Causes of one-way hearing with some sip calls?

Question

Asterisk

Greetings.
The following scheme works:
Mini PBX Panasonic TDA-100 -> sip-gateway D-Link DVG-2102s -> Router -> SIP provider Zadarma
In 20% of calls, we do not hear the called party. Those. first attempt: there is a call tone, after a click, and silence. After experiments, it turned out that the click is a connection and the called subscriber said "hello", he hears us, but we do not hear him. In the operator's logs, the call is displayed as a successful connection with a conversation duration of 1 minute - everything is logical. On the second or third attempt, the connection succeeds.
Theoretically, where could be the snag? What can cause one-way hearing?

Answer 1

Mirror the DVG-2102s connection port and take a full traffic dump.
It's even better if you also dump a dump on the router's WAN at the same time.
Without this, only the assumption that the router breaks SIP in some cases.

Answer 2

Ask for free to record traffic, take a dump and look at the signaling sip to see if there is RTP from you and if it is on the right port. Resolved in 10 minutes.

Answer 3

if the subscriber does not hear you, RTP packets do not leave you,
if there is NAT (and most likely there is), look in his direction, check the RTP ports

Answer 4

First, take a dump on the PBX from which you are calling. If everything is fine, then start analyzing traffic on NAT devices, see what goes in and what goes out)) There really isn’t much work, just knowledge of SIP is needed. Good luck.