RTP Reflection for NAT firewalls ?

Discussion in '3CX Phone System - General' started by rudoo, Nov 8, 2016.

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  1. rudoo

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    Is there a way on 3cx PBX host to enable RTP port reflection for SIP user agents? E.g. Asterisk supports flag "nat=yes" resp. "nat=force_rport,comedia" in sip.conf of each user agent. This allows to run SIP phones behind nearly any types of NAT firewalls just "out-of-the-box" (including those firewalls which are doing port mappings). And all of this without having issues with one-way-sound and without configuring any custom port range bindings, port forwards and STUN servers. (At least this is the theory :D )

    It would be very helpful to have such a feature on 3CX as well and we would get rid of 99% of NAT issues with 3CX PBX host. May be it's already implemented and I haven't yet seen?

    (addendum 1: RTP port reflection shortly explained: https://sipsorcery.wordpress.com/2009/0 ... -problems/)
    (addendum 2: Explanation of the Asterisk mechanism for it: http://www.voip-info.org/wiki/view/Asterisk+sip+nat)

    rudoo
     
  2. rudoo

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    Addendum for anybody who's facing same problem:

    3cx fixed the bugs from server version v14. With new 3cx server version v15. Now the 3cx is behaving correctly according to nat traversal:

    Now with nat traversal enabled and STUN server enabled on your sip clients there is no issue anymore with firewalls between your sip clients and 3cx server v15, you can run your sip clients therefore "naked" without any vpn tunnel installation needed :)
    (Precondition remains, that you have asymmetric nat running, not symmetric nat. But this is simple network standards rules for having stun working correctly and has nothing to deal with 3cx bugs of the past)
     
  3. eagle2

    eagle2 Well-Known Member

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    What difference you observe in NAT behavior between version 14 and 15 ?
    Check also ALLOWSOURCEASOUTBOUND parameter, it should cover your requests of behavior like Asterisk PBX.
     
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