On 07/12/2012 09:19 AM, Benny Amorsen wrote:
"Kevin P. Fleming" <kpflem...@digium.com> writes:
That's quite interesting; can you describe a scenario where this occurs?
Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24
and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to
move phones between the networks without changing the SIP server
address, so you set 192.168.1.1 as the SIP server no matter which
network they happen to be on.
Now the phones which happen to be connected to eth1 will send a request
to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come
from 10.0.2.1. This could be solved if Asterisk did a connect() to the
socket and use the same socket for answering. That would tell the system
IP stack that this is in fact a connection, and so the system would
ensure that the reply source IP would be correct.
I must be missing something. If a phone sends a UDP packet to
192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1
interface on the Asterisk server? The only way I can imagine that
happening is if a router in between the phone and the server has been
told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems
like a bizarre way to construct a network. Getting replies from Asterisk
*back* to the phone would also require the IP stack on the Asterisk
server to route those replies back over the 10.0.2.0/24 interface
instead of the 192.168.1.0/24, which doesn't make any sense either.
chan_sip does have the ability to use connect()-ed sockets for dialogs
now, since that is required for TCP, TLS and WebSocket support. It
wouldn't be a huge leap to use them for UDP as well, if that was beneficial.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
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