11 Jul
2005
11 Jul
'05
12:46 p.m.
Hi all. Thanks to Jan and Andres for the help.
--- ------
Things are working better now.
I did some tests and now I´m sure. Asterisk don´t give a dam to
"a=direction:active". It only
cares about the NAT="yes" parameter in "sip.conf" file - or
sip_buddies table for realtime users... -
For me its ok. My SER/Asterisk host has a public IP. So, it works. I could debug sip and
rtp and I saw it erroring "could not find network to send RTP" and when the UA
send it first packet...TA-DA!! Everything works fine! RTP media path goes like oil!
----------------------------------
I disabled the SDP fix from my "ser.cfg" so I can make my NATed UAs - behind the
same NAT - talk to each other and establish the media path
based on internal values that, in this case, are true values.
To fix when a public UA calls a NATed one, I use a t_on_reply to fix the 200-0K of all UAs
that are being called from public UAs. When the INVITE cames from a NATed UA, I use
another route - for t_on_reply() - and it don´t fix the 200-OK and my first case - media
between NATed - keeps working...
Ok. But when a NATed UA from ANOTHER NATED NETWORK - calls a NATed UA from my network I
got problems. Problems because my "ser.cfg" won´t deal with SDP values and the
internal IP:Port for media path will be kept. With this, I will never be conversation
between then.
DO ANYBODY HAS ANY IDEA/PRODUCTION CASE TO DEAL WITH THAT?
----------------------------------------------------------
Thanks in advance.
Ricardo Poppi.
Date: Fri, 8 Jul 2005 20:51:58 +0200
From: Jan Janak <jan(a)iptel.org>
Subject: Re: [Serusers] NAT - Lots of flavours...
To: Ricardo Poppi <rpoppi77(a)giro.com.br>
Cc: serusers(a)lists.iptel.org, sobomax(a)portaone.com
Message-ID: <20050708185158.GY6497(a)localhost.localdomain>
Content-Type: text/plain; charset=iso-8859-2
Hello, comments inline.
On 04-07-2005 19:44, Ricardo Poppi wrote:
I just quickly looked into nathelper sources and it looks like it can
only rewrite the port number in SDP if you run force_rtp_proxy,
fix_nated_sdp seems to change the IP address only. I CCed to Maxim,
maybe he could clarify this better than myself.
If asterisk does not support symmetric RTP then you will have to put
the rtpproxy between the user agent and asterisk and call
force_rtp_proxy instead of fix_nated_sdp in the script. I am not sure
if I remember it correctly, but I think that asterisk should support
symmetrict rtp, so maybe the problem is in fix_nated_sdp function
which does not alter media ports.
---------
Does anyone on the list know if asterisk supports symmetric RTP ? In
other words, can asterisk interpret a=direction:active parameter from
SDP and send media to the source IP and port of the incoming media
stream, instead of the IP and port from SDP ?
---------
You can also try to put rtpproxy between user agents and asterisk and
call force_rtp_proxy instead fix_nated_sdp. It's not the best
solution, but this way you could verify if the problem is in unaltered
port number in SDP.
Jan.
Hi list,
I´m trying to put to work a NATed environment and want to share some
information and request some I don´t realized yet.
I use an asterisk gateway, with a public IP, working really fine for UAs
with public IPs. At the same machine I runs SER that receives all SIP
messages and handle when it should go to a SIP UA or to asterisk,
rewriting the port (to the one asterisk uses) and sending to it. I don´t
replicate register to asterisk, and use the user accounts as "peer",
instead of "friends".
My ser.cfg is using the "force_rport()" and "fix_nated_contact()" for
every REGISTER it receives from nat UAs - I know when it comes from a
NATed UA using nat_uac_test("2").
Every INVITE that comes from NATed UA passes through a
"fix_nated_sdp("2"), that rewrites the IP address of SDP headers. Using
a onreply route I fix the 200 OK INVITE message, just in case that the
NATed UA is on the called side.
The UAs I´m using are X-Lite, Clipcomm CP-100 IP Phone, and Grandstream
HT-488.
Below I wrote the different kinds of configuration into the UA and in
ser.cfg, and the results I got:
1) Using without touching the UA - It don´t know it is a NATed UA.
-----------------------------------------------------------------------------------------------------------------------------
All REGISTER are treated ok because the force_rport make SER respond to
the register on the same external IP:Port it received. On the same hand,
it stores the right URI into the location database making the UA receive
the subsequent INVITES or other SIP messages through the external IP:Port.
The INVITES that comes from NATed UA have their SDP IP address rewriten
by SER and the external IP takes place. But the port is kept the
internal value, so when the called UA tries to reach the
External_IP:Internal_port the NAT/Firewall probably block/drops the
packets, and the result is a one-way audio - The one-way audio is
probably due to the right value that comes from the SDP headers of the
called UA - asterisk -, that has a public IP.
2)
a=direction:active
----------------------------------
If I add into ser.cfg a "fix_nated_sdp("1")" command, it will add
the
"a=direction:active" parameter to SDP header of INVITE that comes from
NATed UAs. I saw that it´s happening but the asterisk seems to not
understand that and don´t expect for the first RTP packet to get the
IP:Port information of the media. A one-way audio is the result of that.
The asterisk is probably sending RTP packets to the
Ext_IP:Internal_port, and the firewall is blocking the packets.