Norman Zhang [EMAIL PROTECTED] writes:
May I ask what ports are necessary for SIP communication through a
firewall? I read somewhere that UDP/5060 alone is enough. Some
recommends more ports to be opened for RTP.
For outgoing call establishment, you must pass traffic out from your
device to
On Saturday 18 December 2004 13:21, Tom Ivar Helbekkmo wrote:
My home firewall allows my Asterisk PBX to send any UDP traffic to
anyone, and keeps state, so they can answer. It also specifically
allows anyone to connect to UDP port 5060 on the PBX.
Interesting. Does that allow other people
Antony Stone [EMAIL PROTECTED] writes:
My home firewall allows my Asterisk PBX to send any UDP traffic to
anyone, and keeps state, so they can answer. It also specifically
allows anyone to connect to UDP port 5060 on the PBX.
Interesting. Does that allow other people to call you (first
Tom Ivar Helbekkmo wrote:
I guess the first few packets from them to you might get dropped
because they don't match an established outbound connection, but
as soon as you start sending packets to them, your firewall will
allow two-way flow...
That's the trick, yes. It works because RTP streams
Norman Zhang [EMAIL PROTECTED] writes:
Does performance suffers from this?
There shouldn't be any difference.
Do I need canreinvite=yes?
The question is, rather, will reinvites work?. As long as each
local SIP phone is able to initiate UDP communication with an outside
partner, and the
