So your saying the problem may come from the router I have (Linksys E2500) because it's not doing the symmetrical Nat so the RTP is getting lost?
2012/4/19 Tony Graziano <[email protected]> > If there is NAT between your PC and the sipx server then: > > 1. The local firewall to your PC needs to have any SIP helper or > Application Layer Gateway "turned off". > 2. At your firewall where sipx is the SIP helper or Application Layer > Gateway " needs to be turned off AND the NAT type for the outbound NAT from > the sipx server needs to be symmterical. Home brew and residential routers > usually will not do this. > 3. If you have an entry in your intranet subnets that includes the PC > network number it must be listed ONLY IF IF DOES NOT GO THROUGH NAT. > > One of these three (or a combination of them) is keeping RTP from flowing. > > > 2012/4/19 Simon Brûlé <[email protected]> > >> I used the following command on the SipXecs server *tcpdump -n -s 0 -i >> any -w filename.cap* and then I transfer it on my computer so I could >> open it with my Wireshark and have a look at it. >> >> I joined the file to this e-mail so you could take a look and tell me >> what you think. This one is from a call I did from the softphone to the >> Hardphone where this one bugged like I described earlier. >> >> Thanks. >> >> 2012/4/19 Simon Brûlé <[email protected]> >> >>> As i dig more and more in Wireshark i came to the conclusion that the >>> Wireshark information that I just sent you is pretty much useless as I now >>> see it. I will keep looking for some piece of information that could help. >>> >>> Thanks. >>> >>> >>> 2012/4/19 Simon Brûlé <[email protected]> >>> >>>> My Computer is connected in the Lan of the company and my E2500 is >>>> connected in this Lan too. My Server SipXecs and my hardphone are on the >>>> E2500. >>>> >>>> So there is an other router between my computer and the router that >>>> have my Server connected on it. >>>> >>>> For the Wireshark part when I answer the phonecall I do from the >>>> softphone to my hardĥone those request are coming in until i close the >>>> call. >>>> >>>> 109 19.596310 192.168.175.22 192.168.175.136 SIP/SDP 1466 Status: 200 >>>> OK, with session description >>>> >>>> Status-Code: 200 >>>> [resent packet : True] >>>> [Suspected resend of frame:104] >>>> [Request Frame : 57] >>>> [Response Time (ms): 10950] >>>> >>>> followed by this one: >>>> >>>> 110 19.598495 192.168.175.136 192.168.0.1 SIP 714 Request: ACK >>>> sip:[email protected]:5060;transport=tcp >>>> >>>> Request-Line: ACK sip:[email protected]:5060;transport=tcp SIP/2.0 >>>> Method: ACK >>>> Request-URI: sip:[email protected]:5060;transport=tcp >>>> [Resent Packet: False] >>>> [Request Frame: 105] >>>> [Response Time (ms): 512] >>>> >>>> >>>> All those test have been done on Wireshark on the Computer with the >>>> Softphone on it. And the 192.168.0.253 that you see is the hardphone IP >>>> adresse. >>>> >>>> >>>> 2012/4/19 Gerald Drouillard <[email protected]> >>>> >>>>> On 4/19/2012 3:25 PM, Simon Brûlé wrote: >>>>> >>>>> How can I do a capture with wireshark on the SipXecs server? >>>>> >>>>> If you google a little you will find it. >>>>> >>>>> >>>>> About the ALG you think that the other Router that give the DHCP to >>>>> my Laptop and the Wan adresse of my router would have the Sip ALG >>>>> activate? >>>>> >>>>> That would be the only thing inbetween your softphone and the sipx >>>>> server... right? >>>>> http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm >>>>> >>>>> >>>>> >>>>> 2012/4/19 Gerald Drouillard <[email protected]> >>>>> >>>>>> On 4/19/2012 2:58 PM, Simon Brûlé wrote: >>>>>> >>>>>> I added 192.168.175.0/24 to the intranet subnet and I still have the >>>>>> same problem. >>>>>> >>>>>> 2012/4/19 Gerald Drouillard <[email protected]> >>>>>> >>>>>>> On 4/19/2012 2:37 PM, Simon Brûlé wrote: >>>>>>> >>>>>>> Hi, I know I already posted something very similiar to this problem >>>>>>> but I haven't found a solution to it so here i am reposting my problem >>>>>>> but >>>>>>> with more precision this time. >>>>>>> >>>>>>> I have a softphone (Jitis) on a Ubuntu 11.10 installation >>>>>>> connected to the network of the company. >>>>>>> >>>>>>> I have a router Linksys E2500 connected to the same network. The >>>>>>> laptop have the adresse 192.168.175.136 giving by dhcp and the router >>>>>>> have >>>>>>> the adresse 192.168.175.22 giving by dhcp too. >>>>>>> >>>>>>> On that router I have my SipXecs server and 2 hardphones >>>>>>> connected. My SipXecs server have the adresse 192.168.0.1, the internal >>>>>>> adresse of the router is 192.168.0.2 and the 2 hardphones have dhcp >>>>>>> adresse >>>>>>> given by the SipXecs server. >>>>>>> >>>>>>> The problem is the following : >>>>>>> >>>>>>> When I call with the softphone that is registered on the SipXecs >>>>>>> server to a hardphone that is registered on the server too the call get >>>>>>> there but there is no sound on either side and the hardphone is still >>>>>>> flashing like the call is still coming and i didn't answer it. By the >>>>>>> way >>>>>>> the phone is a Polycom 321. >>>>>>> >>>>>>> When i call from the Hardphone to the softphone everything is fine >>>>>>> except that the softphone can't do any sound but he can hear the >>>>>>> hardphone. >>>>>>> >>>>>>> The firewall on the SipXecs server is disabled, the firewall on >>>>>>> the router is disabled too, the SipXecs server is in the DMZ of the >>>>>>> router, >>>>>>> Sip ALG is disabled on the router too. >>>>>>> >>>>>>> On the SipXecs server System --> Internet calling I have the Nat >>>>>>> traversal enabled and the Server behind nat. The intranet domain is the >>>>>>> default one and for the intranet i put the 192.168.0.0/24. >>>>>>> >>>>>>> You may need to add 192.168.175.0/24 also if it is local. >>>>>>> >>>>>> >>>>>> I have seen polycom phones act like this before. In my case: >>>>>> The user portion of a SIP dialog MUST match the ACK and if it does >>>>>> not match exactly the phone will ignore it. Without a valid ACK the phone >>>>>> won’t start sending RTP and the UI won’t show the call as answered. You >>>>>> may want to do a capture on the sipx server and look at the results with >>>>>> wireshark. >>>>>> >>>>>> Sounds like you may still have ALG at the gateway on the >>>>>> 192.168.175.0 network. >>>>>> >>>>>> >>>>>> -- >>>>>> Regards >>>>>> -------------------------------------- >>>>>> Gerald Drouillard >>>>>> Technology Architect >>>>>> Drouillard & Associates, Inc.http://www.Drouillard.biz >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sipx-users mailing list >>>>>> [email protected] >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing [email protected] >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards >>>>> -------------------------------------- >>>>> Gerald Drouillard >>>>> Technology Architect >>>>> Drouillard & Associates, Inc.http://www.Drouillard.biz >>>>> >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> [email protected] >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>> >>>> >>> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~ > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected].**net<[email protected]> > > Helpdesk Customers: > http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> > Blog: http://blog.myitdepartment.net > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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