[Asterisk-Users] Re: Remote SIP Connections

[Blake Krone]

Which is what I was finding out, but once I set it to the DMZ
everything was fine seen as I didn't have to worry about the ports at
all.

Now if I could only get video to work again I'll be all set!

I'll have to look into the IAX2 protocol also.

-blake

On 7/7/05, Rich Adamson <radamson at routers.com> wrote:
> > You can try to open up port for SIP 5060udp and RTP 100000-20000udp...
> > (default setting) to your asterisk box. You will also have to specify
> > that your extensions are nat=yes & your externip=xxx.xxx.xxx.xxx (in
> > SIP.conf) so that the SDP protocol will write the public IP and port
> > translations for RTP (voice data).  If this doesn't work,  switch to
> > IAX2 protocol-  there are many hard-phones out there that support IAX2
> > protocol-  You will only have to open up 4569udp on your firewall to
> > your asterisk box and thats it.
> 
> Better be careful with the RTP statement above as its not necessarily
> true for all implementations and configurations.
> 
> If asterisk initiates the RTP negotiation, "it" will use udp source
> ports from the range shown above. However, each sip phone vendor (hard
> or soft) can choose whatever port range they want.
>  XLite is in the 8,000 range
>  Cisco 79x0's are in the 16384 to 32766 range
>  etc.
> 
> If a remote device initiates the RTP negotiation, it may not fall into
> the range that you've stated. (E.g., don't bank on your favorite itsp
> falling into that range.)
> 
> 
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