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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
