Audio dropping

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Asterisk Users 3 Comments


We see some strange behavior with phone calls, we use Asterisk

SIP clients (all behind NAT at different locations, so not a single NAT
solution is used):
– x-lite
– linksys pap2t
– polycom kirk (multiple type numbers)
– polycom (multiple type numbers, hardware phones)

Our Asterisk servers stays in between (some calls are recorded). Asterisk is
running on a physical server (no virtual server software) with “old”
hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for
buffers). We use a MySQL backend (CDR records are stored in it and SIP users
are stored in a MySQL database).

We use a SIP provider with a trunk for outgoing and incoming calls, this is
also an Asterisk server if I’m correct. We currently do around 1000 calls a
week and max. do 10 calls at the same time. The Asterisk server is not
behind a NAT.

What could the reason be audio in 1 direction is dropping? (Normally from
the Asterisk server to the mentioned SIP clients.) No clear information is
in the logs (it is like the call ended normally) and not all calls are
having problem (most not, but it happens to often for us to start using VoIP
more at the moment).

To test if it was the firewall we disabled the firewall on the Asterisk
server and moved the Asterisk server before the other firewalls we have.

What could the problem be? And even more important what could solve it
(and/or explain it)?

3 thoughts on - Audio dropping

  • While the most usual problem is packet filtering / NAT, this generally
    manifests as no audio at all in one direction, not a drop in mid-call.
    But it’s possible that one of the intermediate transit providers is
    doing something “clever”. (Disabling ping, as you mention in your later
    email, is often a good indicator of a company with insufficient Clue.)

    Are you in a position to tunnel the traffic over a VPN or similarly flat
    and unfilterable network link? (This might be a good idea anyway.)