[asterisk-users] SSRC =0x0 in RTP

[Joshua Colp]

On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote:
> 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?

Asterisk would be sending the RTP to the Grandstream. I'd suggest
getting a packet capture using tcpdump or wireshark to confirm what
they've said though. I just looked at the code and I don't see a way
that we'd ever have the SSRC be 0.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org