On Thursday 06 March 2008, Maxim Sobolev wrote: > Dan Pascu wrote: > > On Thursday 06 March 2008, Bogdan-Andrei Iancu wrote: > >> Hi Maxim, > >> > >> You stated: > >> > >> <quote> > >> The correct behavior of the tm module in this case would be to > >> continue with INVITE re-transmits until we get provisional response > >> and immediate CANCEL once that response comes in. > >> </quote> > >> > >> Is this based on RFC indication or a personal opinion? If RFC based, > >> could you please point me out the relevant section? > > > > I got the same impression that Maxim is trying to solve a problem not > > related to SIP (network packet loss) by hacking the SIP behavior and > > Your impression is completely wrong.
Really? Then why do you write in the other reply <quote> In my situation this problem has been aggravated by the magnitude of packet loss </quote> > Network packet loss is real if you > are using UDP, and even if you don't see it normally, it still happens > from time to time. This is what retransmissions are for. They deal nicely with occasional packet loss. Now if you try to run a SIP service over a network with 50% packet loss, then I can understand why you gasp and try to grab on any idea (no matter how unrelated to the original issue) that could help you through the thing. -- Dan _______________________________________________ Devel mailing list [email protected] http://lists.openser.org/cgi-bin/mailman/listinfo/devel
