Deag Bogdan, I have force_rport() in the beginning of script as you can see in the link http://pastebin.com/mcec311 (highlighted section is where i added NAT traversal logic)
also the log of failure is at this link <<< call made and ACK timed out >>> http://pastebin.com/m1d11246a I tried to figure out the problem, the highlighted parts might be the problem area if you could please give a quick look at see where in configuration script i went wrong? I know i am asking for too much but please help me, I really appreciate your help ! Khan On Fri, Apr 10, 2009 at 7:44 AM, Bogdan-Andrei Iancu <[email protected]> wrote: > Hi Khan, > > The 401 is for a REGISTER (look at the Cseq header). > > anyhow, the lack of an ACK from the caller means the caller didn't received > the reply (200 OK). If the caller is behind a nat, be sure you do > force_rport() in script (at INVITE time) - this will correctly route back > the replies via the NAT. > > > Regards, > Bogdan > > Khan wrote: >> >> Ok, >> >> I guess I sort of see the problem but dont know how to fix it... i >> capture the trafic from the SjPhone UAC which transmit OPTIONS after >> 200 OK, it seems like its getting 401 from server on authentication, >> wonder why? >> >> Here is a link http://pastebin.com/m298ec8c6 >> please let me know if i am on the right track!!!! >> >> Thanks for all your time and efforts... >> >> Khan >> >> On Thu, Apr 9, 2009 at 11:00 AM, Khan <[email protected]> wrote: >> >>> >>> On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens <[email protected]> wrote: >>> >>>> >>>> Khan, >>>> >>>> Would it be possible to add a tcpdump/wirshark on the opensips and on >>>> the client in the external network? That make it much easier to debug. >>>> >>> >>> I haven't done this before so, let me try to get the tcpdump for you, >>> I will install wireshark today (like i said im rookie) >>> I will post the tcpdump today :) >>> >>> >>>> >>>> One question: If you use xlite internaly, is the call dropped after >>>> 35secs or not? >>>> >>> >>> No, it only happens outside the network, I believe my NAT traversal >>> works fine, for some reasons my voice reaches them but theirs is lost >>> somewhere in clouds :) >>> >>> >>>> >>>> BR >>>> >>>> Uwe >>>> >>>> Khan schrieb: >>>> >>>>> >>>>> Uwe, >>>>> >>>>> I am using xlite within my network which works fine the problem is >>>>> outside the network, Xlite sends SUBSCRIBE and SJphone Sends OPTIONS >>>>> request... >>>>> >>>>> An example of debug is as follows, >>>>> >>>>> >>>>> Xlite registered fine the dump during the call process is as follows, >>>>> the call last for 35 seconds in which other party could hear me but i >>>>> can see a message on my Sjphone "ACK message awaiting" and then it >>>>> disconnects with the message "Network failure" please review the >>>>> following link... >>>>> http://pastebin.com/dca5bbb0 >>>>> >>>>> Another example is this SJphone which registers fine but after >>>>> registration constantly sends the OPTIONS requsts. The link is as >>>>> follows: >>>>> http://pastebin.com/d3a4fb379 >>>>> >>>>> My opensips.cfg is at this link: >>>>> http://pastebin.com/d6ce3e43d >>>>> >>>>> Thanks for all your help ... >>>>> >>>>> >>>>> Khan >>>>> >>>>> >>>>> >>>>> On Wed, Apr 8, 2009 at 1:46 PM, Uwe Kastens <[email protected]> wrote: >>>>> >>>>>> >>>>>> Hi Khan, >>>>>> >>>>>> A easy way to debug this problem is to use a kind of network sniffer >>>>>> on >>>>>> your opensips and directly after your UA. Try to debug this issue with >>>>>> a >>>>>> softphone like xlite, so you can start your network dump on the >>>>>> client. >>>>>> >>>>>> BR >>>>>> >>>>>> Uwe >>>>>> >>>>>> Khan schrieb: >>>>>> >>>>>>> >>>>>>> Hi everyone, >>>>>>> >>>>>>> I'm rookie in SIP technology, strugling with several issues. I am >>>>>>> having problem with UAC's outside network. I have 3 UAC registered >>>>>>> within the network (SJ Phone, Xlite) they are working fine, I can >>>>>>> talk >>>>>>> within the network but the problem arrise when I use the UAC outside >>>>>>> my network. I am seeing two different things from two different >>>>>>> UAC's. >>>>>>> >>>>>>> 1. Xlite on a network behind NAT try to register, it registers >>>>>>> successfully after receiving 200 OK it starts senting SUBSCRIBE >>>>>>> requests, which results in 483 Erro (set up in my config) and when >>>>>>> call is placed on this it gives ACK time out, person on the other >>>>>>> side >>>>>>> can hear me but i cant hear him. >>>>>>> >>>>>>> 2. SjPhone is on another network behind NAT, it regiesters fine, and >>>>>>> after registration it starts sending OPTIONS request, which I have >>>>>>> configured to respond as 200 OK. It continiously keep sending the >>>>>>> requst and my config respond to 200 OK. >>>>>>> >>>>>>> My question is several parts, what am I doing wrong, >>>>>>> a) why don't I get ACK after 200 OK, >>>>>>> b) how do i handle SUBSCRIBE requests >>>>>>> c) how do i handle OPTIONS request >>>>>>> >>>>>>> The sever is simply being used as SIP server for calls, no IM, Video, >>>>>>> or other applications are implemented yet. There are OpenSIPS, MySQL >>>>>>> server, and RTPProxy is running on the box. >>>>>>> >>>>>>> Please respond to my request considering my skills in the SIP as >>>>>>> rookie, guide me on how to resolve problem... >>>>>>> >>>>>>> Thanks, >>>>>>> >>>>>>> >>>>>>> Khan.... >>>>>>> >>>>>>> Sorry for such a long email, I am frustrated :( >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> [email protected] >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> kiste lat: 54.322684, lon: 10.13586 >>>>>> >>>>>> >>>> >>>> -- >>>> >>>> kiste lat: 54.322684, lon: 10.13586 >>>> >>>> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
