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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