Questions tagged [rtp]

The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks.

RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.

More information at Wikipedia Page on RTP

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How to intercept packets from RTP-stream on loopback and access the data through C++-code?

I want to be able to intercept the packets of this stream and access the data from C++-code. How do I do this in C++-code? The RTP- media stream are streamed using this server: link I will then FEC-encode the packets; send them over the network;…
Anders Branderud
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Converting AAC into LPCM via AudioConverterFillComplexBuffer

I'm trying to make use of the 'Audio Converter Service' - AudioConverterFillComplexBuffer I'm hitting quite a few issues. Firstly I don't know how to populate the ioOutputDataPacketSize. I've parsed my SDP descriptor, and I'm getting 8000Hz, 1…
Peter
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Trouble packetizing h264 stream in FU-A mode over RTP UDP

I am having some trouble packetizing an h.264 stream using SIP and RTP over UDP. I am using the FU-A packetization mode for a video avc track in an mp4. I spent time to get the fragmentation correct, with the duration timestamp on the RTP packet, as…
diesel10
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RTP payload and NAL units

TL;DR: Android platform; MediaRecorder, using H264 and 3gp; MediaRecorder stream to localSocketServer; Reading the stream; Skipping the first MPEG header (40 bytes); Searching for NAL units, but nothing As it is in here…
Dummas
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encoding/decoding H.264 streams with different parameters

When using H.264 in a point to point session (bidirectional) some kind of media negotiation has to be performed. But, could the two streams have different characteristics, such as different profiles/levels etc.? After all, encoder and decoder are…
Bob
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On the bounding sets in RFC 5104

In section 3.5.4.2 of RFC5104, an algorithm to derive the bounding set of a set of lines is derived. Basically each line is of the form y=mx+b, and the objective is to find the points of intersections that identify the convex hull (equivalent to…
Bob
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Streaming using GStreamer

I have got one HD video "ed_hd.avi" on System#1. Would like to stream it over network and play the content from System#2. I am using GStreamer on Ubuntu 11.04, tried a lot on this. Variety of errors makes this objective difficult to diagnose. Will…
Sumen
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How does SIP/RTP determine two endpoints are on the same LAN?

I am just experimenting with my phone system and I'm wondering how both endpoints know they are on the same LAN, I have both endpoints breaking out to the cloud phone system with two separate public IP addresses, I've segmented them off from each…
Adam
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Corrupted H264 video when streaming via RTP/UDP

I'm trying to stream a video encoded in H264 over RTP/UDP. Sending: gst-launch-1.0 \ videotestsrc ! \ video/x-raw,format=RGBx,width=960,height=540,framerate=25/1 ! \ videoconvert ! \ x264enc bitrate=2000 ! \ rtph264pay config-interval=1…
Dmitrii
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Audio DVR (DAR) software

I have been looking for an open source live audio recording project for recording raw audio over IP. It seems I can only find a handful of video DVR projects, but no specific audio DAR projects. I have been told to take a look at the DVR projects…
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Linking error in test program with RTP (Real-Time Transport Protocol)

I am trying to compile a simple test program using rtp (Real-Time Transport Protocol). Unfortunately I get a linking error I can not get rid off. Here is the Code i try to compile with: g++ -v main.cpp -o rtp -lccrtp Here is the main.cpp #include…
MWende
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Video Stream Analysis metric

Im just new to Video Streaming, and into some video stream deployment. I'm wondering what's the best free analyzer to test RTP streams which can give me bandwidth, jitter etc. Also what the best metric's to look for when monitoring these streams I…
user726720
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I am trying to decode a compressed rtp packet to evs and make it into a wav file

I am trying to decode a compressed rtp packet to evs and make it into a wav file. I use C language in Redhat 6.8 64bit environment. I have rtp packet dump ( evs ) I used EVS_dec in 3GPP TS 26.443 V15.1.0. C source code. rtp packet -> g.192 format…
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How to solve Jiiter Buffer problem in receiving audio RTP stream (bad sound quality) in PJSIP?

I'm a newbie to pjsip and want to build an RTP stream receiver using pjsip. Setup: I want to use specific L16/16000/1 codec and have also enabled it in "config_site.h" during compiling the pjsip project and checked that its…
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Is it possible to implement RTP-based protocol using Google's FlatBuffers library?

I'm especially interested in implementation of RTP-MIDI protocol. The major difficulties could be faced in implementation of bit-fields and non-ordinary MIDI-like timestamps, as I can expect. And maybe if somebody knows already existed open source…