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Questions tagged [rtcp]

**RTP Control Protocol** (RTCP) is a helper protocol for *Real-time Transport Protocol* (RTP).

It is used to gather statistics about RTP sessions and to control the RTP flow. It does not transport multimedia data itself.

Example control message (which names show RTCP purpose): Sender Report (SR), Receiver Report (RR), Source Description (SDES), End of participation (BYE).

Detailed documentation can be found in part of RFC 3550 (RTP: A Transport Protocol for Real-Time Applications) which describes RTCP.

More information at http://en.wikipedia.org/wiki/RTP_Control_Protocol

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CSRC and SSRC in RTP

I am very new to RTP , can some one explain about the CSRC and SSRC in general? From http://www.rfc-editor.org/rfc/rfc3550.txt , what it says is : The SSRC field identifies the synchronization source. Does it mean , in network there may be many…
badri.coder
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How do I determine if a packet is RTP/RTCP?

I am using SharpPCap which is built on WinPCap to capture UDP traffic. My end goal is to capture the audio data from H.323 and save those phone conversations as WAV files. But first thing is first - I need to figure out what my UDP packets are…
Chris Holmes
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RTCP / RTP communication issue

Unfortunately I'm still stuck with a little implementation of a RTP / RTCP communication to access my IP camera properly. What do I want to do The camera has an internal buffer which I want to read. So I communicate with the camera via RTSP and tell…
Toby
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WebRTC: RTCDataChannel is not 'open'

I followed an entire tutorial about WebRTC and implementing a simple p2p chat. My signaling server is working aside on localhost:9090. When I try to send a message, I am receiving: RTCDataChannel.readyState is not 'open' However, the connection…
Arkon
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How to stream pcap file to RTP/RTCP stream?

I have captured three different stream as pcap file with meta datas. How can I stream back to RTP/RTCP stream?
Swaminathan
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WebRTC 'goog-remb' and 'transport-cc' SDP lines

I would like to know what are the meaning of this SDP lines as I am trying to get the smoothest framerate posible with 5% to 10% packet losses. The lines i don´t know are: a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc I don´t know why firefox…
user1428926
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RTSP, RTCP and RTP port numbers

I'm trying to understand how spydroid (https://code.google.com/p/spydroid-ipcamera/) works so I can write a similar app for my phone. Based off of spydroid here is what I understand about RTSP, RTCP and RTP. RTSP runs on a specified port and sets…
Ricky Casavecchia
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How to fix incorrect timestamp calculations? [OpenRtspClient]

My RTSP Source's RTCP SR are not reliable for some of the calculated timestamps of H.264 streams frequently resulting in large negative jumps. Here is an example from the debug log. See how it jumps from 101006.6130 to -4193861.6830 and continues…
Novalis
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Receiving RTCP issues within oRTP library

I am currently getting a segfault when trying to receive RTCP in RECVONLY sessions in oRTP. RTCP works fine when receiving in a SENDRECV session but won't work in RECVONLY sessions. This is my code. The segfault comes from the function on line 42:…
Kyle Dijkstra
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Java RTP/RTCP library using NIO

Is there a Java RTP/RTCP library based on Java NIO or some Java NIO framework (Netty, MINA, ...)?
user17481
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How to make a streaming relay using gstreamer?

I would like to make of some sort of a streaming server. I would like it to receive RTSP streams over the net from live streams (e.g. webcam, ipcam, etc.) then broadcast that same stream on my local network using a different URL. I know gstreamer…
Bien Baldonado
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How to keep RTSP session alive?

I tried streaming on Google nexus S (2.3.7), HTC Desire (2.3.3), and Samsung Galaxy (3.2). And only Google Nexus has RTSP session timeout issue. I read up on some threads about this problem. It seems like I'll have to send RTCP request every second…
user766409
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How to change Webrtc rtcpMuxPolicy?

I am trying to change rtcpMuxPolicy and also the bundlePolicy, but it seems like it cannot be changed This is my code: Attempt 1: var servers = { 'iceServers': [{ 'urls': 'stun-url..' }, { 'urls': 'stun-url-2..' }, { …
ToTa
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GStreamer: Calculate delay in received video frames/buffers to detect communication delay between Tx and Rx

I am looking into an application that requires to detect the delay in receiving video frames and then takes action if a delay is detected. The delay in receiving video frames is perceived as a video freeze on the render window. The action is…
gst
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How to print RTP and RTCP packet messages using JsSIP?

I am using JsSIP to make SIP calls. I am able to see Signalling messages like From, To, Via in the console. In the same way, can we see RTP and RTCP packet transmission messages?
Cherry
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