You'd need RTP ports open for asterisk then.
Transfers and parking can be done at the SIP level, asterisk doesn't
have to be in the RTP path, as it can reinvite itself into the
callpath as necessary.
On Wed, Jan 27, 2010 at 5:23 AM, Vincent codecompl...@free.fr wrote:
Hello
I think I finally understood the issue/solution, but I'd like to make
sure I'm correct:
- In Diana Cionoiu's famous article on Freshmeat
(http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol),
regardless of whether SIP end-points use a public IP or are behind a
NAT, RTP packets flow directly between the two SIP end-points because
the SIP server only acts... as an SIP server, meaning it only acts as
a registrar (for SIP end-points to make themselves know with an IP +
RTP ports), and then as a Central office (to ring the other SIP
end-point, and close the connection when an SIP end-point decides to
hangup)
- OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call
transfer, call parking, etc.), it must remain in the loop, and hence,
by default (canreinvite=no), all RTP packets always go through
Asterisk, even if both SIP end-points live in the same network as the
Asterisk server (and hence, since NAT is not involved, there's no need
for any kung-fu with rewriting information in SDP packets and asking
the NAT box to open the relevant ports for RTP)
Is this correct?
Thank you.
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