SSRC | A high quality audio sampling rate converter | Video Utils library

I'm working on WebRTC streaming which streams video from a device to a browser. This streaming works in Chromium browsers just fine but fails in Firefox. There is a failure with the SDP exchange which then halts the rest of the connection (no ICE candidates sent after SDP exchange).

There are some issues with Firefox's answer SDP I've found but I haven't discovered a reason for the issues: SDP mentions VP8 but we use H264 only; m=video 0 has port 0 but typically that's non zero; I typically get an a=inactive line; a=sendrecv should probably be a=recvonly; many other lines are missing (for example, ICE-specific lines)

SDP examples below:

I used wireshark to capture the RTP stream sent with:

ffmpeg -f lavfi -i "testsrc=duration=5:size=cif:rate=25" -pix_fmt yuv420p -g 25 -bf 2 -an -c:v libx264 -f rtp rtp://127.0.0.1:1234 > play.sdp

ffmpeg -version ffmpeg version git-2020-03-15-c467328 Copyright (c) 2000-2020 the FFmpeg developers

As can be seen in bold, RTP timestamps go forward and backward. I expect them to be the same for every packet in the frame and then only go forward by 40ms (+3600 at 90khz clock) as per the H.264/RTP spec.

Also, according to that spec, the last packet in a frame should have its marker-bit set but here almost all the packets have this bit set.

Am I doing something wrong? Not understanding something? Or is ffmpeg support for writing H.264 RTP simply broken?

SSRC=0xA49C3DC9, Seq=3595, Time=3153114809
SSRC=0xA49C3DC9, Seq=3596, Time=3153114809
SSRC=0xA49C3DC9, Seq=3597, Time=3153114809
SSRC=0xA49C3DC9, Seq=3598, Time=3153114809, Mark
SSRC=0xA49C3DC9, Seq=3599, Time=3153125609, Mark
SSRC=0xA49C3DC9, Seq=3600, Time=3153118409, Mark
SSRC=0xA49C3DC9, Seq=3601, Time=3153122009, Mark
SSRC=0xA49C3DC9, Seq=3602, Time=3153136409, Mark
SSRC=0xA49C3DC9, Seq=3603, Time=3153129209, Mark
SSRC=0xA49C3DC9, Seq=3604, Time=3153132809, Mark
SSRC=0xA49C3DC9, Seq=3605, Time=3153147209, Mark
SSRC=0xA49C3DC9, Seq=3606, Time=3153140009, Mark
SSRC=0xA49C3DC9, Seq=3607, Time=3153143609, Mark
SSRC=0xA49C3DC9, Seq=3608, Time=3153158009, Mark
SSRC=0xA49C3DC9, Seq=3609, Time=3153150809, Mark
SSRC=0xA49C3DC9, Seq=3610, Time=3153154409, Mark
SSRC=0xA49C3DC9, Seq=3611, Time=3153168809, Mark
SSRC=0xA49C3DC9, Seq=3612, Time=3153161609, Mark
SSRC=0xA49C3DC9, Seq=3613, Time=3153165209, Mark
SSRC=0xA49C3DC9, Seq=3614, Time=3153179609, Mark
SSRC=0xA49C3DC9, Seq=3615, Time=3153172409, Mark
SSRC=0xA49C3DC9, Seq=3616, Time=3153176009, Mark
SSRC=0xA49C3DC9, Seq=3617, Time=3153190409, Mark
SSRC=0xA49C3DC9, Seq=3618, Time=3153183209, Mark

I am currently making a small discord bot that can play music to improve my skill. That's why i don't use any discord lib. I want the music as smooth as possible, but when i played some piece of music, the music produced is very choppy. here is my code:

"Answer" of SDP offer is not sent to Web App(running on Windows/Mac/Linux) from Android/iOS application after updation of Chrome to latest v89 (released on 9 March 2021).

It is working fine on Chrome v88 and below.

Offer SDP:

RFC tutorial on RTP / RTCP protocol seems very confusing to me. I cannot find any state transition diagram for this protocol like this. It doesn't clear the difference between NTP and RTP Timestamp. It says it is useful for calculating round trip time. Can't it be calculated with the RTP timestamp alone?

The source will send a SR Report if and only if it recently sent a RTP packet otherwise it's a RR packet. How much the time interval is it actually to determine that if the sender has sent a packet recently?

what does the mixer do exactly? Does it take all the RTP packets coming from multiple sources and then at the application layer read it and repack them to multiple RTP packets with only SSRC being changed now? what if the packets type are different.

Two Chrome browsers: Alice (A), Bob (B). Different networks, so i'm using Coturn server (my own).

The problem is that when A creates an offer - everything is ok, ice connection state goes to "connected", everything works fine. But if B creates offer - every peer receives the same Ice candidates, but ice connection state after 10 sec "checking" goes to "disconnected". It depends on in what network is B. Only on some networks there is such problem.

Here are the details:

Not working case:

B creates an offer. His descriptor is:

Hiii

I am using Kurento media server in my webrtc Project. I am want to Increase Video Quality for my web call.

setmaxsend/recbandwith() is not working for value over 500 kb.

I want to change max bandwidth tell me how can do that.

what I need:

1. Is there any way to to that
2. Can i find actual variable which is used to define the bandwidth.
3. I want to Set Max Bandwidth 2000kb.

My Sdp

I would really appreciate help with this as i'm breaking my head and can't get it right. I am trying to replace multiple strings within files with find and sed.

I am replacing spaces with \s I changed the delimiter for the command to + I am not sure exactly which special characters have to be separated.

find . -type f -exec sed -i 's+\\+\s+g' {} +

This is the string i am trying to change:

I am trying to create an automation wherein I can export a report from the Analytics tab from Studio.Youtube.

When I get to the page where I need to click the export button, nothing happens and it does not export the csv file. I have tried switching frames and windows but nothing happens.

Here is a sample of my code