Thank you for your comments. However for the moment I would like to stay with 0.8.14 [If you don't mind ;)]. I implemented the nathelper/rtpproxy script given in the onsip getting started document. The only difference is that I removed the "has_totag()" from the loose route section. I have two questions.

1) In the route[2] section, should the sl_send_reply("100", "Trying"); be sl_send_reply("200", "OK");?? I couldnt register unless I changed this line and this route deals with the SIP REGISTER message.

2) After I changed this I tried to make a call between two phones (on public addresses) and got a 404 message. Could there be an obvious reason for this? I am eager to stay with this script as it must obviously work and would be more reliable than my own script which is patched together from variors posts on the mailing list.

Vivienne.

"Greger V. Teigre" <greger@teigre.com> wrote:

I wrote the rtpproxy section, so I'll respond for Paul.

---- Original Message ----
From: Vivienne Curran
To: Java Rockx ; serusers@lists.iptel.org
Sent: Friday, April 01, 2005 12:25 PM
Subject: Re: [Serusers] Nathelper/RTPProxy not working for agents
behind NAT

> Hello Paul,
>
> Thank you for responding. I have now read the getting started
> document. I am confused as to why my config should have supported two
> private clients on the same subnet communicating via rtpproxy [even
> though again i acknowledge its not the most efficient way to process
> the call] but anyhow I have decided to try to modify my script
> according to the sample rtpproxy/nathelper enabled scripted in the
> onsip document version 3. I will work from this as it will provide me
> with a solid basis.

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.

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.

> I have a few simple questions though. I am getting an error with the
> parameter "has_totag()". The /var/log/messages says I am missing the
> loadmodule. What loadmodule supports the above parameter? Also I was
> unable to load the module uri_db.so. Is this module usually included
> with 0.8.14?

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.

I recommend that you download the 0.9.0 Getting Started source package on http://onsip.org/ and forget about 0.8.14 unless you have some very special reasons for not doing so.

> Java Rockx <javarockx@gmail.com> wrote:
> Perhaps our "getting started" document at http://www.onsip.org/ will
> help you. It's based on ser-0.9.x, but it does cover both mediaproxy
> and rtpproxy.
>
> Regards,
> Paul
>
>
> On Thu, 31 Mar 2005 19:22:23 +0100 (BST), Vivienne Curran
> wrote:
>>
>>
>> Hi,
>>
>>
>>
>> I am having problems troubleshooting a problem I am experiencing
>> with my SER configuration. I have ser 0.8.14 running with rtpproxy
>> and nathelper enabled. I have two phones on the same subnet behind
>> nat and I would like to make a call between the two. I want to
>> invoke rtpproxy for this as they both have private address [I know
>> this isn't the most efficient way as they're both on the same subnet
>> but I can worry about that later].
>>
>>
>>
>> When I ring from the phone 1 ( 2092) to phone 2 (2093), 2092 can
>> hear voice but 2093 can't. When 2093 ring 2092, there's no audio.
>> These phones are Grandstream BT100's. They have been configured to
>> listen on different SIP and RTP ports.
>>
>>
>>
>> 2092: SIP Port: 5060
>>
>> 2092: RTP Port: 5004
>>
>> 2093: SIP Port: 5061
>>
>> 2093: RTP Port: 5005
>>
>>
>>
>> I have tried to include my ser.cfg and SER message dumps but
>> serbouncers said the attachment was too big. I can try adding them
>> again if requiredI can confirm that my rtpproxy is working
>> (originally I thought it wasn't) by using "strace –d -f –F". I can
>> see a signal being returned.
>>
>>
>>
>> Any help would be appreciated or advise as to how I can proceed
>> troubleshooting.
>>
>> Kindest Regards,
>>
>> Vivienne.
>>
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