We are now calling the phone to enter the Mediasoup conference. The sip signaling of the other party does not contain ssrc. We all know that if we do not know the ssrc, we cannot call PlainTransport.produce(). We noticed that the console outputs the following content:
RTC::Transport::ReceiveRtpPacket( ) | no suitable Producer for received RTP packet [ssrc:1379614267, payloadType:0]
This is the data packet sent by the other party, which contains ssrc. Is there a corresponding Mediasoup API that supports listening to the Rtp data packet received by PlainTransport, and can obtain the ssrc in it? Or by other means? Or we can only ask the third party to add ssrc in the sip signaling(it’s hard).
No. The transport is trying to find a producer by ssrc (or mid, or rid), and if it can’t, the warning that you cited is the last thing that happens to the packet (before it is deleted).
Thank you very much for your reply. So we can only coordinate with a third party to add SSRC in SIP, although this is difficult, but it seems that we have no other feasible way. We know that Mediasoup is designed as a top tier video conference, but our telecom giants here are still relatively backward, such as SIP without SSRC, audio type with AMR, and so on, which is a headache. 
It looks more like SIP without an API to obtain SSRC for the signaling rather than SIP without SSRC, from what you said.
Been in the same situation myself. Ended up by adding a custom ssrc ‘‘x-header’’ after realizing that our SIP stack is impossible to modify to add it in the sdp body (to be standard compliant). Hope it helps.