NAT Problem with UPNP only for RTP - SIP tcp keept open
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
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
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Hi Kevin, Yes, Deutsche Telekom probably does some kind of ALG to make it "easier" But why does not the echo application work and normal voice call does work? Regards, Paul Am Di., 20. Nov. 2018 um 23:36 Uhr schrieb Kevin Olbrich <ko@sv01.de>:
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
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
participants (2)
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Kevin Olbrich
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Paul Muaddib