Lately, I have been doing a lot of tests on parallel forking. I would like
to know if "Flag" is transaction base, message base or uri based.
Say, User2 register with SER in two locations, one is in the public network
(UAPub) and the other is in the private network (UAPri).
Again, suppose the NAT flag is 5.
Now when User1 call User2, after lookup("location"), there will be two
branches, User1 -> UAPub and User1 -> UAPri.
lookup("location");
if (isflagset(5) {
nathelper logic ...
};
What will happen to the flag in the above case? Does each branch get a
different flag?
Zeus
Hi Zeus,
Where can i find a reference for t_relay_to_udp() ? I couldn't find it in the admin manual or maybe you can give me an example for that, thanks in advance.
regards.
Hi List
I am using SEMS and it is working ok for me.....Is there a way around that
we can call a person in the running conference....as far as i know...it is
conference room kind of thing.......
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For those of you who made inquiries about CDRTool (CDR mediation
software for SER, Cisco and Asterisk), I have made it available for
download at:
http://www.ag-projects.com/OSS_CDRtool.html
CDRTool software is free to use for non comercial purposes.
Best regards,
Adrian Georgescu
Andrei,
> > Hello
> >
> > I have following use case:
> >
> > Endpoint1 <------> NAT <------> SER <------> NAT <------> Endpoint2
> > 10.0.0.10 TCP TCP 192.168.0.4
>
>
> First of all try to use UDP. NAT traversal works with UDP. Nobody really
> tested it with TCP. Theoretically you could get it working with unstable
> ser, special options and a well behaved UA (one that will always reuse
> the tcp connections and will keep the nat bindings open).
I have been using UDP for kphone and it alvays works fine. But I use
Windows Messenger too and it seems to me that the TCP is the only
possibility to connect it from a NAT-ed network.
> Yes, ser cannot find an open tcp connection and tries to open a new one.
> To get arround this you should force tcp aliases and use a very high
> timeout for the tcp connections (if a connection is not used, ser will
> close it after some time):
> - use tcp_accept_aliases=yes in your ser.cfg if you have compliant UAs
> (I think only kphone knows about them)
> - use force_tcp_alias(); if your UAs don't know about tcp alias.
> - edit tcp_conn.h and change TCP_CON_TIMEOUT and TCP_CON_SEND_TIMEOUT
> to a very high value
I changed the constants and now the connections are permanent.
But I still haven't solved the problem routing messages. Here is a snippet
of code from my cfg file:
if (uri==myself) {
if (method=="REGISTER") {
# digest authentication
if (!www_authorize("i-tel.sk", "subscriber")) {
log(1, "Authorization failed\n");
www_challenge("i-tel.sk", "0");
log(1, "www_challenge sent\n");
break;
};
# symmetric but don't advertise it -- force use of rport
if (client_nat_test("3")) {
log(1, "Client is behind a NAT\n");
if (! search("^Record-Route:")) {
fix_contact();
force_rport();
};
};
log(1, "Forcing tcp alias\n");
force_tcp_alias();
save("location");
break;
};
};
My undersanding is that the function force_tcp_alias() should add
the source port of the tcp connection to the list of aliases.
Then later existing connections are searched by userid and port
to be reused. Please correct me if I'm wrong.
In my case it doesn't work, I found the following message in the log:
tcpconn_add_alias: alias port 5060 for hash 17, id 1
so it tries to add the port 5060 insted of the source port.
Martin
> Problems:
> 1. SER do not start properly after reboot.
> 2. Can´t not log on to SIP-server trough NAT.
1. Still wondering...
2.[SER]---[NAT]<----> dial.mine.nu <----->[Clients]
I have an old p233 machine with 2st eth-cards if there is any advantage to
set up a Linux-router insted of my us-robotics NAT-firewall it is
possible.
Note: It is only for the hosting of the SER server.
In that case some iptables/SIP-hints would be nice :)
Regarding the connectiontest witch i preformed with kphone.
Internal case:
both sending and reciving states
External case:
Just sending case.....retrying but no responce from SER.
Hello Martin,
Please advise if you have tested with more than 2 Grandstream devices in the
SAME Nat ?
Ri Liang
------------
Date: Sun, 30 May 2004 16:16:17 -0700 (PDT)
From: Martin Winter <mwinter(a)noaccess.com>
Subject: Re: [Serusers] stun - Grandstream
To: Peter Boot <peter.boot(a)ihug.com.au>
Cc: serusers(a)lists.iptel.org
Message-ID: <Pine.NEB.4.60.0405301614070.7226(a)picard.noaccess.com>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Sun, 30 May 2004, Peter Boot wrote:
> Has anyone been able to get the Grandstream HT486 to work with MyStun or
any
>
> other opensource stun server ?
No Problem here.
I use the HT486 with stun 0.91 (Feb 21 - the latest release) from
vovida.org
Source is at http://sourceforge.net/projects/stun/
- Martin
Has anyone been able to get the Grandstream HT486 to work with MyStun or any
other opensource stun server ?
thanks
Peter
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How should one handle re-INVITE with RTP Proxy on? Should one run the
unforce_rtp_proxy(), run the force_rtp_proxy() again or just ignore the
proxy part and forward the request to the remote client.
Zeus