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@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.
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