18 Sep
2005
18 Sep
'05
6:50 a.m.
Mike:
Before I test these two interfaces(Private and Public), I have only
a public interface on my SER proxy. My NATed clients are XLite or any
SIP IP phones or SIP gateways. They work fine with SER and each other.
When I try to make these clients register from private interface,
the problem happens.
So I don't think it is the problem of NAT functions at XLite or Gateways.
On 9/18/05, Mike Williams <mwilliams(a)etc1.net> wrote:
I was having problems with NAT myself, and found this.
I thought it sounded like your problem.
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.h…
---Mike
5.1. SIP with NAT or Firewalls ( Back to Tutorials Page )
1.1. Problem Description:
Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on
itself they only carry call signaling (call setup, teardown,… and use RTP to carry the
audio samples.
The signaling usually uses fixed and standardized ports, but the RTP uses random ports to
exchange both call legs (incoming and outgoing audio).
Most firewalls/NATs are unable to link the signalling protocol packets with the audio
packets and are in some cases unable to tell where to send the audio to.
When making a call, everything will seem to go normal, caller id will get passed, ringing
will start, you can pick up and hangup the call, but no audio in one or both directions.
-----Original Message-----
From: Charles Wang [mailto:lazy.charles@gmail.com]
Sent: Sat 9/17/2005 2:40 AM
To: Mike Williams; serusers(a)iptel.org
Subject: Re: [Serusers] Re: Does SER works on TWO network interfaces with public and
private IP addresses?
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:
--
Best Regards
Charles
--
Best Regards
Charles
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:
--
Best Regards
Charles
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