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] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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