Sounds like your gateway does SIP ALG which in my experience breaks SIP in 99 percent of all communication (because it fixes where nothing is broken).
Had such problems when I specified a STUN server for clients - NAT traversal should be left to the server alone. It's perfectly fine to use private IPs in SIP/SDN and let RPORT handle it all.

Kevin


Am Do., 15. Nov. 2018 um 23:51 Uhr schrieb Paul Muaddib <paul.muaddib83@gmail.com>:

Hi

at the moment I have the setup

Internal.xml
ext-rtp-ip = auto-nat
ext-sip-ip = auto-nat

External.xml
ext-rtp-ip = “my_local_freeswitch_ip”
ext-sip-ip = auto-nat

RTP ports (only) are begin opened by UPNP and SIP via TCP is being keep open with “expire-seconds=600” and “register=true” (Is there an alternative for the expire-seconds ? nat-options-ping is only for endpoints registering to freeswitch, right?)

The reason why I am doing this is, that I dont want to open up a SIP Port in the Firewall.

But this setup up doesn’t seem right. Audio is working, inbound and outbound, both ways. But the echo application is not.

When I change
External.xml
ext-rtp-ip = “auto-nat”

I get one-way audio for inbound calls for the callee. The caller can be heard.

I tried to analyses it in wireshark, but I don’t understand it. The sip provider keeps sending INVITE messages while freeswitch is confirming it with OK but keeps going on until the connection breaks.

Thank you for your help


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