## 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>
For H.264, only when the NAL Type is 1, 2, 3, or 4 is it possible for
B-frames to be present; that is, non-IDR pictures and slice data.
The current `SrsVideoFrame::parse_avc_bframe()` function uses incorrect
logic to determine if a NALU can contain B-frames. The original
implementation only checked for specific NALU types (IDR, SPS, PPS) to
mark as non-B-frames, but this approach misses many other NALU types
that cannot contain B-frames according to the H.264 specification.
According to H.264 specification (ISO_IEC_14496-10-AVC-2012.pdf, Table
7-1), B-frames can **only** exist in these specific NALU types:
- Type 1: Non-IDR coded slice (`SrsAvcNaluTypeNonIDR`)
- Type 2: Coded slice data partition A (`SrsAvcNaluTypeDataPartitionA`)
- Type 3: Coded slice data partition B (`SrsAvcNaluTypeDataPartitionB`)
- Type 4: Coded slice data partition C (`SrsAvcNaluTypeDataPartitionC`)
All other NALU types (IDR=5, SEI=6, SPS=7, PPS=8, AUD=9, etc.) cannot
contain B-frames by definition.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
This PR refactors the RTMP to RTC bridge to support multiple video
tracks by implementing lazy initialization of audio and video tracks.
Instead of pre-determining track parameters during bridge construction,
tracks are now initialized dynamically when the first packet of each
type is received, allowing proper codec detection and track
configuration for dual video track scenarios.
Failed to view WHEP with HEVC before publishing RTMP, because the
default codec is AVC and will not be updated until the stream is
published. This enables better handling of streams with multiple video
tracks in RTMP to WebRTC bridging scenarios. Now, you are able to:
1. View WHEP with HEVC: Play with WebRTC:
http://localhost:8080/players/whep.html?schema=http&&codec=hevc
2. Then publish by RTMP: `ffmpeg -stream_loop -1 -re -i doc/source.flv
-c:v libx265 -profile:v main -preset fast -b:v 2000k -maxrate 2000k
-bufsize 2000k -bf 0 -c:a aac -b:a 48k -ar 44100 -ac 2 -f flv
rtmp://localhost/live/livestream`
Or publish RTMP with HEVC, then view by WHEP.
Note that if the codecs do not match, the error log will display RTC:
`Drop for ssrc xxxxxx not found`. For example, this can occur if you
publish RTMP with HEVC while viewing the stream with AVC.
**Introduce**
This pull request builds upon the foundation laid in
https://github.com/ossrs/srs/pull/4289 . While the previous work solely
implemented unidirectional HEVC support from RTMP to RTC, this
submission further enhances it by introducing support for the RTC to
RTMP direction.
**Usage**
Launch SRS with `rtc2rtmp.conf`
```bash
./objs/srs -c conf/rtc2rtmp.conf
```
**Push with WebRTC**
Upgrade browser to Chrome(136+) or Safari(18+), then open [WHIP
encoder](http://localhost:8080/players/whip.html?schema=http&&codec=hevc),
push stream with URL that enables HEVC by query string `codec=hevc`:
```bash
http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&codec=hevc
```
This query string `codec=hevc` is used to select the video codec, and
generate lines in the answer SDP.
```
m=video 9 UDP/TLS/RTP/SAVPF 49 123
a=rtpmap:49 H265/90000
```
The encoder log also show the codec:
```
Audio: opus, 48000HZ, channels: 2, pt: 111
Video: H265, 90000HZ, pt: 49
```
**Play with RTMP**
Play HEVC stream via RTMP.
```bash
ffplay -i rtmp://localhost/live/livestream
```
You will see the codec in logs:
```
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream #0:1: Video: hevc (Main), yuv420p(tv, bt709), 320x240, 30 fps, 30 tbr, 1k tbn
```
You can also use [WHEP
player](http://localhost:8080/players/whep.html?schema=http&&codec=hevc)
to play the stream.
Important refactor with AI:
* [AI: Refactor packet cache for RTC frame
builder.](b8ffa1630e)
* [AI: Refactor the packet copy and free for
SrsRtcFrameBuilder](f3487b45d7)
* [AI: Refactor the frame detector for
SrsRtcFrameBuilder](4ffc1526b9)
* [AI: Refactor the packet_video_rtmp for
SrsRtcFrameBuilder](81f6aef4ed)
* [AI: Add utests for
SrsCodecPayload.codec](61eb1c0bfc)
* [AI: Add utests for VideoPacketCache in
SrsRtcFrameBuilder.](fd25480dfa)
* [AI: Add utests for VideoFrameDetector in
SrsRtcFrameBuilder.](b4aa977bbd)
* [AI: Add regression test for RTC2RTMP with
HEVC.](5259a2aac3)
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
1. It cannot retrieve codec information on `Firefox` by
`getSenders/getReceivers`
2. It can retrieve codec information on `Chrome` by `getReceivers`, but
incorrect, like this:
3. So, we retrieve codec information from `getStats`, and it works well.
4. The timer is used because sometimes the codec cannot be retrieved
when `iceGatheringState` is `complete`.
5. Testing has been completed on the browsers listed below.
- [x] Chrome
- [x] Edge
- [x] Safari
- [x] Firefox
---------
Co-authored-by: winlin <winlinvip@gmail.com>
1. When the chunk message header employs type 1 and type 2, the extended
timestamp denotes the time delta.
2. When the DTS (Decoding Time Stamp) experiences a jump and exceeds
16777215, there can be errors in DTS calculation, and if the audio and
video delta differs, it may result in audio-video synchronization
issues.
---------
`TRANS_BY_GPT4`
---------
Co-authored-by: 彭治湘 <zuolengchan@douyu.tv>
Co-authored-by: Haibo Chen(陈海博) <495810242@qq.com>
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
To enable H.265 support for the WebRTC protocol, upgrade the pion/webrtc
library to version 4.
---------
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: winlin <winlinvip@gmail.com>
SrsUniquePtr does not support array or object created by malloc, because
we only use delete to dispose the resource. You can use a custom
function to free the memory allocated by malloc or other allocators.
```cpp
char* p = (char*)malloc(1024);
SrsUniquePtr<char> ptr(p, your_free_chars);
```
This is used to replace the SrsAutoFreeH. For example:
```cpp
addrinfo* r = NULL;
SrsAutoFreeH(addrinfo, r, freeaddrinfo);
getaddrinfo("127.0.0.1", NULL, &hints, &r);
```
Now, this can be replaced by:
```cpp
addrinfo* r = NULL;
getaddrinfo("127.0.0.1", NULL, &hints, &r);
SrsUniquePtr<addrinfo> r2(r, freeaddrinfo);
```
Please aware that there is a slight difference between SrsAutoFreeH and
SrsUniquePtr. SrsAutoFreeH will track the address of pointer, while
SrsUniquePtr will not.
```cpp
addrinfo* r = NULL;
SrsAutoFreeH(addrinfo, r, freeaddrinfo); // r will be freed even r is changed later.
SrsUniquePtr<addrinfo> ptr(r, freeaddrinfo); // crash because r is an invalid pointer.
```
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: john <hondaxiao@tencent.com>
```
../../../src/utest/srs_utest_st.cpp:27: Failure
Expected: (st_time_2 - st_time_1) <= (100), actual: 119 vs 100
[ FAILED ] StTest.StUtimeInMicroseconds (0 ms)
```
Maybe github's vm, running the action jobs, is slower. I notice this
error happens frequently, so let the UT pass by increase the number.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: winlin <winlinvip@gmail.com>
