What solution to a NAT problem?

Hi

I’ve read this is a common problem but I haven’t found a documented solution yet :

My Asterisk server is behind a NAT:
Router is a Cisco I can fully manage, Adsl line has dynamic IP and DDNS is implemented on Cisco router
UDP port 5060 as well a range of 11 UDP ports (10000 to 10010 for RTP) are forwarded to asterisk

My co-worker has a SIP phone behind another NAT and he uses also a dynamic IP adsl line but without DDNS and no allowed to manage the router provided by ISP

The problem is the well known no-voice after a placed call is answered
In fact , SIP phone correctly registers to Asterisk and any called SIP phone (inside Asterisk lan or the co-worker one) ring OK

I’ve played around sip.conf :

externhost=
canreinvite=
localnet=
nat=
etc…

as well as rtp.conf, by setting the same UDP port range i’ve forwarded by router

The problem is still the same, no voice.

As it seems to me this is a well known and common issue, is there any known solution about this scenario, working and (appreciate) documented ??

Thank you very much

voip-info.org/wiki/view/Aste … +solutions