# General Discussion

**URL:** https://discourse.gstreamer.org/c/general/4.md?page=2

[Latest](https://discourse.gstreamer.org/latest.md) · [Categories](https://discourse.gstreamer.org/categories.md) · [Tags](https://discourse.gstreamer.org/tags.md)

**Page:** 3

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## [Webrtcbin generating invalid answer](https://discourse.gstreamer.org/t/webrtcbin-generating-invalid-answer/4441)

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**Author:** [@vr7bd](https://discourse.gstreamer.org/u/vr7bd)\
**Replies:** 13\
**Last updated:** [December 31, 2025, 1:52pm UTC](https://discourse.gstreamer.org/t/webrtcbin-generating-invalid-answer/4441 "2025-12-31T13:52:18Z")

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I have an application where my gstreamer app in rust is behaving as the receiver and the browser behaves as the sender. I have functionality in the sender that can stop and start screen sharing. While connecting, the se…

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## [Why rtp socket in alloc\_ports\_one\_family() in rtsp-stream.c need bind first?](https://discourse.gstreamer.org/t/why-rtp-socket-in-alloc-ports-one-family-in-rtsp-stream-c-need-bind-first/5573)

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**Author:** [@snew310](https://discourse.gstreamer.org/u/snew310)\
**Replies:** 0\
**Last updated:** [December 26, 2025, 11:12am UTC](https://discourse.gstreamer.org/t/why-rtp-socket-in-alloc-ports-one-family-in-rtsp-stream-c-need-bind-first/5573 "2025-12-26T11:12:00Z")

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Hello, expert, In the alloc\_ports\_one\_family() function, I understand why RTCP needs to bind to a specific IP address and port for the client to send data. However, since RTP sockets are sent to the client, why do they…

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## [Playback gap during pasuse](https://discourse.gstreamer.org/t/playback-gap-during-pasuse/5564)

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**Author:** [@deepak191093](https://discourse.gstreamer.org/u/deepak191093)\
**Replies:** 0\
**Last updated:** [December 15, 2025, 6:34am UTC](https://discourse.gstreamer.org/t/playback-gap-during-pasuse/5564 "2025-12-15T06:34:31Z")

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i am trying to pause my pipeline but when new buffer comes it comes with time which inclued pause time what can i do so when i play the i dont see blank play during pause time

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## [RTSP streams to WebRTC using GStreamer](https://discourse.gstreamer.org/t/rtsp-streams-to-webrtc-using-gstreamer/5561)

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**Author:** [@abhay9989](https://discourse.gstreamer.org/u/abhay9989)\
**Replies:** 1\
**Last updated:** [December 13, 2025, 9:34pm UTC](https://discourse.gstreamer.org/t/rtsp-streams-to-webrtc-using-gstreamer/5561 "2025-12-13T21:34:00Z")

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Has anyone successfully converted RTSP streams to WebRTC using GStreamer without adding noticeable delay? I am trying to achieve sub-200ms real-time playback in browsers. I found this article saying that this multi prot…

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## [How to send & receive RTP on the same UDP port (Go + GStreamer + FreeSWITCH)?](https://discourse.gstreamer.org/t/how-to-send-receive-rtp-on-the-same-udp-port-go-gstreamer-freeswitch/5507)

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**Author:** [@ggwebrc](https://discourse.gstreamer.org/u/ggwebrc)\
**Replies:** 9\
**Last updated:** [December 3, 2025, 7:24pm UTC](https://discourse.gstreamer.org/t/how-to-send-receive-rtp-on-the-same-udp-port-go-gstreamer-freeswitch/5507 "2025-12-03T19:24:51Z")

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Hi all, I’m dealing with an RTP port issue involving FreeSWITCH, a Go RTP listener, and GStreamer. Scenario My Go app listens for RTP on a specific UDP port A. GStreamer needs to send RTP from the same source port …

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## [Android Examples updated for Android Studio](https://discourse.gstreamer.org/t/android-examples-updated-for-android-studio/1159)

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**Author:** [@almita](https://discourse.gstreamer.org/u/almita)\
**Replies:** 2\
**Last updated:** [December 1, 2025, 7:22am UTC](https://discourse.gstreamer.org/t/android-examples-updated-for-android-studio/1159 "2025-12-01T07:22:22Z")

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Hello! I hope you´re doing great. I´m very very new in Gstreamer and I need to play a RTSP with h265 for Android mobile. I donwload the examples to have a reference that can work and I found a couple of issues. The …

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## [Rtspsrc: EOS when server changes RTP SSRC mid‑RTSP session (Parrot Anafi drone) – can it follow the new SSRC?](https://discourse.gstreamer.org/t/rtspsrc-eos-when-server-changes-rtp-ssrc-mid-rtsp-session-parrot-anafi-drone-can-it-follow-the-new-ssrc/5538)

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**Author:** [@Rz-Rz](https://discourse.gstreamer.org/u/Rz-Rz)\
**Replies:** 0\
**Last updated:** [November 26, 2025, 3:28pm UTC](https://discourse.gstreamer.org/t/rtspsrc-eos-when-server-changes-rtp-ssrc-mid-rtsp-session-parrot-anafi-drone-can-it-follow-the-new-ssrc/5538 "2025-11-26T15:28:46Z")

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Hi, I have an RTSP client built with GStreamer talking to a Parrot Anafi USA drone, and I am trying to keep the live video stream running while triggering still photos via a separate control channel (Parrot Olympe API). …

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## [How to solve the error when using gstremaer to read h264 file and send it via avtp plugin?](https://discourse.gstreamer.org/t/how-to-solve-the-error-when-using-gstremaer-to-read-h264-file-and-send-it-via-avtp-plugin/4321)

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**Author:** [@xyy0324](https://discourse.gstreamer.org/u/xyy0324)\
**Replies:** 3\
**Last updated:** [November 26, 2025, 8:11am UTC](https://discourse.gstreamer.org/t/how-to-solve-the-error-when-using-gstremaer-to-read-h264-file-and-send-it-via-avtp-plugin/4321 "2025-11-26T08:11:18Z")

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run cmd：./gst-launch-1.0 filesrc location=sample\_outdoor\_car\_1080p\_10fps.h264 do-timestamp=true ! h264parse ! avtpcvfpay ! avtpsink ifname=lo error： Setting pipeline to PAUSED … Pipeline is PREROLLING … \*\* ERROR:…/s…

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## [Splitmuxsink Chunk Size](https://discourse.gstreamer.org/t/splitmuxsink-chunk-size/5536)

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**Author:** [@danijel00](https://discourse.gstreamer.org/u/danijel00)\
**Replies:** 2\
**Last updated:** [November 25, 2025, 7:26am UTC](https://discourse.gstreamer.org/t/splitmuxsink-chunk-size/5536 "2025-11-25T07:26:04Z")

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Hello, I’m using the splitmuxsink element to create an MP4 video. If I set the property max-size-bytes=8,000,000, I get video chunks with a size of 6.3–6.5 MB. Why is this happening? According to the documentation, 1 M…

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## [Android plugins not found (Unity)](https://discourse.gstreamer.org/t/android-plugins-not-found-unity/5533)

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**Author:** [@xovobobo](https://discourse.gstreamer.org/u/xovobobo)\
**Replies:** 1\
**Last updated:** [November 23, 2025, 1:27pm UTC](https://discourse.gstreamer.org/t/android-plugins-not-found-unity/5533 "2025-11-23T13:27:06Z")

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Hello. Im trying to integrate gstreamer into a Unity application. It works great on Linux (with gstreamer from the apt repository), but now i\`m having problems with integration on android. I built gstreamer for Androud …

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## [Can GStreamer Receive Multiple RTP Streams on the Same UDP Port?](https://discourse.gstreamer.org/t/can-gstreamer-receive-multiple-rtp-streams-on-the-same-udp-port/5522)

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**Author:** [@ggwebrc](https://discourse.gstreamer.org/u/ggwebrc)\
**Replies:** 2\
**Last updated:** [November 21, 2025, 12:52am UTC](https://discourse.gstreamer.org/t/can-gstreamer-receive-multiple-rtp-streams-on-the-same-udp-port/5522 "2025-11-21T00:52:59Z")

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Hello everyone, I’m exploring an RTP processing setup and I have a question about GStreamer’s capabilities. Is it possible for GStreamer to receive multiple independent RTP streams on the same UDP port and still be abl…

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## [How can I prevent EOS when using multifilesrc to loop video?](https://discourse.gstreamer.org/t/how-can-i-prevent-eos-when-using-multifilesrc-to-loop-video/5525)

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**Author:** [@sunshinechaos](https://discourse.gstreamer.org/u/sunshinechaos)\
**Replies:** 0\
**Last updated:** [November 19, 2025, 8:28am UTC](https://discourse.gstreamer.org/t/how-can-i-prevent-eos-when-using-multifilesrc-to-loop-video/5525 "2025-11-19T08:28:28Z")

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I’m using gstreamer 1.20.7 in windows platform, trying to loop a video to rtsp server. gst-launch-1.0 -e \\ multifilesrc location=“prod-test.h264” loop=1 \\ caps=“video/x-h264,stream-format=byte-stream” \\ ! h264parse …

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## [Building a gstreamer pipeline to cast directly in web browser](https://discourse.gstreamer.org/t/building-a-gstreamer-pipeline-to-cast-directly-in-web-browser/5518)

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**Author:** [@Joshua](https://discourse.gstreamer.org/u/Joshua)\
**Replies:** 2\
**Last updated:** [November 15, 2025, 10:07pm UTC](https://discourse.gstreamer.org/t/building-a-gstreamer-pipeline-to-cast-directly-in-web-browser/5518 "2025-11-15T22:07:45Z")

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Hi, I’m working on a project that requires streaming video output from a GStreamer pipeline to a web browser. I am looking into WebRTC as this seems to be the simplest solution to implement. I want to build a GStreamer…

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## [Selecting audio track from MKV file fails in playbin3](https://discourse.gstreamer.org/t/selecting-audio-track-from-mkv-file-fails-in-playbin3/5513)

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**Author:** [@akanashin](https://discourse.gstreamer.org/u/akanashin)\
**Replies:** 3\
**Last updated:** [November 13, 2025, 7:39pm UTC](https://discourse.gstreamer.org/t/selecting-audio-track-from-mkv-file-fails-in-playbin3/5513 "2025-11-13T19:39:58Z")

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Hey! I’m using playbin3 to play MKV file. What i have found is that playbin3 is unable to select an audio stream from this file. Externally it looks like property current-audio (I used other approach in C++ with messag…

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## [Prerecording Data using Gstreamer / Circular Buffer in Gstreamer](https://discourse.gstreamer.org/t/prerecording-data-using-gstreamer-circular-buffer-in-gstreamer/949)

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**Author:** [@dongremayur777](https://discourse.gstreamer.org/u/dongremayur777)\
**Replies:** 4\
**Last updated:** [November 13, 2025, 12:51pm UTC](https://discourse.gstreamer.org/t/prerecording-data-using-gstreamer-circular-buffer-in-gstreamer/949 "2025-11-13T12:51:27Z")

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What I want is to record last n seconds before trigger occurs , means suppose i have set n as 30 sec so the data would be continuously storing in a circular buffers which will have last n secons and when trigger occurs f…

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## [Choosing protocol for pipeline communication](https://discourse.gstreamer.org/t/choosing-protocol-for-pipeline-communication/5515)

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**Author:** [@sannet](https://discourse.gstreamer.org/u/sannet)\
**Replies:** 1\
**Last updated:** [November 13, 2025, 10:20am UTC](https://discourse.gstreamer.org/t/choosing-protocol-for-pipeline-communication/5515 "2025-11-13T10:20:05Z")

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I am currently working with a basic gstremer pipeline which pre builded. Pipeline structure is not customizable. However, there is a class (located in a script file) and it consists of 2 different methods (\_init\_, \_call\_…

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## [Building static gstreamer with gst-plugin-webrtchttp plugin](https://discourse.gstreamer.org/t/building-static-gstreamer-with-gst-plugin-webrtchttp-plugin/5499)

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**Author:** [@murugs](https://discourse.gstreamer.org/u/murugs)\
**Replies:** 3\
**Last updated:** [November 7, 2025, 5:48am UTC](https://discourse.gstreamer.org/t/building-static-gstreamer-with-gst-plugin-webrtchttp-plugin/5499 "2025-11-07T05:48:40Z")

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Trying to build static libraries for gstreamer and gst-plugins-rs. While running gst-inspect the plugin in looking for shared library instead of static gstreamer. git clone --recurse-submodules https://gitlab.freedeskt…

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## [MP4 playback starts ahead when dynamically added to running pipeline](https://discourse.gstreamer.org/t/mp4-playback-starts-ahead-when-dynamically-added-to-running-pipeline/5490)

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**Author:** [@gcube](https://discourse.gstreamer.org/u/gcube)\
**Replies:** 3\
**Last updated:** [November 4, 2025, 3:17pm UTC](https://discourse.gstreamer.org/t/mp4-playback-starts-ahead-when-dynamically-added-to-running-pipeline/5490 "2025-11-04T15:17:54Z")

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Dear GStreamer community, I’m trying to play a local MP4 video that is added to a pipeline after the pipeline has already started. The main pipeline starts by playing an SRT stream, which is linked to a compositor elem…

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## [m3u8 to mpeg1video](https://discourse.gstreamer.org/t/m3u8-to-mpeg1video/5493)

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**Author:** [@fatihdeveci05-creato](https://discourse.gstreamer.org/u/fatihdeveci05-creato)\
**Replies:** 3\
**Last updated:** [November 4, 2025, 10:59am UTC](https://discourse.gstreamer.org/t/m3u8-to-mpeg1video/5493 "2025-11-04T10:59:47Z")

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Hello, I want to convert TV channels’ m3u8 broadcasts to mpeg1 and publish them on my site. Latency isn’t a concern, and image quality isn’t that important. Can I do this with Gstreamer? I used FFMPEG, but I experienced …

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## [High cpu usage!](https://discourse.gstreamer.org/t/high-cpu-usage/5487)

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**Author:** [@ggwebrc](https://discourse.gstreamer.org/u/ggwebrc)\
**Replies:** 1\
**Last updated:** [November 2, 2025, 9:18pm UTC](https://discourse.gstreamer.org/t/high-cpu-usage/5487 "2025-11-02T21:18:28Z")

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Hello guys, I’m currently running the following GStreamer pipeline to mix two VP8 video streams and play audio. The problem is that the CPU usage is kind of high, likely because of decoding, scaling, and compositing VP…

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## [Gstreamer and webrtc](https://discourse.gstreamer.org/t/gstreamer-and-webrtc/5480)

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**Author:** [@ggwebrc](https://discourse.gstreamer.org/u/ggwebrc)\
**Replies:** 4\
**Last updated:** [October 29, 2025, 1:52pm UTC](https://discourse.gstreamer.org/t/gstreamer-and-webrtc/5480 "2025-10-29T13:52:29Z")

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Hey everyone! I’m new here and was wondering if anyone has experimented with GStreamer and WebRTC. Is it possible to use GStreamer to mix multiple video streams from a peer connection into a single grid-style video (an…

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## [Strange behaviour of fakesink: does not start and blocks the whole chain?](https://discourse.gstreamer.org/t/strange-behaviour-of-fakesink-does-not-start-and-blocks-the-whole-chain/5485)

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**Author:** [@akanashin](https://discourse.gstreamer.org/u/akanashin)\
**Replies:** 1\
**Last updated:** [October 29, 2025, 5:51am UTC](https://discourse.gstreamer.org/t/strange-behaviour-of-fakesink-does-not-start-and-blocks-the-whole-chain/5485 "2025-10-29T05:51:26Z")

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Hi! I’m having strange issue with pipeline ending with fakesink. The pipeline looks like this (after removing capsfilters): decklinkteletextsrc ! tee name=t1 ! teletextdec ! appsink t1 ! fakesink Usually (\*) the p…

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## [Are there any issues when switching from a GSTBaseTransform-based project to a GST Element approach?](https://discourse.gstreamer.org/t/are-there-any-issues-when-switching-from-a-gstbasetransform-based-project-to-a-gst-element-approach/5477)

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**Author:** [@kty1210](https://discourse.gstreamer.org/u/kty1210)\
**Replies:** 0\
**Last updated:** [October 25, 2025, 9:31am UTC](https://discourse.gstreamer.org/t/are-there-any-issues-when-switching-from-a-gstbasetransform-based-project-to-a-gst-element-approach/5477 "2025-10-25T09:31:21Z")

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When I pass a DMA buffer to the next element using the gstbasetransform class, everything works fine. However, when I implemented the next element as a regular gstelement (not using the gstbasetransform framework) and s…

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## [Unable to build gstreamer using Meson on Windows](https://discourse.gstreamer.org/t/unable-to-build-gstreamer-using-meson-on-windows/4454)

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**Author:** [@floppyhammer](https://discourse.gstreamer.org/u/floppyhammer)\
**Replies:** 8\
**Last updated:** [October 25, 2025, 9:02am UTC](https://discourse.gstreamer.org/t/unable-to-build-gstreamer-using-meson-on-windows/4454 "2025-10-25T09:02:46Z")

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I just ran meson setup buildDir in a clean gstreamer clone and got an error in meson-log.txt --- stderr --- Traceback (most recent call last): File "mesonbuild\\mesonmain.py", line 193, in run File "mesonbuild\\mesonm…

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## [Whipclientsink The SDP negotiation for simultaneously pushing audio and video streams is in chaos](https://discourse.gstreamer.org/t/whipclientsink-the-sdp-negotiation-for-simultaneously-pushing-audio-and-video-streams-is-in-chaos/5471)

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**Author:** [@mifan](https://discourse.gstreamer.org/u/mifan)\
**Replies:** 6\
**Last updated:** [October 20, 2025, 9:36am UTC](https://discourse.gstreamer.org/t/whipclientsink-the-sdp-negotiation-for-simultaneously-pushing-audio-and-video-streams-is-in-chaos/5471 "2025-10-20T09:36:17Z")

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When I use the following statement to push streams and capture packets, during SDP negotiation, sometimes the video port is 0 and sometimes the audio port is 0 gst-launch-1.0 d3d11screencapturesrc ! video/x- raw,width=1…

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## [Unable to find CUDA runtime compilation libraries](https://discourse.gstreamer.org/t/unable-to-find-cuda-runtime-compilation-libraries/5429)

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**Author:** [@Gemms](https://discourse.gstreamer.org/u/Gemms)\
**Replies:** 3\
**Last updated:** [October 16, 2025, 10:25am UTC](https://discourse.gstreamer.org/t/unable-to-find-cuda-runtime-compilation-libraries/5429 "2025-10-16T10:25:00Z")

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I have a C++ application, that previously built a pipeline using elements from the nvcodec. I had set up h264 encode/decode, and vp9 decode. I had noticed that the docs ( nvcodec ) suggested there was also a vp8dec eleme…

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## [stream video file over multicast](https://discourse.gstreamer.org/t/stream-video-file-over-multicast/5359)

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**Author:** [@bschelling](https://discourse.gstreamer.org/u/bschelling)\
**Replies:** 4\
**Last updated:** [October 10, 2025, 4:38pm UTC](https://discourse.gstreamer.org/t/stream-video-file-over-multicast/5359 "2025-10-10T16:38:00Z")

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Hi, i am using the following pipeline to stream a video over multicast / rtp: IFACE=en0 gst-launch-1.0 -v \\ filesrc location=../videofiles/video.mp4 ! \\ qtdemux name=demux demux.video\_0 ! \\ h264parse ! \\ q…

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## [Need help with synchronisation between two cameras](https://discourse.gstreamer.org/t/need-help-with-synchronisation-between-two-cameras/3426)

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**Author:** [@AdityaWasnik91](https://discourse.gstreamer.org/u/AdityaWasnik91)\
**Replies:** 3\
**Last updated:** [October 10, 2025, 4:43am UTC](https://discourse.gstreamer.org/t/need-help-with-synchronisation-between-two-cameras/3426 "2025-10-10T04:43:01Z")

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I have a pipeline that looks like rtspsrc → |-\> funnel → appsink appsrc → rtsp stream is h264 10 FPS. appsrc is reading images from a custom built camera that is running at 40 FPS. (I can not change this). \[1\] Bo…

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## [Python test failed with gstreamer 1.24.5](https://discourse.gstreamer.org/t/python-test-failed-with-gstreamer-1-24-5/3758)

<div class="topic-metadata">

**Author:** [@ravhad934](https://discourse.gstreamer.org/u/ravhad934)\
**Replies:** 5\
**Last updated:** [October 8, 2025, 2:37pm UTC](https://discourse.gstreamer.org/t/python-test-failed-with-gstreamer-1-24-5/3758 "2025-10-08T14:37:37Z")

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I am trying to build Gstreamer 1.24.5 with python using below command : meson setup . ../gstreamer --buildtype release --unity off --strip --prefix=/opt/ --libdir = /opt/lib --default-library shared --wrap-mode=nofallba…

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## [Srtclientsink latency after reconnect](https://discourse.gstreamer.org/t/srtclientsink-latency-after-reconnect/5433)

<div class="topic-metadata">

**Author:** [@isthari](https://discourse.gstreamer.org/u/isthari)\
**Replies:** 0\
**Last updated:** [October 1, 2025, 10:25pm UTC](https://discourse.gstreamer.org/t/srtclientsink-latency-after-reconnect/5433 "2025-10-01T22:25:59Z")

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I’m using mediamtx srt server for testing If I launch this pipeline, and listen on the other side, the latency is correct: gst-launch-1.0 pulsesrc ! \\ audioconvert ! \\ audioresample ! \\ opusenc bitrate=6400…

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