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Describe the bug
I attempted to implement RTMP to WebRTC conversion following the instructions provided on the OSSRS (Open Source Streaming Realtime Server) website. However, I encountered an error when verifying the remote session description protocol (SDP). The error message indicated a server error with code 5018, which corresponds to an SDP negotiation failure. The specific issue mentioned that the remote SDP check failed because it currently only supports the BUNDLE group policy. I am using the Chrome browser to play the WebRTC stream, and the player and the SRS server are not hosted on the same machine but are located within the same local network.
Version
SRS 5.0Release
To Reproduce
Steps to reproduce the behavior:
1../objs/srs -c conf/rtmp2rtc.conf
2. ffmpeg -re -i ./doc/source.flv -c copy -f flv rtmp://localhost/live/livestream
3. Chrome open "http://x.x.x.x:1985/rtc/v1/whep/?app=live&stream=livestream" (x.x.x.x is my SRS server ip)
4. [2024-01-31 15:58:32.245][INFO][30155][5uo5683s] HTTP #0 10.1.40.89:14549 GET http://10.1.40.164:1985/rtc/v1/whep/?app=live&stream=livestream, content-length=-1 [2024-01-31 15:58:32.247][INFO][30155][5uo5683s] RTC whip play /live/livestream, clientip=10.1.40.89, app=live, stream=livestream, offer=0B, eip=, codec=, srtp=, dtls=, ufrag=, pwd=, param=app=live&stream=livestream [2024-01-31 15:58:32.249][INFO][30155][5uo5683s] TCP: before dispose resource(HttpConn)(0x6070000477c0), conns=2, zombies=0, ign=0, inz=0, ind=0 [2024-01-31 15:58:32.249][ERROR][30155][5uo5683s][11] serve error code=5018(RtcSdpNegotiate)(RTC do SDP negotiate failed) : process request=0 : cors serve : serve http : serve : serve : remote sdp check failed : now only support BUNDLE, group policy= thread [30155][5uo5683s]: process_requests() [./src/app/srs_app_http_conn.cpp:209][errno=11] thread [30155][5uo5683s]: process_request() [./src/app/srs_app_http_conn.cpp:236][errno=11] thread [30155][5uo5683s]: serve_http() [./src/protocol/srs_protocol_http_stack.cpp:768][errno=11] thread [30155][5uo5683s]: serve_http() [./src/app/srs_app_rtc_api.cpp:648][errno=11] thread [30155][5uo5683s]: do_serve_http() [./src/app/srs_app_rtc_api.cpp:752][errno=11] thread [30155][5uo5683s]: serve_http() [./src/app/srs_app_rtc_api.cpp:198][errno=11] thread [30155][5uo5683s]: check_remote_sdp() [./src/app/srs_app_rtc_api.cpp:273][errno=11](Resource temporarily unavailable)
Expected behavior
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Screenshots
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Additional context
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TRANS_BY_GPT4
The text was updated successfully, but these errors were encountered:
First of all, you should provide the entire log for the session, not just one or a few lines, or a partial part of the logs. This way, you can filter the log for the session by grepping the session ID 5uo5683s.
Second, this issue is normally caused by an empty HTTP body, meaning the HTTP request is empty, so SRS cannot parse the boundary ID from it. Furthermore, the version you are using does not display the issue correctly in the logs. Therefore, you can upgrade to the latest version, and the logs should then clearly address this issue.
Describe the bug
I attempted to implement RTMP to WebRTC conversion following the instructions provided on the OSSRS (Open Source Streaming Realtime Server) website. However, I encountered an error when verifying the remote session description protocol (SDP). The error message indicated a server error with code 5018, which corresponds to an SDP negotiation failure. The specific issue mentioned that the remote SDP check failed because it currently only supports the BUNDLE group policy. I am using the Chrome browser to play the WebRTC stream, and the player and the SRS server are not hosted on the same machine but are located within the same local network.
Version
SRS 5.0Release
To Reproduce
Steps to reproduce the behavior:
1.
./objs/srs -c conf/rtmp2rtc.conf
2.
ffmpeg -re -i ./doc/source.flv -c copy -f flv rtmp://localhost/live/livestream
3. Chrome open "http://x.x.x.x:1985/rtc/v1/whep/?app=live&stream=livestream" (x.x.x.x is my SRS server ip)
4.
[2024-01-31 15:58:32.245][INFO][30155][5uo5683s] HTTP #0 10.1.40.89:14549 GET http://10.1.40.164:1985/rtc/v1/whep/?app=live&stream=livestream, content-length=-1 [2024-01-31 15:58:32.247][INFO][30155][5uo5683s] RTC whip play /live/livestream, clientip=10.1.40.89, app=live, stream=livestream, offer=0B, eip=, codec=, srtp=, dtls=, ufrag=, pwd=, param=app=live&stream=livestream [2024-01-31 15:58:32.249][INFO][30155][5uo5683s] TCP: before dispose resource(HttpConn)(0x6070000477c0), conns=2, zombies=0, ign=0, inz=0, ind=0 [2024-01-31 15:58:32.249][ERROR][30155][5uo5683s][11] serve error code=5018(RtcSdpNegotiate)(RTC do SDP negotiate failed) : process request=0 : cors serve : serve http : serve : serve : remote sdp check failed : now only support BUNDLE, group policy= thread [30155][5uo5683s]: process_requests() [./src/app/srs_app_http_conn.cpp:209][errno=11] thread [30155][5uo5683s]: process_request() [./src/app/srs_app_http_conn.cpp:236][errno=11] thread [30155][5uo5683s]: serve_http() [./src/protocol/srs_protocol_http_stack.cpp:768][errno=11] thread [30155][5uo5683s]: serve_http() [./src/app/srs_app_rtc_api.cpp:648][errno=11] thread [30155][5uo5683s]: do_serve_http() [./src/app/srs_app_rtc_api.cpp:752][errno=11] thread [30155][5uo5683s]: serve_http() [./src/app/srs_app_rtc_api.cpp:198][errno=11] thread [30155][5uo5683s]: check_remote_sdp() [./src/app/srs_app_rtc_api.cpp:273][errno=11](Resource temporarily unavailable)
Expected behavior
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Screenshots
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Additional context
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TRANS_BY_GPT4
The text was updated successfully, but these errors were encountered: