+1
From: [email protected] [mailto:[email protected]] On Behalf Of Joegen Baclor Sent: Friday, April 20, 2012 2:29 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Nat Problem I forgive all of you for not noticing that there is no ACK for the 200 OK. Forget RTP, it wont be there. :-) Check the jitsi side if it is respecting the Record-Route presented in 200 Ok. Take a wireshark capture from the PC too and send it together with the one from sipx. On 04/20/2012 06:14 AM, Michael Picher wrote: 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 list [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] Helpdesk Customers: 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 _______________________________________________ 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. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
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