or could be stun too if the media relay's just off in left field. i would setup for static outside address if you haven't.
mike On Thu, Apr 19, 2012 at 4:59 PM, Tony Graziano <[email protected] > wrote: > Most likely if that is the router sitting in front of sipx, yes. It's a > prerequisite for media relay to function. > > 1. Server behind nat has to be enabled. > 2. Support remote workers needs to be enabled (sip trunking does not need > to be enabled for remote workers, media relay does). > 3. NAT for server needs to have access to a public IP address by getting > it via STUN server or manually entering it. > 4. Firewall in front of sipx needs to be able to do symmetrical outbound > NAT (AON or FULL CONE NATE, at least for the outbound NAT of the sipx > internal address) and have any SIP helper turned off. > > If the two routers are adjacent and you can route (without NAT) you would > simply do so and add the PC subnet to the intranets page in sipx ONLY IF it > does not have to pass through NAT (i.e route or site to site vpn, etc.). > > Your router is changing the ports and hence sipx is expecting the audio to > come back on a port that isnt being sent by the router. > > 2012/4/19 Simon Brûlé <[email protected]> > >> 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/ >>> >> >> >> _______________________________________________ >> 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/ > -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square**** Suite 201**** Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com ------------------------------------------------------------------------------------------------------------ There are 10 kinds of people in the world, those who understand binary and those who don't.
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