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srs/trunk/3rdparty/srs-bench/vendor/modules.txt
Haibo Chen(陈海博) 5dc292ce64
NEW PROTOCOL: Support viewing stream over RTSP. v7.0.47 (#4333) 
## 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>
2025-07-11 08:18:40 -04:00

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# github.com/bluenviron/gortsplib/v4 v4.13.1
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github.com/bluenviron/gortsplib/v4/pkg/base
github.com/bluenviron/gortsplib/v4/pkg/bytecounter
github.com/bluenviron/gortsplib/v4/pkg/conn
github.com/bluenviron/gortsplib/v4/pkg/description
github.com/bluenviron/gortsplib/v4/pkg/format
github.com/bluenviron/gortsplib/v4/pkg/format/rtpac3
github.com/bluenviron/gortsplib/v4/pkg/format/rtpav1
github.com/bluenviron/gortsplib/v4/pkg/format/rtph264
github.com/bluenviron/gortsplib/v4/pkg/format/rtph265
github.com/bluenviron/gortsplib/v4/pkg/format/rtplpcm
github.com/bluenviron/gortsplib/v4/pkg/format/rtpmjpeg
github.com/bluenviron/gortsplib/v4/pkg/format/rtpmpeg1audio
github.com/bluenviron/gortsplib/v4/pkg/format/rtpmpeg1video
github.com/bluenviron/gortsplib/v4/pkg/format/rtpmpeg4audio
github.com/bluenviron/gortsplib/v4/pkg/format/rtpmpeg4video
github.com/bluenviron/gortsplib/v4/pkg/format/rtpsimpleaudio
github.com/bluenviron/gortsplib/v4/pkg/format/rtpvp8
github.com/bluenviron/gortsplib/v4/pkg/format/rtpvp9
github.com/bluenviron/gortsplib/v4/pkg/headers
github.com/bluenviron/gortsplib/v4/pkg/liberrors
github.com/bluenviron/gortsplib/v4/pkg/multicast
github.com/bluenviron/gortsplib/v4/pkg/ringbuffer
github.com/bluenviron/gortsplib/v4/pkg/rtcpreceiver
github.com/bluenviron/gortsplib/v4/pkg/rtcpsender
github.com/bluenviron/gortsplib/v4/pkg/rtplossdetector
github.com/bluenviron/gortsplib/v4/pkg/rtpreorderer
github.com/bluenviron/gortsplib/v4/pkg/rtptime
github.com/bluenviron/gortsplib/v4/pkg/sdp
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github.com/bluenviron/mediacommon/v2/pkg/codecs/h264
github.com/bluenviron/mediacommon/v2/pkg/codecs/h265
github.com/bluenviron/mediacommon/v2/pkg/codecs/jpeg
github.com/bluenviron/mediacommon/v2/pkg/codecs/mpeg1audio
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# github.com/ghettovoice/gosip v0.0.0-20220929080231-de8ba881be83
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# github.com/ossrs/go-oryx-lib v0.0.9
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github.com/ossrs/go-oryx-lib/flv
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github.com/pion/dtls/v3/pkg/crypto/ciphersuite
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github.com/pion/webrtc/v4/internal/mux
github.com/pion/webrtc/v4/internal/util
github.com/pion/webrtc/v4/pkg/media
github.com/pion/webrtc/v4/pkg/media/h264reader
github.com/pion/webrtc/v4/pkg/media/h264writer
github.com/pion/webrtc/v4/pkg/media/h265reader
github.com/pion/webrtc/v4/pkg/media/ivfwriter
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# golang.org/x/crypto v0.39.0
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# golang.org/x/term v0.32.0
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