Hello, I have a problem where on an outgoing call a Grandstream phone (GXP2130)
closes the incoming voice stream about 1 second into the call (the remote party hears the Grandstream, the Grandstream doesn’t hear thr remote party). I have verified with logs and traces that this is not a NAT issue or any other network-related problem. All incoming RTP packets arrive at the phone on the correct port etc. as declared in the SDP. I opened a ticket with Grandstream and they replied: “
*the phone starts receiving RTP with SSRC =0x0 which is wrong”.*
Is this an Asterisk problem or the phones? Is this something that can be fixed on the Asterisk side?