I'm testing a VoIP system based on OpenSIPS. It has no RTPproxy, so calls do not pass through OpenSIPS.
I tried to make a call between two smartphones, and it succeeded. I also turned on Wireshark, and got this result:
Does that mean that the voice call from the first phone to the second phone went through the first RTP stream, and the voice call from the second phone to the first phone went through the second RTP stream?
Why couldn't it just use one RTP stream? It could just go back and forth.