Hi Greger,
The system is currently being tested by someone else
but I believe they are behind a Linksys VPN router.
Are you suggesting it could simply be the settings in
this?
I "think" I understand the nat issues associated with
sip and sdp fairly ok so would I be correct in saying
that if my two clients are behind nat(the same nat)on
the same subnet the rtpproxy should be invoked? This
would be my understanding of the situation but then I
saw a recent email (see message header below)which
suggests an external script should be used.
Re: FW: [Serusers] calls between UA´s b ehind same NAT
us ing nathelper/rtpproxy
Also what confuses me is that the scenario works
sometimes and yet other times it doesnt. I will
attempt to get a full message dump (of both the
working and non working scenario)from the tester if
that will help.
Kindest Regards,
Pat.
--- "Greger V. Teigre" <greger(a)teigre.com> wrote:
Pat,
You haven't said anything of the type of NAT you are
behind. To me it sounds
like an ALG (Application layer gateway) problem. Try
to turn of the SIP ALG
in your router. If not, please post a full SIP
message exchange. You need
to find out if they communicate through the NAT
(hairpin media) or directly.
That depends on the SDP payload in the INVITE and OK
messages.
The new Getting Started document on
http://onsip.org/ (you need to
register) has a thorough review of NAT issues and
rewriting. Recommend! (I
wrote it ;-) )
g-)
pat newham wrote:
Following on from my below email, I can now
definately
say the problem is not nat pings. Just to recap I
am
experiencing intermittent audio. It works when
the
phones have very recently registered, then
sometimes
theres one way audio and then sometimes no audio.
Does
anyone have any ideas what the problem could be
or
where I could begin to troubleshoot this?
Hi,
I have a strange problem. I have two grandstream
budgetone clients on the same subnet behind nat
registering with ser on a public address.
Obviously
their public addresses would be the same but
they
listen on different ports. When they initially
register, I can the call,audio is transmitted and
everything is successful.
However sometimes theres only one way audio, other
times theres no audio and then other times it
works....I am guessing that this is because the
nat
router is forgetting the nat mapping so after a
while
when the nat mapping is "forgotten" and
a packet
arrives destined for a client, the router drops
it....
Could someone verify this for me??...Am I on the
right
track?? I have the following settings in ser.cfg
which
I thought would keep the nat settings alive.
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) #
Ping
interval 30 s
modparam("nathelper", "ping_nated_only", 1) #
Ping
only clients behind NAT
I also increased the nat keep alives "pings" sent
in
the configuration settings of the grandstream
phone....Any further ideas??
Regards,
Pat.
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