srs/trunk/3rdparty/srs-bench/blackbox/rtsp_test.go
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

120 lines
3.7 KiB
Go

// The MIT License (MIT)
//
// # Copyright (c) 2025 Winlin
//
// Permission is hereby granted, free of charge, to any person obtaining a copy of
// this software and associated documentation files (the "Software"), to deal in
// the Software without restriction, including without limitation the rights to
// use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
// the Software, and to permit persons to whom the Software is furnished to do so,
// subject to the following conditions:
//
// The above copyright notice and this permission notice shall be included in all
// copies or substantial portions of the Software.
//
// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
// FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
// COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
// IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
// CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
package blackbox
import (
"context"
"fmt"
"math/rand"
"os"
"path"
"sync"
"testing"
"time"
"github.com/ossrs/go-oryx-lib/errors"
"github.com/ossrs/go-oryx-lib/logger"
)
func TestFast_RtmpPublish_RtspPlay_Basic(t *testing.T) {
// This case is run in parallel.
t.Parallel()
// Setup the max timeout for this case.
ctx, cancel := context.WithTimeout(logger.WithContext(context.Background()), time.Duration(*srsTimeout)*time.Millisecond)
defer cancel()
// Check a set of errors.
var r0, r1, r2, r3, r4, r5 error
defer func(ctx context.Context) {
if err := filterTestError(ctx.Err(), r0, r1, r2, r3, r4, r5); err != nil {
t.Errorf("Fail for err %+v", err)
} else {
logger.Tf(ctx, "test done with err %+v", err)
}
}(ctx)
var wg sync.WaitGroup
defer wg.Wait()
// Start SRS server and wait for it to be ready.
svr := NewSRSServer(func(v *srsServer) {
v.envs = []string{
"SRS_RTSP_SERVER_ENABLED=on",
"SRS_VHOST_RTSP_ENABLED=on",
"SRS_VHOST_RTSP_RTMP_TO_RTSP=on",
}
})
wg.Add(1)
go func() {
defer wg.Done()
r0 = svr.Run(ctx, cancel)
}()
// Start FFmpeg to publish stream.
streamID := fmt.Sprintf("stream-%v-%v", os.Getpid(), rand.Int())
streamURL := fmt.Sprintf("rtmp://localhost:%v/live/%v", svr.RTMPPort(), streamID)
ffmpeg := NewFFmpeg(func(v *ffmpegClient) {
v.args = []string{
"-stream_loop", "-1", "-re", "-i", *srsPublishAvatar, "-c", "copy", "-f", "flv", streamURL,
}
})
wg.Add(1)
go func() {
defer wg.Done()
<-svr.ReadyCtx().Done()
r1 = ffmpeg.Run(ctx, cancel)
}()
// Start FFprobe to detect and verify stream.
duration := time.Duration(*srsFFprobeDuration) * time.Millisecond
ffprobe := NewFFprobe(func(v *ffprobeClient) {
v.dvrFile = path.Join(svr.WorkDir(), "objs", fmt.Sprintf("srs-ffprobe-%v.mp4", streamID))
v.streamURL = fmt.Sprintf("rtsp://localhost:%v/live/%v", svr.RTSPPort(), streamID)
v.duration, v.timeout = duration, time.Duration(*srsFFprobeTimeout)*time.Millisecond
})
wg.Add(1)
go func() {
defer wg.Done()
<-svr.ReadyCtx().Done()
r2 = ffprobe.Run(ctx, cancel)
}()
// Fast quit for probe done.
select {
case <-ctx.Done():
case <-ffprobe.ProbeDoneCtx().Done():
defer cancel()
str, m := ffprobe.Result()
if len(m.Streams) != 2 {
r3 = errors.Errorf("invalid streams=%v, %v, %v", len(m.Streams), m.String(), str)
}
if ts := 90; m.Format.ProbeScore < ts {
r4 = errors.Errorf("low score=%v < %v, %v, %v", m.Format.ProbeScore, ts, m.String(), str)
}
if dv := m.Duration(); dv < duration {
r5 = errors.Errorf("short duration=%v < %v, %v, %v", dv, duration, m.String(), str)
}
}
}