On Fri, Feb 2, 2018, at 10:37 AM, Benoit Panizzon wrote: > Hello List > > Asterisk 13.14.1 in use with pjsip stack. > > On the remote side is a SBC which performs some 'nat' detection. I > suppose this means the SBC listens from where it is getting RTP data > and then replies to that ip. > > As long as the asterisk is initiating the call this is fine, the > asterisk start sending RTP to the media IP of the SBC and the SBC is > sending media back. > > Now I want to do a hairpin call, simulating call forward on no answer > (yes this is the situation I observed the problem first) > > So incoming AND outgoing calls are via SBC. > > exten => destination,1,Progress() > exten => same,n,Playtones(ring) > exten => same,n,Wait(5) > exten => same,n,Dial(PJSIP/sip:external@sbc) > > What I now observe when I dissect this call via Wireshark (and set rtp > debug on etc). > > Call to destination is established, up to the Wait(5) we have two way > RTP audio between the SBC and the Asterisk. > > The external destination picks up the call. From what I see the media > ip addresses and ports are correct, no direct media is attempted. So > asterisk should 'simple bridge' oder 'native bridge' the call localy. > > But for some reason, the asterisk server is NOT forwarding any rtp, nor > is the SBC forwarding any rtp it is getting from it's remote side which > is definitely sending rtp data. (yes I have access to the SBC and did > sniff both sides). > > I fear, that both, the asterisk side and the sbc side are attempting > the same kind of nat detection and do not forward rtp until they > receive any packets. > > I did probably try all possible permutations of: > > direct_media=no > rtp_symmetric=yes > force_rport=yes > > But still no audio. > > Any hints on how to force asterisk to send the first rtp packet?
The "rtp_keepalive" option can be used to have the RTP stack send an RTP packet out. Try that and see what happens. -- 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
