@suzp1984 suzp1984 mentioned this issue Dec 23, 2024
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@suzp1984
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@cxkjnszstm #4263 is a similar issue, play your whep or rtc-play url with codec parameter.
Here is SRS supported url encoded parameters:

srs/trunk/src/app/srs_app_rtc_api.cpp

Lines 142 to 150 in 7416134

// For client to specifies the candidate(EIP) of server.
string eip = r->query_get("eip");
if (eip.empty()) {
eip = r->query_get("candidate");
}
string codec = r->query_get("codec");
// For client to specifies whether encrypt by SRTP.
string srtp = r->query_get("encrypt");
string dtls = r->query_get("dtls");

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@suzp1984 @winlinvip @cxkjnszstm