[Asterisk-Users] Re: Remote SIP Connections

[Rich Adamson]

> 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.)