[Serusers] Intermittent Audio

[Greger V. Teigre]

Pat,

> The system is currently being tested by someone else
> but I believe they are behind a Linksys VPN router.

Have a look at these:
http://lists.iptel.org/pipermail/serusers/2005-January/014620.html
http://www.linksysinfo.org/modules.php?name=Forums&file=viewtopic&p=8820

> Are you suggesting it could simply be the settings in
> this?
Yes.

> 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.

I'm afraid it's not that simple. ;-) If the clients are directly routable on 
your LAN, you would be best off not doing any nathelper/rtpproxy. The rtp 
will then flow directly between the clients.  However, if you use nathelper, 
it depends on your knowledge of the clients.  If you send the SIP messages 
through an ALG, the ALG will change the SDP IP addresses (probably), unless 
it detects that both are on it's own LAN. This is dependent on the 
implementation.  Anyway, an ALG rewriting to send rtp in a hairpin (turn in 
the NAT) will probably support hairpin.  The problem is that you get all 
sorts of problems if the ALG is rewriting and you make some assumptions 
about the SDP payload.
You should probably dump the SIP messages on the outside of the NAT (I 
believe you said ser is on a public address outside?)

External scripts should never be used. If a script blocks, you block the ser 
ser process. If the same error is triggered in several processes, ser will 
stop responding.

> 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.

:-) Welcome to the club...

g-)

> 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 at 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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>>
>>
>
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