[Serusers] Intermittent Audio

[pat newham]

Thank you for the explanation...perhaps my view of the
nat issues are too simplified!! I realise now that
there is no point in invoking rtpproxy when the user
agents are behind the same nat on the same subnet. I
do not think the linksys router has any sip alg
settings. However while I acknowledge this is bad
design practice, I suspect now that the problem is
that my rtpproxy isnt working correctly anyway.

I found the following messages in /var/log/messages:

localhost /usr/sbin/ser: ERROR: send rtpp_test: cant
read reply from a rtp proxy
localhost /usr/sbin/ser: WARNING: rtpp_test: cant get
version of the RTP proxy
localhost /usr/sbin/ser: WARNING: support for the
rtpproxy has been temporarily disabled

I searched the archives and determined that I am
probably running the wrong version of RTPProxy so from
the /usr/sbin directory I ran the following command:

cvs
-d:pserver:anonymous at cvs.ser.berlios.de:/cvsroot/ser
co rtpproxy.

However this seems not to have made any difference.
Any further ideas? Do you think this is the source of
the problem?

Regards,
Pat.

 --- "Greger V. Teigre" <greger at teigre.com> wrote: 
> 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.
> >>>
> >>> Send instant messages to your online friends
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> >>>
> >>> _______________________________________________
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> >>
> >>
> >
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> 
>  

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