GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2023-10-24T10:26:53Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/835rtspsrc: Not handling valid Transport headers for RTSP 2.02023-10-24T10:26:53ZSebastian Drögertspsrc: Not handling valid Transport headers for RTSP 2.0See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/129 for details.See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/129 for details.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/834rtspsrc: Not handling valid RTP-Info headers for RTSP 2.02023-10-24T10:27:04ZSebastian Drögertspsrc: Not handling valid RTP-Info headers for RTSP 2.0See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/128 for details.See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/128 for details.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1508webrtcbin: send video and audio to janus2021-01-24T08:48:42Zhei-pawebrtcbin: send video and audio to janusI need to send video and audio to a janus gateway video-room-plugin.
As an example I have used @slomo janus rust implementation and added an audio bin.
For some reason I am not able to send video and audio the same time.
Sending video ...I need to send video and audio to a janus gateway video-room-plugin.
As an example I have used @slomo janus rust implementation and added an audio bin.
For some reason I am not able to send video and audio the same time.
Sending video on it's own and audio on it's own does work.
But in the same pipeline, the video-room-plugin does not record video and audio the same time.
https://github.com/hei-pa/gst-janus-webrtc
Do I miss something?https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/84Binary release of Mac development packages have a download speed of ~10Kbps2021-01-24T10:10:12ZFabioBinary release of Mac development packages have a download speed of ~10KbpsYou can try it under this URL: https://gstreamer.freedesktop.org/data/pkg/osx/1.18.3/gstreamer-1.0-devel-1.18.3-x86_64.pkg
Versions 1.18.3 and 1.18.2 are affected. The not devel version download just fine.
With these speeds it is hard t...You can try it under this URL: https://gstreamer.freedesktop.org/data/pkg/osx/1.18.3/gstreamer-1.0-devel-1.18.3-x86_64.pkg
Versions 1.18.3 and 1.18.2 are affected. The not devel version download just fine.
With these speeds it is hard to get them to download at all.https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/833souphttpsrc: gzip content-encoding causes application/x-gzip output2021-11-24T15:21:53ZKyrylo Polezhaievsouphttpsrc: gzip content-encoding causes application/x-gzip outputApple recommends HLS playlist to be encoded with gzip.
When Content-Encoding is set to gzip, we have "application/x-gzip", not "application/vnd.apple.mpegURL". But when Content-Type is "application/vnd.apple.mpegURL", we should have "app...Apple recommends HLS playlist to be encoded with gzip.
When Content-Encoding is set to gzip, we have "application/x-gzip", not "application/vnd.apple.mpegURL". But when Content-Type is "application/vnd.apple.mpegURL", we should have "application/vnd.apple.mpegURL" output, even if Content-Encoding is gzip.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1507transcoding: handle ctrl+c to avoid output file corrupt.2021-03-04T14:32:50ZKevin Songtranscoding: handle ctrl+c to avoid output file corrupt.gst-transcoder.c need handle SIGINT to avoid output file corrupt. I meet corrupted mp4 file if input ctrl+c when transcoding. Do you have any suggestion? Should be send EOS when SIGINT?gst-transcoder.c need handle SIGINT to avoid output file corrupt. I meet corrupted mp4 file if input ctrl+c when transcoding. Do you have any suggestion? Should be send EOS when SIGINT?https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/307Beginner friendly/ good first issue?2021-01-29T08:19:56ZTejas SanapBeginner friendly/ good first issue?Hi! I couldn't see a "good-first issue" tag. I was interested in contributing. Could someone please direct me to some beginner friendly issues?Hi! I couldn't see a "good-first issue" tag. I was interested in contributing. Could someone please direct me to some beginner friendly issues?https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/860playsink: Needs to be hardware-memory-aware for deinterlacing2021-01-21T15:39:07ZSeungha Yangseungha@centricular.complaysink: Needs to be hardware-memory-aware for deinterlacing`playsink` configures `videoconvert ! deinterlace` element in any case even if configured video decoder and video sink elements are hardware elements. But there might be alternative (much faster) hardware-accelerated deinterlacing element`playsink` configures `videoconvert ! deinterlace` element in any case even if configured video decoder and video sink elements are hardware elements. But there might be alternative (much faster) hardware-accelerated deinterlacing elementhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/832souphttpsrc: unable to connect a non secure connection.2021-02-22T16:22:39ZStéphane Cerveauscerveau@igalia.comsouphttpsrc: unable to connect a non secure connection.```
gst-launch-1.0 playbin uri=https://pbbradio.com:8443/pbb128
```
Unable to connect to the stream https://pbbradio.com:8443/pbb128 with this error:
```
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: ring-buffer-max-size = 0
/Gst...```
gst-launch-1.0 playbin uri=https://pbbradio.com:8443/pbb128
```
Unable to connect to the stream https://pbbradio.com:8443/pbb128 with this error:
```
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: ring-buffer-max-size = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = -1
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: force-sw-decoders = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = https://pbb.laurentgarnier.com:8000/pbb128
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed = 0
/GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = "\(GstSoupHTTPSrc\)\ source"
0:00:00.037487493 67322 0x55a766cf2c30 WARN structure gststructure.c:2091:priv_gst_structure_append_to_gstring: No value transform to serialize field 'session' of type 'SoupSession'
Got context from element 'source': gst.soup.session=context, session=(SoupSession)NULL, force=(boolean)false;
0:00:00.169485397 67322 0x55a766cd3400 WARN souphttpsrc gstsouphttpsrc.c:1382:gst_soup_http_src_parse_status:<source> error: Secure connection setup failed.
0:00:00.169554405 67322 0x55a766cd3400 WARN souphttpsrc gstsouphttpsrc.c:1382:gst_soup_http_src_parse_status:<source> error: Error performing TLS handshake: An unexpected TLS packet was received. (6), URL: https://pbb.laurentgarnier.com:8000/pbb128, Redirect to: (NULL)
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstSoupHTTPSrc:source: Secure connection setup failed.
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/859glfilterapp recordgraphic example doesn't work2021-01-22T08:19:25ZAndrey Burmaginglfilterapp recordgraphic example doesn't workHello everyone.
I'm trying to use `glfilterapp` gl-plugin to draw on the videostream.
For this I try to run this example:
`https://github.com/GStreamer/gst-plugins-base/blob/1.18.3/tests/examples/gl/generic/recordgraphic/main.cpp`
Bu...Hello everyone.
I'm trying to use `glfilterapp` gl-plugin to draw on the videostream.
For this I try to run this example:
`https://github.com/GStreamer/gst-plugins-base/blob/1.18.3/tests/examples/gl/generic/recordgraphic/main.cpp`
But here I get a link error.
Anyway, if I get rid of the caps filtering and put only `glimagesink` after `glfilterapp` and `gltestsrc` before it, I run into black screen.
Also if I don't pass a draw-callback into `glfilterapp`, videostream flows just fine.
I can't use `glimagesink` for drawing, like here:
https://github.com/GStreamer/gst-plugins-base/blob/1.18.3/tests/examples/gl/generic/cube/main.cpp
Because I just need to draw on the video without displaying it immediately.
So, could you please explain me how to change this example in order to draw something using `glfilterapp`?
My GStreamer version is 1.16.2.
Thanks.https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/651GStreamer gst_pad_get_offset and gst_pad_set_offset not working2024-03-28T17:53:31ZPercusGStreamer gst_pad_get_offset and gst_pad_set_offset not workingHello everyone !
First of all I want to clarify that I do not speak English very well, so please apologize in advance if I do not express myself well.
**CONTEXT**
According to the command ***gst-inspect-1.0 --gst-version***, I have ve...Hello everyone !
First of all I want to clarify that I do not speak English very well, so please apologize in advance if I do not express myself well.
**CONTEXT**
According to the command ***gst-inspect-1.0 --gst-version***, I have version ***1.14.5*** of gstreamer.
I started a dynamic gstreamer project whose simplified pipeline structure looks like this *(obviously I specify that I use a programming language and not gst-launch)*:
AnElementNoMatterWhich ! videomixer ! AnElementNoMatterWhich ! tee ! queue ! AnElementNoMatterWhich
Now imagine that after *10 seconds* I want to add an element, *no matter witch*, to the ***videomixer*** sink. Or let's say I want to add an item to the ***tee*** src (this is the same as with the *videomixer*, except that I would add a tail between the tee and the element so as *not to block* other branches).
It will take about 10 seconds before the stream passes into the sink pad of the videomixer . Likewise for the src pad of the tee. By analogy, the time spent before adding an element in a pipeline (and connecting it to the videomixer or to the tee) is around the same time that the videomixer and/or tee pads must wait before passing the stream.
This is all explained by the **difference** in execution of the pipeline compared to the newly created videomixer and / or tee pads.
This is where the two methods ***gst_pad_get_offset()*** and ***gst_pad_set_offset()*** come in.
**MY PROBLEMS**
- First of all I can not understand what the ***offset unit (gint64)*** is. Is it a unit of time ? Millisecond ? Microsecond ? Nanosecond ? Or is it something else ?
- Also I can't understand why ***gst_pad_get_offset()*** always returns **0**. However, I have to get the offset if I want to be able to set the right offset with the ***gst_pad_set_offset()*** method. According to the documentation I have to do it on a ***src pad***, but where ?
- And finally I don't understand why ***gst_pad_set_offset()*** is not working. I obviously run it on the sink pad of the newly created videomixer or on the src pad of the tee *(whatever)*. It doesn't change anything. Besides, if I have the idea of giving the value 10 (for example) to gst_pad_set_offset () and that I then run gst_pad_get_offset () on the same pad, I should theoretically have the value 10. However, I get always 0. It is as if gst_pad_get_offset () and gst_pad_set_offset () had not been implemented.
*I followed all the gstreamer tutorials and spent several days on the forums. I do not know what to do. However, I know that without the proper functioning of these two methods, my project is blocked. What can I do ? Do you have an idea ?*https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1506msdk: msdkh265enc encoder is failing, Video Encode Query failed (undeveloped...2021-03-12T05:49:45ZNaveen Kumar Sainimsdk: msdkh265enc encoder is failing, Video Encode Query failed (undeveloped feature)OS: Ubuntu 18.04
Gst-version: 1.18.3.1[gst_1.18_msdkh265enc_failure.log](/uploads/c9f32cd689d46eaba82ad1e39176748e/gst_1.18_msdkh265enc_failure.log)
Installed using gst-build [https://gitlab.freedesktop.org/gstreamer/gst-build/-/tree/1....OS: Ubuntu 18.04
Gst-version: 1.18.3.1[gst_1.18_msdkh265enc_failure.log](/uploads/c9f32cd689d46eaba82ad1e39176748e/gst_1.18_msdkh265enc_failure.log)
Installed using gst-build [https://gitlab.freedesktop.org/gstreamer/gst-build/-/tree/1.18]
Encoding command:
$ gst-launch-1.0 -ev videotestsrc num-buffers=120 ! timeoverlay ! msdkh265enc ! video/x-h265,profile=main ! h265parse ! filesink location=./sample.h265
libva info: VA-API version 1.11.0
libva info: Trying to open /usr/local/lib/dri/iHD_drv_video.so
libva info: Found init function __vaDriverInit_1_11
libva info: va_openDriver() returns 0
Setting pipeline to PAUSED ...
libva info: VA-API version 1.11.0
libva info: Trying to open /usr/local/lib/dri/iHD_drv_video.so
libva info: Found init function __vaDriverInit_1_11
libva info: va_openDriver() returns 0
Pipeline is PREROLLING ...
Got context from element 'msdkh265enc0': gst.msdk.Context=context, gst.msdk.Context=(GstMsdkContext)"\(GstMsdkContext\)\ msdkcontext1";
/GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0.GstPad:src: caps = video/x-raw, format=(string)Y410, width=(int)320, height=(int)240, framerate=(fraction)30/1, multiview-mode=(string)mono, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1
/GstPipeline:pipeline0/GstTimeOverlay:timeoverlay0.GstPad:src: caps = video/x-raw, format=(string)Y410, width=(int)320, height=(int)240, framerate=(fraction)30/1, multiview-mode=(string)mono, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1
0:00:00.033465534 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undeveloped feature)
0:00:00.033719838 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undefined behavior)
/GstPipeline:pipeline0/GstTimeOverlay:timeoverlay0.GstPad:video_sink: caps = video/x-raw, format=(string)Y410, width=(int)320, height=(int)240, framerate=(fraction)30/1, multiview-mode=(string)mono, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1
0:00:00.035234069 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undefined behavior)
0:00:00.043979433 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undefined behavior)
ERROR: from element /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0: Internal data stream error.
Additional debug info:
../subprojects/gstreamer/libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0:
streaming stopped, reason not-negotiated (-4)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
0:00:00.044251473 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undefined behavior)
0:00:00.044573326 13779 0x55fc40c9a0a0 ERROR msdkenc gstmsdkenc.c:613:gst_msdkenc_init_encoder:<msdkh265enc0> Video Encode Query failed (undefined behavior)
Freeing pipeline ...https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/831rtspsrc: Does not handle "Unsupported Transport" correctly with RTSP 2.0 and ...2023-10-24T10:27:24ZSebastian Drögertspsrc: Does not handle "Unsupported Transport" correctly with RTSP 2.0 and pipelined requestsProblem here is that we'd send a SETUP for each stream at once, and then they could all come back with "Unsupported Transport". For RTSP 1.0 we would try other transports then, for RTSP 2.0 this completely fails instead of trying other t...Problem here is that we'd send a SETUP for each stream at once, and then they could all come back with "Unsupported Transport". For RTSP 1.0 we would try other transports then, for RTSP 2.0 this completely fails instead of trying other transports.
In addition, for RTSP 2.0 (and 1.0 actually!) we could provide the list of all transports we would support in the initial SETUP and let the server select something instead of probing via "Unsupported Transport" and trying multiple times.
CC @thiblahutehttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1505Copy paste bug in gstav1parser.c2023-05-30T16:37:17ZRafał MikrutCopy paste bug in gstav1parser.chttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/codecparsers/gstav1parser.c#L3386-3389
```
frame_header->ref_frame_idx[GST_AV1_REF_LAST_FRAME -
GST_AV1_REF_LAST_FRAME] = frame_header->last_fram...https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/codecparsers/gstav1parser.c#L3386-3389
```
frame_header->ref_frame_idx[GST_AV1_REF_LAST_FRAME -
GST_AV1_REF_LAST_FRAME] = frame_header->last_frame_idx;
frame_header->ref_frame_idx[GST_AV1_REF_GOLDEN_FRAME -
GST_AV1_REF_LAST_FRAME] = frame_header->gold_frame_idx;
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1504Copy paste bug in gstvkutils.c2021-01-22T13:03:43ZRafał MikrutCopy paste bug in gstvkutils.chttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/vulkan/gstvkutils.c#L448-451
```
&& view->create_info.subresourceRange.levelCount ==
info->subresourceRange.levelCount
&& view->create_...https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/vulkan/gstvkutils.c#L448-451
```
&& view->create_info.subresourceRange.levelCount ==
info->subresourceRange.levelCount
&& view->create_info.subresourceRange.levelCount ==
info->subresourceRange.levelCount
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1503Copy paste bug in sctpassociation.c2023-05-31T17:29:33ZRafał MikrutCopy paste bug in sctpassociation.chttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/ext/sctp/sctpassociation.c#L926-927
```c
if (!(sr->strreset_flags & SCTP_STREAM_RESET_DENIED) &&
!(sr->strreset_flags & SCTP_STREAM_RESET_DENIED)) {
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/ext/sctp/sctpassociation.c#L926-927
```c
if (!(sr->strreset_flags & SCTP_STREAM_RESET_DENIED) &&
!(sr->strreset_flags & SCTP_STREAM_RESET_DENIED)) {
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1502Copy paste bug in gstav1parse.c2021-01-23T11:18:24ZRafał MikrutCopy paste bug in gstav1parse.cFirst and latest condition are identical
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst/videoparsers/gstvp9parse.c#L580-591
```c
if (self->subsampling_x == 1 && self->subsampling_y == 1)
chroma_forma...First and latest condition are identical
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst/videoparsers/gstvp9parse.c#L580-591
```c
if (self->subsampling_x == 1 && self->subsampling_y == 1)
chroma_format = "4:2:0";
else if (self->subsampling_x == 1 && self->subsampling_y == 0)
chroma_format = "4:2:2";
else if (self->subsampling_x == 0 && self->subsampling_y == 1)
chroma_format = "4:4:0";
else if (self->subsampling_x == 1 && self->subsampling_y == 1)
chroma_format = "4:4:4";
```
and also
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst/videoparsers/gstav1parse.c#L567-574
```c
if (self->parser->subsampling_x == 1 && self->parser->subsampling_y == 1)
chroma_format = "4:2:0";
else if (self->parser->subsampling_x == 1 &&
self->parser->subsampling_y == 0)
chroma_format = "4:2:2";
else if (self->parser->subsampling_x == 0 &&
self->parser->subsampling_y == 1)
chroma_format = "4:4:0";
else if (self->parser->subsampling_x == 1 &&
self->parser->subsampling_y == 1)
chroma_format = "4:4:4";
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1501Copy paste bug in gstvafilter.c2021-01-22T09:32:00ZRafał MikrutCopy paste bug in gstvafilter.chttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/sys/va/gstvafilter.c#L150-153
```
self->min_height = 1;
self->max_height = G_MAXINT;
self->min_width = 1;
self->max_height = G_MAXINT;
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/sys/va/gstvafilter.c#L150-153
```
self->min_height = 1;
self->max_height = G_MAXINT;
self->min_width = 1;
self->max_height = G_MAXINT;
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1500Copy paste bug in gstav1parser.c2021-01-23T11:18:23ZRafał MikrutCopy paste bug in gstav1parser.chttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/codecparsers/gstav1parser.c#L2605-2607
```
lr_params->frame_restoration_type[0] = GST_AV1_FRAME_RESTORE_NONE;
lr_params->frame_restoration_type[...https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/gst-libs/gst/codecparsers/gstav1parser.c#L2605-2607
```
lr_params->frame_restoration_type[0] = GST_AV1_FRAME_RESTORE_NONE;
lr_params->frame_restoration_type[0] = GST_AV1_FRAME_RESTORE_NONE;
lr_params->frame_restoration_type[0] = GST_AV1_FRAME_RESTORE_NONE;
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/830gstudpnetutils.c:69:gst_udp_parse_uri: parse IPV4 address '::1:50001'2022-09-27T19:48:00ZLeo Gaspardgstudpnetutils.c:69:gst_udp_parse_uri: parse IPV4 address '::1:50001'Hello,
I was trying to open a RTSP stream from `rtsp://[user]:[pass]@localhost:8554` just today, and found out that gstreamer resolves `localhost` as `::1` and then fails in `gst_udp_parse_uri` because it assumes that IPv6 addresses are...Hello,
I was trying to open a RTSP stream from `rtsp://[user]:[pass]@localhost:8554` just today, and found out that gstreamer resolves `localhost` as `::1` and then fails in `gst_udp_parse_uri` because it assumes that IPv6 addresses are wrapped in `[]` there.
That being said, as I don't have a development environment set up for gstreamer (yet), I couldn't investigate more to figure out where that hostname resolution happens to add the brackets at the right place… sorry about it!
As a workaround, just hardcoding the IPv4 address of the RTSP server works.
Hope that helps!
Leo