Hi,
Thank you to everyone who has replied to me to date. Unfortunately
the problem is still there. Regarding Mr B's reply below. The BT100
client is behind NAT and I have rtp ports opened on my linksys router
with port forwarding enabled. I have recaped the problem again below
and would really appreciate any more thoughts or commenst to help
resolve this problem as soon as possible...My demo this week will not
go well if I cant figure this out!!
Recap:
Basically I think the problem comes down to nat.
I still get the rtp errors mentioned in below emails in the
var/log/messages
file whether voice works or not(strangely enough)...but here are the
scenarios that work and dont work:
I have ser and asterisk on a private natted network. They are reached
via a router which has a public address and does port forwarding. One
my clients(a bt100 hardphone is also on this lan but registers with
ser throught the public address anyway...so my understanding is that
the bt100's sip register message goes out onto the internet and back
in again.
Now if the bt100 rings a client (e.g. xlite or windows messenger)or
vice versa that are also on this lan...the call works and voice is
transmitted (even though they again register through the public
address by routing their sip messages onto the internet and back in
again). However if the bt100 tries to ring a client not on the lan
(on another network which may or may not be behind nat)...np voice is
transmitted....Surely this is a nat problem even through the rtp
errors are being displayed??....
If anybody has any idea how to fix this or if any more information is
required in order to troubleshoot please let me know. Like I
mentioned before this must be demoed next week...This natting
scenarion worked when clients registered direct to asterisk so i
presume its possible with ser.
Thanks in advance,
Aisling.
From: "Mr B" <"cabalitomb atshaw.ca"(a)fox.iptel.org>
Subject: RE: Re: [Serusers] SER and NAT and RTProxy
To: serusers(a)iptel.org
Message-ID: <0IB3004283LBCJ@l-daemon>
Content-Type: text/plain; charset=us-ascii
What I have found on the BT-100 (or any of the BT Series) is that the
unit
does not function very well behind any NAT without a RTP port open.
Also you may want to check your router if you are using one.
These are my findings of routers
SMC - seems to work the best for VOIP
Linksys - okay
D-Link - very problematic - seems to always retain some old info
Just my .000001% of a cent opinion
Peter
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org]
On
Behalf Of Ashling O'Driscoll
Sent: Saturday, January 29, 2005 8:56 AM
To: serusers(a)iptel.org
Subject: FW: Re: [Serusers] SER and NAT and RTProxy
After doing a small bit of testing I have a bit more information
which may help identify the problem....Basically I think the problem
comes down to nat.
I still get the rtp errors mentioned below in the var/log/messages
file whether voice works or not(strangely enough)...but here are the
scenarios that work and dont work:
I have ser and asterisk on a private natted network. They are reached
via a router which has a public address and does port forwarding. One
my clients(a bt100 hardphone is also on this lan but registers with
ser throught the public address anyway...so my understanding is that
the bt100's sip register message goes out onto the internet and back
in again.
Now if the bt100 rings a client (e.g. xlite or windows messenger)or
vice versa that are also on this lan...the call works and voice is
transmitted (even though they again register through the public
address by routing their sip messages onto the internet and back in
again). However if the bt100 tries to ring a client not on the lan
(on another network which may or may not be behind nat)...np voice is
transmitted....Surely this is a nat problem even through the rtp
errors are being displayed??....
If anybody has any idea how to fix this or if any more information is
required in order to troubleshoot please let me know. Like I
mentioned before this must be demoed next week...This natting
scenarion worked when clients registered direct to asterisk so i
presume its possible with ser.
Thanks in advance,
Aisling.
---- Original Message ----
From: ashling.odriscoll(a)cit.ie
To: serusers(a)iptel.org
Subject: FW: Re: [Serusers] SER and NAT and RTProxy
Date: Sat, 29 Jan 2005 13:58:14 -0000
Unfortunately this still hasnt worked. I changed the modparam line in
ser.cfg and ran rtpproxy again as follows:
cd /root/Desktop/rtpproxy
./rtpproxy -s 127.0.0.1
Then I restarted SER.
However now voice isnt being transmitted AT ALL...and i cant make it
work even by running:
cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsrot/ser co rtpproxy
Whats so weird is that I tested all this on wed evening and
everything worked fine, now I repeatedly get the rtp errors
documented in my first email on the /var/log/messages file.
Does anyone have any more ideas...Im supposed to be demonstrating
this sytem as prototype next week and it wont look very impressive if
voice wont transmit....
Very stumped,
Aisling.
---- Original Message ----
From: info(a)marikar.com
To: serusers(a)iptel.org
Subject: Re: [Serusers] SER and NAT and RTProxy
Date: Sat, 29 Jan 2005 02:30:43 +0100
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello,
I got the same error-messages. The rtpproxy works fine if I change
the line
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
to modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1")
and start
the
rtpproxy with the option -s 127.0.0.1
I hope this will help,
Achim
Am Donnerstag, 27. Januar 2005 22:01 schrieb Ashling O'Driscoll:
Hi all,
I have a strange problem with the audio with some calls. I have
setup
RTPProxy and nathelper modules. I came across an
error saying the
RTP
proxy was disabled but solved it by searching the
archives and
executing:
cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co
rtpproxy.
This worked fine and my audio was transmitted. However every now
and
again (apparently at random) my audio doesnt
work.When I look at
the
> error logs in /var/log/messages, I see the following:
>
> ERROR: send rtpp_command: cant read reply from a rtp proxy
> WARNING: rtpp_test: cant get version of the RTP proxy
> WARNING: rtpp_test: support for the rtp proxy has been temporarily
> disabled
> ERROR: force_rtp_proxy2: support for porxy disabled.
>
> The CVS command fixes it temporarily. Does this mean I just have to
> run the command randonly every so often?...Is there a way to
> permanently fix this?
>
> Thanks,
> Aisling.
>
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