[asterisk-dev] STUN support in chan_sip revisited

[Klaus Darilion]

Am 09.08.2010 14:25, schrieb Simon Perreault:
> On 2010-08-09 04:35, Klaus Darilion wrote:
>
>> Based on your arguments I see the following conclusion (Asterisk as
>> client): Asterisk should use STUN for SIP only as a keep-alive when
>> using UDP and RFC 5626 is supported by the server. For RTP, STUN should
>> be used as part of ICE. In all other cases STUN should not be used
>
> Right.
>
>> and the proxy (service provider) should do NAT traversal.
>
> No. In this discussion, I only care about Asterisk.  It would be futile
> to impose constraints on the service provider.

Not explicit but implicit. If Asterisk is behind NAT and it does not do 
any NAT traversal (because the proxy does not support RFC5626) then this 
implies that the proxy has to do the NAT traversal or it wont work.

regards
klaus