Is RTCP SR/RR Required to play Webrtc Video

[Jaishreepadma R]

Hi ,

I would like to know, to play Remote video , are RTCP SR/RR packets mandatory??

Without receiving  RTCP SR/RR  video will not start to play?

Regards

Jaishree

[Christoffer Jansson]

Question is why do you not want to use RTCP? And yes, it's required for WebRTC (essentially by RTP).

[Jaishreepadma R]

Hi

Actually I was playing only Audio. With out sending RTCP Audio could able to play.

Now   along with  audio added the Video. I have doubt  that, do I need to send the RTCP data.

Now I am sending RTCP data From Originator to Terminator.

Both Audio  and Video is not playing...

Here with attached the chrome internals .. It shows that  received Video and audio packets.

None of packets are decoded. It  shows height, width everything  as zero.  Frames received per second is 18...

is it mean that could not able to decrypt the packet???

Regards

jaishree

[Jaishreepadma R]

Since Chrome logging was broken in Windows regardless of command line flag in M43.

Looking  for windows chrome logging enabled version.

Tried  to get log using chromium  browser in ubuntu.. No luck...

[Christoffer Jansson]

Have you tried appending =stderr or =stdout to enable-logging? You should see it in the console directly then rather than in the chrome log file. E.g.

-enable-logging=stderr --vmodule=*/webrtc/*=2,*/libjingle/*=2,*=-2

[sandhya rajasekar]

Hi , Im also facing the same issue as posted in the above image. I just wanted to know if its because of RTCP? 

Thanks and Regards,

Sandhya

[Peter Boström]

RTCP shouldn't be required for video to start playing, these packets usually arrive a bit later and would have added a delay of about maybe a second or so if they were required (but they're still required to be implemented by WebRTC endpoints, according to the standard).

--

---
You received this message because you are subscribed to the Google Groups "discuss-webrtc" group.
To unsubscribe from this group and stop receiving emails from it, send an email to discuss-webrt...@googlegroups.com.
To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/3b5130da-d502-4e9f-9692-bbd5662baf29%40googlegroups.com.
For more options, visit https://groups.google.com/d/optout.

[sandhya rajasekar]

Hi peter,

Thanks for replying. My remote video gets played after a delay of some 2 minutes though I can see rtp packets in wireshark. Can you suggest me on some reasons for that?

Thanks & Regards,

Sandhya

[Peter Boström]

If you're not receiving keyframe requests (RTCP) and the first packet gets lost then it won't be decodable until the next keyframe occurs anyways, which is every 10k frames if I recall correctly.

To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/01276d16-c417-41b8-b2df-b7a62c307b67%40googlegroups.com.

[sandhya rajasekar]

How to generate the RTCP keyframe requests? 

[Peter Boström]

Sending FIR packets should do it.

To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/b756ab97-28a1-4804-9872-8c9d486c7ef9%40googlegroups.com.

[sandhya rajasekar]

Thanks for the response. Will check that.

[Ashish Parab]

Are you able to generate FIR request? I am facing same issue.