RTSP stands for Real-Time Streaming Protocol, standardized by RFC 7826. It's commonly used to stream audio and video from IP cameras.
In GStreamer, RTSP handling was traditionally done by rtspsrc, a widely used and mature element, but it still had several architectural limitations that were hard to address within the existing codebase.
rtspsrc ties the server state to client/pipeline state. This resulted in issues with
- Multicast playback
- PAUSE being sent on error, immediately followed by tear down
- Flushing seeks involved state changes causing glitches and possible deadlock
As a result, the element has been rewritten from scratch in Rust as rtspsrc2. See this GStreamer Conference Talk and README for more details.
Until now, some features were missing, so if your application relied on them, rtspsrc2 couldn't replace the original.
Many of these have now been implemented:
Authentication
Basic and Digest authentication are supported. Digest covers MD5, SHA-256 and SHA-512-256, so rtspsrc2 can connect to servers that require a username/password and can perform the challenge response handshake automatically.
TLS/TCP support
rtspsrc2 can connect over TLS (the rtsps:// scheme) and supports client certificate authentication for servers that require it. Like rtspsrc, it also has tls-validation-flags property to ignore specific certificate errors like expired certificates.
HTTP tunnelling
Some networks block RTSP but allow HTTP. rtspsrc2 can tunnel RTSP over HTTP (rtsph://) and exposes an extra-http-request-headers property for custom headers
(useful for proxies or authentication).
Keep alive
RTSP keep-alive is required because RTSP session state is entirely independent of the underlying transport. During active playback, RTCP serves as the primary proof of liveness. However, during paused states when media and RTCP traffic stop, explicit RTSP keep-alive requests are needed to prevent session timeout.
rtspsrc2 sends periodic keep-alive to prevent session timeout. This is enabled by default via do-rtsp-keep-alive. Disable it if you need compatibility with older servers.
Stream selection
This is one major feature that rtspsrc doesn't implement. rtspsrc2 is streams-aware.
When a source contains multiple streams (for example audio and video), rtspsrc2 exposes them as GstStream objects so applications can choose which streams to set up before playback using the StreamCollection API. For uses of this API by other elements, also see adaptive demuxers.
Consider an example scenario where the RTSP stream has audio and video. An application can choose to ignore audio and select only video via the stream collection API. rtspsrc2 then sends a SETUP request only for video and not audio. An example showing how this can be done in Rust can be seen here.
Unlinked pads
Linking of exposed source pads is now optional, for example, if you only link the video pad and leave the audio pad unlinked, rtspsrc2 handles that without error. Internally, it uses a flow combiner like the original rtspsrc, so selective linking works as expected.
GET/SET_PARAMETER
GET_PARAMETER and SET_PARAMETER requests are supported using signals similar to rtspsrc. These requests are used to retrieve or set the value of a parameter or a set of parameters for a presentation or stream specified by the URI.
Secure Real-time Transport Protocol (SRTP)
SRTP adds encryption and authentication to RTP streams. The RTSP source internally instantiates an RTP session manager element that handles the messages to and from the server, jitter removal, packet reordering along with providing a clock for the pipeline.
While rtspsrc uses rtpbin, rtspsrc2 can use rtpbin or the newer rtpbin2. rtpbin2 unlike rtpbin splits out the sender and receiver RTP session management in two separate elements rtpsend and rtprecv. See this GStreamer Conference Talk for understanding the motivations behind the new RTP elements. By setting the environment variable USE_RTP2=1, rtspsrc2 can use rtpsend and rtprecv.
Secure RTP (SRTP) as of this writing, is only supported when using rtspsrc2 with rtpbin, including MIKEY key exchange embedded in SDP.
With these additions, rtspsrc2 is much closer to being a drop-in replacement for rtspsrc. If you haven't tried it yet, now is a good time.
A GStreamer pipeline to test rtspsrc2 given a RTSP server address.
gst-launch-1.0 -e -vvv rtspsrc2 location=rtsp://some.server/url ! queue ! decodebin ! queue ! videoconvert ! autovideosinkThe following key features still need to be implemented before rtspsrc2 achieves complete feature parity with rtspsrc.