[Serusers] sip phones in different private networks have onewayaudio

[Greger V. Teigre]

If you use rtpproxy, you don't need to configure stun/nat on any UAs. I 
assume that everything is ok if phone1 calls phone2?  If no, you have a 
problem with on_reply processing as phone2 will not be registered as behind 
NAT and the initial INVITE will not force rtp_proxy.  If yes, but calling 
phone1 from phone2 is the problem, you probably have something wrong in the 
processing of either REGISTER (phone1 does not have the NAT flag set when 
you run serctl ul show) or in processing of INVITEs (the INVITE does not get 
a: IN rtpproxy_ip_address in the SDP payload and nortpproxy is not added at 
the end of the payload).
Hope this helps,
g-)

Steven Wang wrote:
> Hello
>
> I have one phone (phone1) in one network, the other (phone2) in public
> network. both can call the other side; phone1 can be heard by phone2,
> phone2 can't be heard. I don't have NAT set on both end, but I use
> rtpproxy on SER. Is NAT still necessary to be set on both phones?
>
> Thank you!
> steven
>
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