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<DIV>Dear Vivienne,</DIV>
<DIV>I wrote the rtpproxy section, so I'll respond for Paul.</DIV>
<DIV>See inline.</DIV>
<DIV>g-)</DIV>
<DIV>---- 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></DIV>
<DIV>Please note that the example in the document is based on the setup (figure)
found at the beginning of the document. The tests done to detect NAT will
match for your two private clients as they will have private addresses.
Thus, calls between the two will be proxied even though it is not necessary (as
I believe you want). The nat_uac_test() function can be modifed to do other
tests if you have some knowledge (due to registration or other processing) about
whether the caller/callee is NATed or not.</DIV>
<DIV> </DIV>
<DIV>As to the Grandstream config, there is no need to have them listen on
different ports as they will have different IP addresses. Do you register to SER
with the server's public IP address or the private? If you use the public, SIP
messaging will go through your NAT and if you have a SIP ALG (application layer
gateway), it will attempt to change the addresses to public for the phone using
port 5060 and (maybe) not for the one using 5061. The simplest is to use
the private address in the Grandstream phones as SIP server address.</DIV>
<DIV> </DIV>
<DIV><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></DIV>
<DIV>The Getting Started document is built on 0.9.0, which will shortly be
released as stable (according to the core team). The has_totag() can
be found in the uri module. Please verify that have the latest rtpproxy.cfg file
as there were a couple of issues with an early version.</DIV>
<DIV>I recommend that you download the 0.9.0 Getting Started source package
on <A href="http://onsip.org/">http://onsip.org/</A> and forget about 0.8.14
unless you have some very special reasons for not doing so.</DIV>
<DIV> </DIV>
<DIV>Regards,</DIV>
<DIV>Greger</DIV>
<DIV> </DIV>
<DIV>> 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>>>
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list<BR>>> serusers@lists.iptel.org<BR>>>
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