Hi, Mike:
the default gateway is 192.168.11.254 not 192.168.11.1.
So i dont think the flow is 192.168.11.2 to 192.168.11.1.
On 9/17/05, Mike Williams <mwilliams(a)etc1.net> wrote:
I believe your problem is simple. With the SIP
protocol, you are sending
the streams like this:
192.168.11.2 -> 192.162.11.1 -> 221.21.X.X
After you answer, the clients negotiate for RTP traffic, and try to send
data directly from 192.168.11.2 to 221.21.X.X, not using the SIP server.
You are probably having problems actually routing the data (trying
pinging the 221.21.X.X box from your 192.168.11.2 client) or you're
having NAT issues. Is far as I know, you must have a direct route from
the caller to the callee to pass RTP streams; you can't proxy them
through the SIP server.
Good luck, and let me know if you have any more questions.
Mike Williams (mwilliams(a)etc1.net)
Charles Wang wrote:
On 9/13/05, Charles Wang
<lazy.charles(a)gmail.com> wrote:
Hi, ALL:
I use ser + mediaproxy + PSTN support, and my ser with two interfaces.
One is public IP address such as 211.21.xxx.xxx.
Another one is private IP address such as 192.168.11.1.
And I use XLite (192.168.11.2) register to SER's private interface(via
HUB only).
It can register sucessfully.
But when I make a call to PSTN with this XLite, the callee rings and I
answer it.
I can not hear any sounds from each side.
I try to register another XLite(192.168.11.3), and make a call to
another private XLite(192.168.11.2). I can hear rings but it is still
no any sounds from each side.
Can anyone tell me what it happens?
Best Regard
Charles