rtspclient

by 7956968 C++ Version: Current License: GPL-3.0

X-Ray Key Features Code Snippets Community Discussions(10)Vulnerabilities Install Support

kandi X-RAY | rtspclient Summary

rtspclient is a C++ library. rtspclient has no bugs, it has no vulnerabilities, it has a Strong Copyleft License and it has low support. You can download it from GitHub.

rtspclient

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Quality

Security

License

Reuse

Support

rtspclient has a low active ecosystem.

It has 5 star(s) with 6 fork(s). There are 5 watchers for this library.

It had no major release in the last 6 months.

rtspclient has no issues reported. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of rtspclient is current.

Quality

rtspclient has no bugs reported.

Security

rtspclient has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

License

rtspclient is licensed under the GPL-3.0 License. This license is Strong Copyleft.

Strong Copyleft licenses enforce sharing, and you can use them when creating open source projects.

Reuse

rtspclient releases are not available. You will need to build from source code and install.

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rtspclient Key Features

No Key Features are available at this moment for rtspclient.

rtspclient Examples and Code Snippets

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Community Discussions

Trending Discussions on rtspclient

Is there a way for Rust to use the default associated types?

Send jpeg images (motion jpeg) through RTSP - gStreamer

Receiving multicast RTP stream (containing multiple subsessions) from a recorded RTSP session (pcap) using Live555

How to use dynamic dispatch in Rust for trait with generics?

How to change gstreamer omxh264enc profile?

VLC doenst work on 1 of 3 Pipelines with same encoding and depay from gst-rtsp-server

Type of lambda function that receives an argument

Live555 RTSP server does not use UDP

How to stream MIC input by RTSP with gstreamer

A/libc: Fatal signal 11 (SIGSEGV) at 0x00000005 (code=1), thread 26834 Android app crash

QUESTION

Is there a way for Rust to use the default associated types?

Asked 2021-May-04 at 01:14

Following https://stackoverflow.com/a/65924807/5884503 I'm adding associated types so I can return futures from my trait methods:

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ANSWER

Answered 2021-May-04 at 01:14

There are multiple misunderstandings here. The Future<...> part of the associated type is not a default it is a constraint. Future is not a type, its a trait. Its saying an implementation's PlayResult, for example, must satisfy the Future<...> constraint.

A default associated type would look like this:

Source https://stackoverflow.com/questions/67375914

QUESTION

Send jpeg images (motion jpeg) through RTSP - gStreamer

Asked 2021-Feb-13 at 11:48

Hi I am trying to build a video streaming pipeline using gstreamer and I have a hard time making it work. I have a camera that supports MJPG so I want to pass jpeg image to jpegparse and convert to rtp with rtpjpegpay.

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ANSWER

Answered 2021-Feb-13 at 11:46

There you go. You missed my comment regarding space.. You need to use spaces after ( and before )

Source https://stackoverflow.com/questions/66166929

QUESTION

Receiving multicast RTP stream (containing multiple subsessions) from a recorded RTSP session (pcap) using Live555

Asked 2020-Nov-27 at 16:12

I have to implement an RTSP Client which connects to an existing RTSP session without being able to send commands to the RTSP Server (recvonly).

To simulate such an environment, I have recorded a RTSP/RTP stream between testH264VideoStreamer and testRTSPClient examples from Live555 with Wireshark, and played it back using tcpreplay while trying to receive stream data with a modified version of testRTSPClient.

I've also stored the SDP information provided by the testH264VideoStreamer as an SDP file.

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ANSWER

Answered 2020-Nov-27 at 16:12

I've found out that although wireshark was showing me the incoming packets with valid checksums, udp port received no packets.

I've tried following commands (as sudo) to avoid kernel discarding the packets but they simply don't help on Debian Buster.

Source https://stackoverflow.com/questions/65009916

QUESTION

How to use dynamic dispatch in Rust for trait with generics?

Asked 2020-Sep-23 at 01:54

I'm using a trait called RtspClient so I can create different Stream objects with different rtsp clients:

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ANSWER

Answered 2020-Sep-23 at 01:54

You can change the function's signature to accept:

A reference to a dyn, &dyn Fn(EncodedPacket),
A boxed dyn, Box, useful for storing instead of (or in addition to) using immediately
A function pointer fn(EncodedPacket), which does not allow keeping track of state

See Advanced Functions and Closures.

You can also move the method to a different trait, and even have the other trait automatically implement the trait usable with dyn.

Playground demo

Source https://stackoverflow.com/questions/64019215

QUESTION

How to change gstreamer omxh264enc profile?

Asked 2019-Nov-26 at 11:02

I am trying to convert a webcam on a raspberry pi to x264, but keep running into an error about an " Unsupported profile constrained-baseline".

...

ANSWER

Answered 2019-Nov-26 at 11:02

You can change that through the SRC cap, try e.g.:

Source https://stackoverflow.com/questions/58780795

QUESTION

VLC doenst work on 1 of 3 Pipelines with same encoding and depay from gst-rtsp-server

Asked 2019-Nov-09 at 11:44

Hey Guys

I have a strange thing here, I have 3 diffrent Media Fatory's added to my RTSP Server, One of that (third one below) I can only play with gst-launch on a diffrent machine.

The first two below works fine on vlc and gst-launch, what I didnt understand is why third one doesnt work, all pipelines has only diffrent sources, encoding and depay is same on all 3 pipelines

Work on other machine with vlc and gst-launch

compositor name=mix sink_0::xpos=0 sink_0::ypos=0 sink_1::xpos=640 sink_1::ypos=0 sink_2::xpos=0 sink_2::ypos=480 sink_3::xpos=640 sink_3::ypos=480 ! queue max-size-time=0 ! videoconvert ! videoscale ! video/x-raw, width=1920, height=1080 ! x264enc key-int-max=5 speed-preset=ultrafast tune=zerolatency ! rtph264pay name=pay0 filesrc location=error.jpg ! jpegdec ! imagefreeze ! textoverlay text="K1 Camera not reachable, check camera or IP-Address or Hostname" valignment=top font-desc="Arial, 13" ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. filesrc location=error.jpg ! jpegdec ! imagefreeze ! textoverlay text="K2 Camera not reachable, check camera or IP-Address or Hostname" valignment=top font-desc="Arial, 13" ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix.

Work on other machine with VLC and gst-launch

compositor name=mix sink_0::xpos=0 sink_0::ypos=0 sink_2::xpos=0 sink_2::ypos=480 sink_4::xpos=0 sink_4::ypos=960 ! queue max-size-time=0 ! videoconvert ! videoscale ! video/x-raw, width=1920, height=480 ! x264enc key-int-max=5 speed-preset=ultrafast tune=zerolatency ! rtph264pay name=pay0 souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///mjpg/video.mjpg" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///mjpg/video.mjpg" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///mjpg/video.mjpg" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix.

Work on other machine only with gst-launch

compositor name=mix sink_0::xpos=0 sink_0::ypos=0 sink_1::xpos=640 sink_1::ypos=0 sink_2::xpos=1280 sink_2::ypos=0 sink_2::xpos=0 sink_2::ypos=480 sink_3::xpos=640 sink_3::ypos=480 sink_4::xpos=1280 sink_4::ypos=480 ! queue max-size-time=0 ! videoconvert ! videoscale ! video/x-raw, width=1920, height=960 ! x264enc key-int-max=5 speed-preset=ultrafast tune=zerolatency ! rtph264pay name=pay0 rtspsrc latency=200 user-id="admin" user-pw="enercon525244" location="rtsp:///Streaming/channels/102" ! rtph264depay ! h264parse ! avdec_h264 ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///control/faststream.jpg?stream=full&preview&camera=left&size=640x480&fps=4.0" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///control/faststream.jpg?stream=full&preview&camera=right&size=640x480&fps=4.0" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///control/faststream.jpg?stream=full&preview&camera=left&size=640x480&fps=4.0" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix. souphttpsrc ssl-strict=false is-live=true user-id="" user-pw="" location="http:///control/faststream.jpg?stream=full&preview&camera=right&size=640x480&fps=4.0" do-timestamp=true ! multipartdemux ! jpegdec ! videoscale ! video/x-raw, width=640, height=480 ! queue ! mix.

With GST_DEBUG=1 on my gst-rtsp-server instance it returns that error on the last Pipeline when I tried it with VLC:

...

ANSWER

Answered 2019-Nov-09 at 11:15

First of all you should check the VLC log what it doesn't like. Since your last one has more inputs I can imagine that your compositor has a different configuration.

Out of the blue I would guess that in your last one the compositor outputs 4:4:4 data and creates a 4:4:4 profile H.264 stream. These streams are rarely supported by players.

If that's the case you should make sure a 4:2:0 format goes into x264enc or define a non-4:4:4 H.264 target profile on the output caps.

Source https://stackoverflow.com/questions/58778620

QUESTION

Type of lambda function that receives an argument

Asked 2019-Sep-13 at 09:22

I'm trying to receive lambda functions as parameters, however I'm having problem with its types.

Here's how I'm calling the function

...

ANSWER

Answered 2019-Sep-13 at 08:20

lambda is not std::function, so T cannot be deduced.

You might get rid of std::function and take functor:

Source https://stackoverflow.com/questions/57917993

QUESTION

Live555 RTSP server does not use UDP

Asked 2019-Feb-20 at 20:23

I have a pretty basic live555 RTSP server and client to stream a h264 stream written in c++.

Here's the code I have for the client (adapted from testProgs/testRTSPClient.cpp, bundled with live555)

...

ANSWER

Answered 2019-Feb-20 at 20:23

Okay so I figured out the answer. To help anyone else who is curious about this, the code is actually all correct. There is also no mis-interpretation of netstat. RTSP does indeed run over TCP not UDP. However the transport method of the A/V data runs on RTP, a connection that RTSP simply negotiates and instantiates. RTP almost always will run over UDP. To figure out what port and protocol the A/V data stream is going over you will need to sniff the packets sent out via RTSP. In my case the A/V data stream was indeed still going over UDP.

Source https://stackoverflow.com/questions/54774511

QUESTION

How to stream MIC input by RTSP with gstreamer

Asked 2018-Jan-23 at 12:30

I would like to stream mic input from RTSP server. I use gstreamer1.0 and gstreamer1.0-rtsp-server (v1.12.3).

I tried the following commands, but RTSP server created by the pipeline is not responsed. How can I stream it?

...

ANSWER

Answered 2018-Jan-23 at 12:30

you need to set the name for rtpL16pay, try the following pipeline for TX:

For testing initially start with audiotestsrc:

test-launch "(audiotestsrc ! audioconvert ! rtpL16pay name=pay0 )"

and from your mic(if its connected at 0 use the following pipeline):

test-launch "(alsasrc device="hw:0" ! audioparse ! decodebin ! audioconvert ! audioresample ! avenc_g722 ! rtpg722pay name=pay0 )"

And for RX try the following pipeline:

gst-launch-1.0 -v -e rtspsrc location=rtsp://127.0.0.1:8554/test ! rtpjitterbuffer latency=100 ! rtpL16depay ! audioconvert ! alsasink

Source https://stackoverflow.com/questions/48394817

QUESTION

A/libc: Fatal signal 11 (SIGSEGV) at 0x00000005 (code=1), thread 26834 Android app crash

Asked 2017-Apr-21 at 06:46

I checked this and this links for this problem but couldn't find any solution.

I am implementing an RTSP player in android and I have used Easy Player for this. Using this player I have implementd my demo application and entered an rtsp:// URL.

PROBLEM: When I enter the same URL in library sample code it is working fine, but when I try with my demo app the app crashes with following stacktrace.

...

ANSWER

Answered 2017-Apr-21 at 06:46

I found my mistake, in Easy player, a unique key is generated for every package name and in my case that key was incorrect. I don't know why it is giving memory leakage error for an invalid key!! Replacing package name and key solved this problem. Strange but true.

Source https://stackoverflow.com/questions/43518102

Community Discussions, Code Snippets contain sources that include Stack Exchange Network

Vulnerabilities

No vulnerabilities reported

Install rtspclient

You can download it from GitHub.

Support

For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .

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