1 Apr
2005
1 Apr
'05
5:39 p.m.
Greger,
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.
Regards,
Vivienne.
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Dear Vivienne,
I wrote the rtpproxy section, so I'll respond for Paul.
See inline.
g-)
---- Original Message ----
From: Vivienne Curran
To: Java Rockx ; serusers(a)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.
Regards,
Greger
Java Rockx <javarockx(a)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:
Send instant messages to your online friends http://uk.messenger.yahoo.com
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.
Send instant messages to your on line friends
http://uk.messenger.yahoo.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Send instant messages to your online friends
http://uk.messenger.yahoo.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers