SSRC Value

mediasoup libraries
1

Hi,
Is it normal to see a always changing SSRC in the consumer.getStats?

http://paste.ubuntu.com/p/9TCfSVpBZc/

If I want to output the RTPPlain to FFMEPG OR GST, is the SSRC value mandatory? Which one should I load?

2

This is documented. I understand that what to put in RtpParameters is hard, but this is documented:

3

Thanks IBC, so mainly I should provide the same listtype of RTPParams for an ffmpeg inputing into mediasoup ?
“[select=a:f=rtp:ssrc={AUDIO_SSRC}:payload_type={AUDIO_PT}]rtp://{audioTransportIp}:{audioTransportPort}?rtcpport={audioTransportRtcpPort}|[select=v:f=rtp:ssrc={VIDEO_SSRC}:payload_type={VIDEO_PT}]rtp://{videoTransportIp}:{videoTransportPort}?rtcpport={videoTransportRtcpPort}”

<— those are used for rtpIN, I should use the same type for an rtp out right? no additional stuff

4

For RTP in… I don’t know much about ffmpeg but something like that.

For RTP out: no, you must read the server side generated Consumer: https://mediasoup.org/documentation/v3/communication-between-client-and-server/#consuming-media-in-an-external-endpoint

5

I have been on this for days now, I am in close contact with the gstreamer community and no-one is really finding an answer for this, they tell me check with the mediasoup guys how they send their RTP.

And I am stuck in the middle here is the pipeline I came up with :to read from mediasoup consumer plain rtp.

The data is being sent correctly from mediasoup to RTP as I capured the packets audio and video and their payload types from the pcap files.

Here is my command:
GST_DEBUG=4 gst-launch-1.0 rtpbin name=rtpbin latency=50 buffer-mode=0 sdes=“application/x-rtp-source-sdes” udpsrc port=5557 caps=“application/x-rtp,media=(string)video,clock-rate=(int)90000,payload=(int)101,encoding-name=(string)VP8,ssrc=696790109” ! rtpbin.recv_rtp_sink_0 rtpbin. ! queue ! rtpvp8depay ! webmmux name=mux ! filesink location=123.web

This is for testing video only, I took the SSRC value from the consumer.getStats generated in realtime and executed this command afterwards. I get these errors:

https://pastebin.com/wbfiv9Ls

6

I don’t know what that is related to the doc I referenced:

You asked about the SSRC in RTP-out (Consumer side). Such a link explains that you must read the SSRC and codec payload type, etc etc from the server side consumer.rtpParameters. And you must transmit that info to the receiving GStreamer and tell it to receive the corresponding SSRC, payload type, etc.

Other than that, I cannot help with Gstreamer.