NEW PROTOCOL: Support viewing stream over RTSP. v7.0.47

[duiniuluantanqin]

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.
• AI: Add crutial logs for each RTSP stage.
• AI: Support AAC doec for RTSP.
• AI: Add option --rtsp for RTSP.
• AI: Extract SrsRtpVideoBuilder for RTC and RTSP.

---

Co-authored-by: Jacob Su [email protected]

[Copilot]

Pull Request Overview

This PR adds support for RTSP stream pulling by introducing a new RTSP protocol stack and connection type, updating server setup and configuration, and extending SDP processing.

• Added unit tests in the configuration suite to verify RTSP server environment variables.
• Introduced a new RTSP protocol stack header and corresponding connection implementation (SrsRtspConn).
• Updated server initialization, heartbeat reporting, and configuration handling to incorporate RTSP endpoints.

Reviewed Changes

Copilot reviewed 15 out of 17 changed files in this pull request and generated no comments.

Show a summary per file

File	Description
trunk/src/utest/srs_utest_config.cpp	Added unit test for RTSP configuration environment variables.
trunk/src/protocol/srs_protocol_rtsp_stack.hpp	Introduced RTSP protocol stack definitions and constants.
trunk/src/app/srs_app_server.hpp & .cpp	Added RTSP listener initialization, management, and integration into connection dispatching.
trunk/src/app/srs_app_rtsp.hpp	Defined SrsRtspConn and related RTSP connection functionality.
trunk/src/app/srs_app_rtc_sdp.hpp & .cpp	Extended SDP encoding to include RTSP control attributes.
trunk/src/app/srs_app_rtc_conn.hpp	Updated do_send_packet to be virtual for proper RTSP connection overrides.
trunk/src/app/srs_app_latest_version.cpp, srs_app_heartbeat.cpp, srs_app_config.hpp & .cpp	Integrated RTSP feature flags and configuration adjustments.

Files not reviewed (2)

• trunk/conf/rtsp.conf: Language not supported
• trunk/configure: Language not supported