<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2604" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV>> Thank you for your comments. However for the moment I would like
to<BR>> stay with 0.8.14 [If you don't mind ;)]. I implemented the<BR>>
nathelper/rtpproxy script given in the onsip getting started<BR>> document.
The only difference is that I removed the "has_totag()"<BR>> from the loose
route section. I have two questions. <BR></DIV>
<DIV>For NAT, the difference is not really that big. However, 0.9.0 has a
lot of new features and has proven stable in many large-scale setups. However,
0.8.14 is fine for the basics.</DIV>
<DIV> <BR>> 1) In the route[2] section, should the
sl_send_reply("100",<BR>> "Trying"); be sl_send_reply("200", "OK");?? I
couldnt register unless<BR>> I changed this line and this route deals with
the SIP REGISTER<BR>> message. <BR></DIV>
<DIV>No, this only sends an OK when things not really are OK. save("location")
will send the 200 OK if you get there. You are probably stopped by
the check_to(). If username (before@) is different from auth name in your
client, you will not be authorized.</DIV>
<DIV> <BR>> 2) After I changed this I tried to make a call between two
phones (on<BR>> public addresses) and got a 404 message. Could there be an
obvious<BR>> reason for this? I am eager to stay with this script as it
must<BR>> obviously work and would be more reliable than my own script which
is<BR>> patched together from variors posts on the mailing
list. <BR></DIV>
<DIV>Because you are not registered (never reached save() ).</DIV>
<DIV>g-)</DIV>
<DIV><BR>> Regards,<BR>> Vivienne.<BR>> <BR>> "Greger V. Teigre"
<greger@teigre.com> wrote:<BR>> Dear Vivienne,<BR>> I wrote the
rtpproxy section, so I'll respond for Paul.<BR>> See inline.<BR>>
g-)<BR>> ---- Original Message ----<BR>> From: Vivienne Curran<BR>> To:
Java Rockx ; serusers@lists.iptel.org<BR>> Sent: Friday, April 01, 2005 12:25
PM<BR>> Subject: Re: [Serusers] Nathelper/RTPProxy not working for
agents<BR>> behind NAT<BR>> <BR>>> Hello Paul,<BR>>>
<BR>>> Thank you for responding. I have now read the getting
started<BR>>> document. I am confused as to why my config should have
supported two<BR>>> private clients on the same subnet communicating via
rtpproxy [even<BR>>> though again i acknowledge its not the most efficient
way to process<BR>>> the call] but anyhow I have decided to try to modify
my script<BR>>> according to the sample rtpproxy/nathelper enabled
scripted in the<BR>>> onsip document version 3. I will work from this as
it will provide me<BR>>> with a solid basis.<BR>> <BR>> Please note
that the example in the document is based on the setup<BR>> (figure) found at
the beginning of the document. The tests done to<BR>> detect NAT will
match for your two private clients as they will have<BR>> private
addresses. Thus, calls between the two will be proxied even<BR>> though
it is not necessary (as I believe you want). The<BR>> nat_uac_test() function
can be modifed to do other tests if you have<BR>> some knowledge (due to
registration or other processing) about<BR>> whether the caller/callee is
NATed or not. <BR>> <BR>> As to the
Grandstream config, there is no need to have them listen on<BR>> different
ports as they will have different IP addresses. Do you<BR>> register to SER
with the server's public IP address or the private?<BR>> If you use the
public, SIP messaging will go through your NAT and if<BR>> you have a SIP ALG
(application layer gateway), it will attempt to<BR>> change the addresses to
public for the phone using port 5060 and<BR>> (maybe) not for the one using
5061. The simplest is to use the<BR>> private address in the
Grandstream phones as SIP server address. <BR>>
<BR>> <BR>>> I have a few simple questions though. I am getting an
error with the<BR>>> parameter "has_totag()". The /var/log/messages says I
am missing the<BR>>> loadmodule. What loadmodule supports the above
parameter? Also I was<BR>>> unable to load the module uri_db.so. Is this
module usually included<BR>>> with 0.8.14?<BR>> <BR>> The Getting
Started document is built on 0.9.0, which will shortly be<BR>> released as
stable (according to the core team). The has_totag() can<BR>> be found
in the uri module. Please verify that have the latest<BR>> rtpproxy.cfg file
as there were a couple of issues with an early<BR>>
version. <BR>> I recommend that you download the 0.9.0
Getting Started source<BR>> package on http://onsip.org/ and forget about
0.8.14 unless you have<BR>> some very special reasons for not doing so.
<BR>> <BR>> Regards,<BR>> Greger<BR>> <BR>>> Java Rockx
<javarockx@gmail.com> wrote:<BR>>> Perhaps our "getting started"
document at http://www.onsip.org/ will<BR>>> help you. It's based on
ser-0.9.x, but it does cover both mediaproxy<BR>>> and
rtpproxy.<BR>>> <BR>>> Regards,<BR>>> Paul<BR>>>
<BR>>> <BR>>> On Thu, 31 Mar 2005 19:22:23 +0100 (BST), Vivienne
Curran<BR>>> wrote:<BR>>>> <BR>>>> <BR>>>>
Hi,<BR>>>> <BR>>>> <BR>>>> <BR>>>> I am
having problems troubleshooting a problem I am experiencing<BR>>>> with
my SER configuration. I have ser 0.8.14 running with rtpproxy<BR>>>>
and nathelper enabled. I have two phones on the same subnet
behind<BR>>>> nat and I would like to make a call between the two. I
want to<BR>>>> invoke rtpproxy for this as they both have private
address [I know<BR>>>> this isn't the most efficient way as they're
both on the same subnet<BR>>>> but I can worry about that
later].<BR>>>> <BR>>> ! ><BR>>>> <BR>>>>
When I ring from the phone 1 ( 2092) to phone 2 (2093), 2092 can<BR>>>>
hear voice but 2093 can't. When 2093 ring 2092, there's no
audio.<BR>>>> These phones are Grandstream BT100's. They have been
configured to<BR>>>> listen on different SIP and RTP
ports.<BR>>>> <BR>>>> <BR>>>> <BR>>>> 2092:
SIP Port: 5060<BR>>>> <BR>>>> 2092: RTP Port:
5004<BR>>>> <BR>>>> 2093: SIP Port: 5061<BR>>>>
<BR>>>> 2093: RTP Port: 5005<BR>>>> <BR>>>>
<BR>>>> <BR>>>> I have tried to include my ser.cfg and SER
message dumps but<BR>>>> serbouncers said the attachment was too big. I
can try adding them<BR>>>> again if requiredI can confirm that my
rtpproxy is working<BR>>>> (originally I thought it wasn't) by using
"strace –d -f –F". I can<BR>>>> see a signal being
returned.<BR>>>> <BR>>>> <BR>>>> <BR>>>> Any
help would be appreciated or advise as to! how I can proceed<BR>>>>
troubleshooting.<BR>>>> <BR>>>> Kindest
Regards,<BR>>>> <BR>>>> Vivienne.<BR>>>>
<BR>>>> Send instant messages to your on line friends<BR>>>>
http://uk.messenger.yahoo.com<BR>>>>
_______________________________________________<BR>>>> Serusers mailing
list<BR>>>> serusers@lists.iptel.org<BR>>>>
http://lists.iptel.org/mailman/listinfo/serusers<BR>>>> <BR>>>>
<BR>>>> <BR>>> <BR>>> Send instant messages to your online
friends<BR>>> http://uk.messenger.yahoo.com<BR>>> <BR>>>
<BR>>> <BR>>>
_______________________________________________<BR>>> Serusers mailing
list<BR>>> serusers@lists.iptel.org<BR>>>
http://lists.iptel.org/mailman/listinfo/serusers<BR>> Send instant messages to
your online friends<BR>> http://uk.messenger.yahoo.com </DIV></BODY></HTML>