[Asterisk-Users] SIP and NAT - best practices?

[Leo Ann Boon]

Trevor G. Hammonds wrote:

>While I have not used siproxd, I have read a bit about it.  From my
>understanding of the docs, the local SIP agents register to siproxd, but
>siproxd does not register to Asterisk.  So the calls will traverse the NAT
>properly, but features like MWI will not work in this scenario.  Also, this
>would be pure SIP URL dialling (e.g. usernam at domain.com) as opposed to
>traditional telephone dialling (e.g. 1-213-555-8080).  
>
>Please correct me if I am wrong, because I would really like to be (in this
>case).  :-)
>  
>
The docs are a little confusing. Look in the FAQ section: What types of 
operation does siproxd support?
Here's the text.

>   1) Siproxd as outbound proxy:
>      - Configure your local client to register with some 3rd party service
>        like Sipphone, FWD, Sipgate or any other.
>      - Configure your local client to use siproxd as OUTBOUND PROXY
>
>      Note: In this case, the local client does NOT register with siproxd
>      but only with the external SIP restration service. The only condition
>      is that siproxd needs to stay in the path of communication, therefore
>      the local client must be configured as to use an OUTBOUND PROXY.
>
That's all you need to do. All your clients will still register to 
Asterisk through siproxd, siproxd will take care of rewritting the SIP 
headers to differentiate requests for each client.

Leo