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[madduck at madduck.net]

Hey,

we're experiencing a weird problem with Asterisk 1.8.13.1
(1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via
a PBX (sipgate.de) work perfectly fine, almost 100% of the time.

However, calls that are routed to sipgate.de, which then routes the
call back to our Asterisk instance are "silent" most of the time.

What I mean with that is that even though RTP traffic flows, neither
side can hear anything from the other.

This problem happens when people at site A dial someone at site
B using the number provided by sipgate.de, but also if people call
each other within a site through the external number, i.e. if I dial
089-1234567-100 from 089-1234567-200.

I have not been able to reproduce this problem with purely internal
calls, i.e. calling ext. 100 directly, so I am assuming there's
a problem due to sipgate's involvement. However, as far as
I understand, once the call is established (and both parties' phones
suggest that), the traffic flows only via Asterisk (directmedia
= update,nonat), so the problem is likely to be found there, no?

Before I shower you with debug logs and traces, I am wondering if
this sounds familiar to anyone…?

Thanks,

--
martin | http://madduck.net/ | http://two.sentenc.es/

if god had meant for us to be naked,
we would have been born that way.

spamtraps: madduck.bogus@madduck.net

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[madduck at madduck.net]

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 46.244.255.146:8058 (type 00, seq 000680, ts 340914880, len 000160)
Sent RTP packet to 46.244.255.146:8058 (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 46.244.255.146:8058 (type 08, len 000160)
Sent RTP P2P packet to 46.244.255.146:8058 (type 08, len 000160)

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

I did have a sniffer running on 46.244.255.146, 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.

Thoughts?

--
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@madduck.net

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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