Re: [Serusers] Intermittent Audio

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@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(a)teigre.com> wrote: http://lists.iptel.org/pipermail/serusers/2005-January/014620.html http://www.linksysinfo.org/modules.php?name=Forums&file=viewtopic&p… >> 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(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 >>>>>>> Send instant messages to your online friends >> http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com