[asterisk-users] SSRC =0x0 in RTP

[Harel Cohen]

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?

Thank you,

Harel
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