[asterisk-users] Internal calls without voice transport

martin f krafft madduck at madduck.net
Mon Jul 28 07:52:36 CDT 2014

By chance, I managed to fig into this a bit and found the exact
moment when audio stops. It is exactly the moment when the
counterparty picks up and RTP debug output says:

  Got  RTP packet from (type 00, seq 000680, ts 340914880, len 000160)
  Sent RTP packet to (type 00, seq 026000, ts 3578986600, len 000160)
      -- SIP/lehel-sipgate-00003573 answered SIP/lehel-martin-00003572
      -- Remotely bridging SIP/lehel-martin-00003572 and SIP/lehel-sipgate-00003573
  Sent RTP P2P packet to (type 08, len 000160)
  Sent RTP P2P packet to (type 08, len 000160)

so RTP switches to RTP P2P and no more packets are received from the

I did have a sniffer running on, and Wireshark really
rocks, so now I know that the gateway firewall is at fault, and
indeed, for some reason, nf_conntrack_sip and nf_nat_sip were not
loaded. Now I am wondering how it worked in the first place, but
that's that. Maybe this will fix things.

Anyway, I don't quite yet understand what RTP P2P packets are or why
they are sometimes used and not at other times. I assume they are
packets intended to be exchanged directly between the two clients,
but since I have MixMonitor() on Asterisk, this shouldn't actually
be possible as Asterisk should always force itself into the middle.


martin | http://madduck.net/ | http://two.sentenc.es/
dies ist eine manuell generierte email. sie beinhaltet
tippfehler und ist auch ohne großbuchstaben gültig.
spamtraps: madduck.bogus at madduck.net
-------------- next part --------------
A non-text attachment was scrubbed...
Name: digital_signature_gpg.asc
Type: application/pgp-signature
Size: 1107 bytes
Desc: Digital signature (see http://martin-krafft.net/gpg/sig-policy/999bbcc4/current)
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140728/ec31f099/attachment.pgp>

More information about the asterisk-users mailing list