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

75 questions
0
votes
0 answers

Freeswitch sending empty RTP in time with receiving RTCP

FreeSWITCH Version: 1.10.3 I was hoping that someone maybe able to help me. Intermittently on inbound calls from the our sip provider we are sending a empty RTP packet back to the provider. After doing some investigation, it happens when FreeSWITCH…
aturner
  • 23
  • 3
0
votes
0 answers

Live555 Streaming Server RTP Lost Packet Retransmit

Does the live555 streaming server retransmit lost RTP packets? When I search for "live555 retransmit" it turns up a few results: In 2011 Francesco Tonelli described a mechanism to implement retransmission of lost packets. This means there was no…
minghua
  • 5,981
  • 6
  • 45
  • 71
0
votes
2 answers

FFmpeg - RTCP BYE packets

I’m working on some C++ project which depends on Wi-Fi RAK5206 electronic board. I’m using ffmpeg library to obtain video and audio stream and I have issue where I can start and stop stream for four times, but when I want to start for the fifth time…
DZoki019
  • 382
  • 2
  • 13
0
votes
1 answer

[rtp/rtcp server]How to prepare a stored media file for steaming?

Now i'm trying to understand the rtp/rtcp protocol(RFC3550). I knew that in common case,the audio and video steaming is separately. But if i want to steaming a stored media file(such as *.mp4) in the server, how does the server get those tracks from…
zbw
  • 13
  • 4
0
votes
1 answer

Where to start learning about basics about sending rtp packets for audio video

Can someone provide training or documentation about how actually video flows between two sip clients. I know basics of where client apis but don’t have much knowledge on how rtp packets are formatted , send over wire and received by other client…
0
votes
0 answers

How to extract RTCP information in a real-time stream using JRTPLIB? C++

By using jrtplib, I've written both server and client side for RTP h264 video streaming and adapted them for the client can use RTSP to have control over the quality of the video that is streamed by the server. Now I am asked to improve these…
0
votes
1 answer

Does it make sense to use RTP protocol for multiple streamers and single receiver?

I am in a process of learning and trying to use the RTP/RTCP protocol. My situation is that there is 1 to n streamers and 1 (or potentially 1 to m if needed) receiver(s), but in a way that the streamers themselves do not know about each other (they…
user5125238
0
votes
1 answer

Recording RTP stream and subsequent playback

I am looking for some open source library written in C that dumps RTP packets (RTCP packets are available too) in a file so that that file can be then played. For simplicity, to get a hang of the issues, I want to just record audio (e.g. opus…
Sunny
  • 9,245
  • 10
  • 49
  • 79
0
votes
1 answer

RTP sequence number rounding up after reaching max value

Sequence number in RTP packet is a 16 bit field. It starts with a random value. Lets say I keep sending RTP packet, at some point of time it will reach max value 0xFFFF. Then I need to set the sequence number to smaller value. That time, should I…
Austin
  • 1,709
  • 20
  • 40
0
votes
0 answers

janus or pjsip for rtp-rtcp implementation

I am evaluating implementation of a simple SFU for webRTC. I have looked cursorily at the two source codes but find that support for REMB and PLI is missing in pjsip. How come it works with WebRTC then? I just want a simple gateway so I can forward…
asinix
  • 966
  • 1
  • 9
  • 22
0
votes
1 answer

MQTT streaming QoS control

Probably relating to the SO question MQTT for realtime data streaming, how would the realtime multimedia quality be achieved with MQTT? This is different from the MQTT defined QoS 0, 1, or 2. In realtime streaming with RTP and RTCP, these extra…
minghua
  • 5,981
  • 6
  • 45
  • 71
0
votes
1 answer

Why RTSP over TCP does not work well when RTT is large

I'm using live555 to stream H264 video from On-demand RTSP server using RTSP over TCP as follows: ./openRTSP -F tcp-test -Q -d 60 -b 500000 -4 -P 62 -w 3840 -h 2160 -f 30 rtsp://ip-address:8554/h264ESVideoTest It works well when RTT is less than…
earslan
  • 1
  • 3
0
votes
0 answers

Receive real-time rtp/rtcp video stream on windows

I'm trying to send and receive video stream with delay as low as possible. For sending I use gstreamer on linux: gst-launch-1.0 -v rtpbin ntp-sync=false name=rtpbin \ v4l2src do-timestamp=true ! jpegenc ! rtpjpegpay ! rtpbin.send_rtp_sink_0 \ …
Amomum
  • 6,217
  • 8
  • 34
  • 62
0
votes
1 answer

Duplicated Source Identifier at RTP streams. Can mess up RTCP reporting?

First of all, this is somenting like RTP: SSRC collision detection in unicast sessions, but the question is on other scope. The scenario: I have a bunch of media agents communicating with a central media gateway. An arrangement like {DeviceA}…
wesley.mesquita
  • 795
  • 5
  • 12
0
votes
1 answer

Understand tcpdump output for RTCP RR and SR

Can somebody explain the SR/RR parts of this tcpdump output? Example: tcpdump -n udp -x port 5091 and less 129 -T rtcp 16:58:15.034159 IP 1.2.3.4.5091 > 10.2.3.4.45041: sr @3665059093.56 3025985984 1003p 160480b 3l 1012s 12j @23811.54+1.80 sdes…
Andy Thompson
  • 284
  • 1
  • 3
  • 21