Stack Overflow
Stack Overflow

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
1
vote
0 answers

Convert Jitter from RTP timestamp unit to millisseconds

I have a video conference app and I want to display the Interarrival Jitter to the user. I am getting this information from FFmpeg, and it follows the RFC 3550 Appendix A.8, so the information is in timestamp units. I am not sure how to convert it.…
Vitor Vanacor
  • 407
  • 4
  • 6
1
vote
1 answer

How do I relate WebRTC's RTCP packets to the respective RTP streams?

How do RTP packets relate to RTCP? I expected them to share ssrc (synchronization source identifier) with the RTP stream it describes, but it sometimes matches (for Sender Reports) and other times not. Which RFC or draft should I read for this?
Vi.
  • 37,014
  • 18
  • 93
  • 148
1
vote
1 answer

libcurl RTSP client implementation c programming issue

I'm implementing a RTSP client in c programming using libcurl. In order to properly configure the RTP/RTCP sessions I require having the transmission port of the RTSP server (to send RTCP RR to it). I've been looking on the documentation but without…
user3676705
  • 11
  • 2
1
vote
1 answer

RTSP client sending $-delimited data before session has been completely set up (PLAY command issued) resulting in a 405 Method Not Allowed Error

I have a custom live555-based implementation of an RTSP client and server. I am using the 16/08/2013 build of Live555. I am streaming using Interleaved RTP-OVER-TCP as the firewalls we'll be using won't allow us to use UDP. (In other words, RTP,…
Stanley
  • 5,261
  • 9
  • 38
  • 55
1
vote
1 answer

How to send rtcp bye message using Java Media Framework (JMF)?

I am using a java application to set up a SIP session with nuance speech server using JAIN SIP java API. Then I prepared the system for recognition by setting it up by sending some MRCP commands like GET_PARAMS, SET-PARAMS, Define Grammar &…
Mr Matrix
  • 1,128
  • 2
  • 14
  • 28
1
vote
1 answer

How to receive a rtp, rtcp or udp, from a stream of gstreamer, on video HTML5?

I'm trying to get a video stream RTP/RTCP using HTML5, the stream was generated by gstreamer. I used examples of gstreamer, so I can pass through RTP ports:5000, and RTCP:5001, and can receive streams using gstreamer. But using HTML5 could not…
David Viana
  • 516
  • 2
  • 8
  • 17
1
vote
0 answers

AVPF scheduling algorithm

According to RFC4585: If R's FB message(s) was not suppressed by other receiver FB messages as per 5, when te is reached, R MUST transmit the (minimal) compound RTCP packet containing its FB message(s). R then MUST set…
Bob
  • 10,741
  • 27
  • 89
  • 143
1
vote
1 answer

Early media settings in AVPF

According to RFC4585, AVPF profile allows a device to send feedback earlier than the usual transmission of a regular RTCP packet. But, depending on the bandwidth, number of users and periodicity of regular RTCP packets, it is possible for the…
Bob
  • 10,741
  • 27
  • 89
  • 143
0
votes
0 answers

How can I get the data of a rtcp packet in freeswitch?

Recently I want to calculate the lossrate during a call in freeswitch,however,I can't find the way of read a rtcp packet.And what is the definition of flaws in freeswitch？
cqupt_zz
  • 1
  • 1
0
votes
0 answers

Is there any way to use sipjs or other method to get rtcp packets？

I use sip.js to mkae a webrtc test. And I wang to get some information from rtcp. I tried to use getStatus() API,but I want the rtcp packet form the other side.Because getStatus() can not calculate the out-bound rtp packt lost and jitter. Is there…
cqupt_zz
  • 1
  • 1
0
votes
1 answer

NTP Timestamp in RTCP Sender Report is Incorrect

I'm using Amazon's sample code to upload a rtsp stream from an IP camera to Kinesis video streams. Code found here: https://github.com/awslabs/amazon-kinesis-video-streams-producer-sdk-cpp/blob/master/samples/kvs_gstreamer_sample.cpp I would like to…
John Bantle
  • 1
0
votes
1 answer

RTP timestamp in data packets vs RTCP SR packets

I'm trying to understand the difference between the RTP timestamp as it occurs in RTP data packets vs as it is used in RTCP Sender Report (SR) packets. For the RTP timestamp in data packets I have established that: They are not based on wall-clock…
schneida
  • 729
  • 3
  • 11
  • 37
0
votes
1 answer

how to calculate Delay in RTP packets using RTP time and NTP time from RTCP

I am sending a video stream from the browser to gstreamer using webrtc. I can get the RTP time of the packets and NTP time from RTCP SR packets in gstreamer. At receiver At receiver I want to calculate the time elapsed since that packet was…
Usama
  • 159
  • 1
  • 13
0
votes
1 answer

RTP/RTCP Question Regarding Encryption in DES-CBC Mode

I have been working to create an RTP/RTCP client and server implementation as a project. I have hit a block when it comes to parsing incoming packets because I am having trouble understanding the encryption method. I have read RFC 3550 several…
hipeople321
  • 122
  • 1
  • 9
0
votes
2 answers

Using libVLCsharp to stream pw protected IP Camera Video but video not playing

I am trying to construct a UI for Onvif complaint devices. I've been beating my head against the wall for sometime now. I believe I have the custom URI correctly constructed. According to the ONVIF Programmers Guide we need to Get Profiles,…
BBridges
  • 130
  • 14
1 2
3
4 5