5dc292ce64
## Introduce This PR adds support for viewing streams via the RTSP protocol. Note that it only supports viewing streams, not publishing streams via RTSP. Currently, only publishing via RTMP is supported, which is then converted to RTSP. Further work is needed to support publishing RTC/SRT streams and converting them to RTSP. ## Usage Build and run SRS with RTSP support: ``` cd srs/trunk && ./configure --rtsp=on && make -j16 ./objs/srs -c conf/rtsp.conf ``` Push stream via RTMP by FFmpeg: ``` ffmpeg -re -i doc/source.flv -c copy -f flv rtmp://localhost/live/livestream ``` View the stream via RTSP protocol, try UDP first, then use TCP: ``` ffplay -i rtsp://localhost:8554/live/livestream ``` Or specify the transport protocol with TCP: ``` ffplay -rtsp_transport tcp -i rtsp://localhost:8554/live/livestream ``` ## Unit Test Run utest for RTSP: ``` ./configure --utest=on & make utest -j16 ./objs/srs_utest ``` ## Regression Test You need to start SRS for regression testing. ``` ./objs/srs -c conf/regression-test-for-clion.conf ``` Then run regression tests for RTSP. ``` cd srs/trunk/3rdparty/srs-bench go test ./srs -mod=vendor -v -count=1 -run=TestRtmpPublish_RtspPlay ``` ## Blackbox Test For blackbox testing, SRS will be started by utest, so there is no need to start SRS manually. ``` cd srs/trunk/3rdparty/srs-bench go test ./blackbox -mod=vendor -v -count=1 -run=TestFast_RtmpPublish_RtspPlay_Basic ``` ## UDP Transport As UDP requires port allocation, this PR doesn't support delivering media stream via UDP transport, so it will fail if you try to use UDP as transport: ``` ffplay -rtsp_transport udp -i rtsp://localhost:8554/live/livestream [rtsp @ 0x7fbc99a14880] method SETUP failed: 461 Unsupported Transport rtsp://localhost:8554/live/livestream: Protocol not supported [2025-07-05 21:30:52.738][WARN][14916][7d7gf623][35] RTSP: setup failed: code=2057 (RtspTransportNotSupported) : UDP transport not supported, only TCP/interleaved mode is supported ``` There are no plans to support UDP transport for RTSP. In the real world, UDP is rarely used; the vast majority of RTSP traffic uses TCP. ## Play Before Publish RTSP supports audio with AAC and OPUS codecs, which is significantly different from RTMP or WebRTC. RTSP uses commands to exchange SDP and specify the audio track to play, unlike WHEP or HTTP-FLV, which use the query string of the URL. RTSP depends on the player’s behavior, making it very difficult to use and describe. Considering the feature that allows playing the stream before publishing it, it requires generating some default parameters in the SDP. For OPUS, the sample rate is 48 kHz with 2 channels, while AAC is more complex, especially regarding the sample rate, which may be 44.1 kHz, 32 kHz, or 48 kHz. Therefore, for RTSP, we cannot support play-then-publish. Instead, there must already be a stream when playing it, so that the audio codec is determined. ## Opus Codec No Opus codec support for RTSP, because for RTC2RTSP, it always converts RTC to RTMP frames, then converts them to RTSP packets. Therefore, the audio codec is always AAC after converting RTC to RTMP. This means the bridge architecture needs some changes. We need a new bridge that binds to the target protocol. For example, RTC2RTMP converts the audio codec, but RTC2RTSP keeps the original audio codec. Furthermore, the RTC2RTMP bridge should also support bypassing the Opus codec if we use enhanced-RTMP, which supports the Opus audio codec. I think it should be configurable to either transcode or bypass the audio codec. However, this is not relevant to RTSP. ## AI Contributor Below commits are contributed by AI: * [AI: Remove support for media transport via UDP.](755686229f) * [AI: Add crutial logs for each RTSP stage.](9c8cbe7bde) * [AI: Support AAC doec for RTSP.](7d7cc12bae) * [AI: Add option --rtsp for RTSP.](f67414d9ee) * [AI: Extract SrsRtpVideoBuilder for RTC and RTSP.](562e76b904) --------- Co-authored-by: Jacob Su <suzp1984@gmail.com> Co-authored-by: winlin <winlinvip@gmail.com>
53 lines
2.0 KiB
Modula-2
53 lines
2.0 KiB
Modula-2
module github.com/ossrs/srs-bench
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go 1.23.0
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require (
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github.com/bluenviron/gortsplib/v4 v4.13.1
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github.com/ghettovoice/gosip v0.0.0-20220929080231-de8ba881be83
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github.com/google/gopacket v1.1.19
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github.com/haivision/srtgo v0.0.0-20230627061225-a70d53fcd618
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github.com/ossrs/go-oryx-lib v0.0.9
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github.com/pion/ice/v4 v4.0.10
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github.com/pion/interceptor v0.1.40
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github.com/pion/logging v0.2.4
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github.com/pion/rtcp v1.2.15
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github.com/pion/rtp v1.8.20
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github.com/pion/sdp/v3 v3.0.14
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github.com/pion/transport/v3 v3.0.7
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github.com/pion/webrtc/v4 v4.1.3
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github.com/pkg/errors v0.9.1
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github.com/yapingcat/gomedia/codec v0.0.0-20220617074658-94762898dc25
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github.com/yapingcat/gomedia/mpeg2 v0.0.0-20220617074658-94762898dc25
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)
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require (
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github.com/bluenviron/mediacommon/v2 v2.1.0 // indirect
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github.com/gobwas/httphead v0.1.0 // indirect
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github.com/gobwas/pool v0.2.1 // indirect
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github.com/gobwas/ws v1.1.0-rc.1 // indirect
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github.com/google/uuid v1.6.0 // indirect
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github.com/konsorten/go-windows-terminal-sequences v1.0.2 // indirect
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github.com/mattn/go-colorable v0.1.4 // indirect
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github.com/mattn/go-isatty v0.0.8 // indirect
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github.com/mattn/go-pointer v0.0.1 // indirect
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github.com/mgutz/ansi v0.0.0-20170206155736-9520e82c474b // indirect
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github.com/pion/datachannel v1.5.10 // indirect
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github.com/pion/dtls/v3 v3.0.6 // indirect
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github.com/pion/mdns/v2 v2.0.7 // indirect
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github.com/pion/randutil v0.1.0 // indirect
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github.com/pion/sctp v1.8.39 // indirect
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github.com/pion/srtp/v3 v3.0.6 // indirect
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github.com/pion/stun/v3 v3.0.0 // indirect
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github.com/pion/turn/v4 v4.0.2 // indirect
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github.com/satori/go.uuid v1.2.1-0.20181028125025-b2ce2384e17b // indirect
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github.com/sirupsen/logrus v1.4.2 // indirect
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github.com/tevino/abool v0.0.0-20170917061928-9b9efcf221b5 // indirect
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github.com/wlynxg/anet v0.0.5 // indirect
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github.com/x-cray/logrus-prefixed-formatter v0.5.2 // indirect
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golang.org/x/crypto v0.39.0 // indirect
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golang.org/x/net v0.41.0 // indirect
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golang.org/x/sys v0.33.0 // indirect
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golang.org/x/term v0.32.0 // indirect
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)
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