[Serusers] Intermittent Audio

[Greger V. Teigre]

Yes, an update could be of help. There was a protocol change (between 
rtpproxy and ser/nathelper) at one point. I don't remember when.
I would agree that getting rid of the error messages would be priority 1. 
As the warning says, rtpproxy is disabled and force_rtp_proxy will not help 
you...
    An updated version of the Getting Started document with NAT (both 
mediaproxy and rtpproxy) is in the works and will be released within the 
next couple of weeks (probably). Until then I suggest that you make sure 
that rtpproxy is really updated and running correctly and that they are able 
to communicate (permissions) through the /var/run/rtpproxy.sock socket.
g-)

pat newham wrote:
> 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.
>>>>>
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>>
>
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