Questions tagged [rtp]
The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks.
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Setting SSRC in invite sdp from asterisk [closed]
We are using asterisk 1.8. Wanted to know if there is any way to generate ssrc in invite sdp from asterisk? Or is there any flag in asterisk to do this.
We are using chan_sip drivers in asterisk.
We ...
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GStreamer VC++ for converting RTP (Payload Type 96 (PT-96) ) to wav audio file
My application requirement is that I need to sniff network packets from the network card and filter RTP packet of payload type 96 (PT-96) and convert and save them in a wav audio file.
I have filtered ...
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There is a pcap file containing vp8. How do we extract the vp8 frames from the RTP and put it into playable format to be played by VLC?
There is a pcap file containing vp8. How do we extract the vp8 frames from the RTP, piece them in proper containers and put it into playable format to be played by VLC?
I have tried the following:
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How can I achieve a SIP client that works behind a NAT using PJSAU2 and Python?
I'm currently working on a SIP client using python and the python bindings for pjsua2 (pjsip) and I'm running into the issue, that calls behind NAT are able to send all SIP messages but fail in ...
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GStreamer access buffers from synchronised RTP streams
I connect to an RTSP server that produces two RTP streams (same camera), one H.264 and another uncompressed raw video. The streams are need to be used together on the client side.
On the client side, ...
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Gstreamer vp8 use of temporal layer not working
I am trying to run gstreamer vp8 rtp stream, and temporal layers need to be used do to switch video quality on sfu side. But looks like parameters not supported. Can someone help or at least point me ...
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GStreamer stream synchronised h.264 video and random data
I am designing a camera that produces H.264 video streamed over RTP. I now need to also stream raw data (just data, not video nor audio) alongside the video and be able to synchronise the video and ...
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GStreamer pipe to dynamically control alsasink
I have a GStreamer pipeline that receives an udp/rtp stream and outputs it to four soundcard channels, e.g. alsasinks:
gst-launch-1.0 udpsrc name=m_udpsrc multicast-group=239.255.255.245 auto-...
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Fixed packet size in low-latency rtp stream using queues
I am trying to stream a v4l2src source encoded as h265, with low latency mode turned on.
Below is the gstreamer command I use.
gst-launch-1.0 v4l2src device=/dev/video0 ! video/x-raw, format=NV12, ...
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Building an Virtual Camera that Receives frames with RTP
I litle interduction to my Project so i have an virtual camera that i build with Directshow, Works how t supposed to work, then i installed an external library uvgRTP and made the code so i receive ...
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How to get access to WebRTC RTP packet sequence number
In WebRTC, each RTP packet will have a sequence number in the header. (See here: https://www.geeksforgeeks.org/real-time-transport-protocol-rtp/ )
But is there a way to access this raw sequence number ...
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RTP distribution gateway in golang
I am writing a media gateway for a calling application. It doesn't need to process the RTP in any way and hence it just receives RTP (from the caller) on a UDP socket and sends to other members in the ...
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Node.js SIP.js Setup: Can't Receive Real-time Audio - Need Direction
In the development of the communication infrastructure, three key entities are involved
PSTN + Session Border Controller (SBC) alias ExtVoipGW (Freeswitch)
A Gateway (Node JS)
Node JS Server AI Bot
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Video streaming + real-time chat implementation
We would like to add a real-time chat function to an app that provides video streaming.
Video streaming communicates by modifying the RTP protocol to TCP to provide special video.
Therefore, files are ...
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SIP load Testing via SIPP
I want to test load on a oracle acme packet SBC using Sipp tool. Sipp already installed on ubuntu 16.04 server. Need sample script for load testing.do we need to edit any .xml files?
Need to send ...
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