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

Reply via email to