William M. Sandiford
2005-Apr-07 12:57 UTC
[Asterisk-Users] SIP UA behind NAT and REINVITE ???
Hello: I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? Original Call Made SIP UA 1<--> NAT FIREWALL <---> Asterisk <--> SIP UA 2 Then REINVITE occurs and SIP UA 1<--> NAT FIREWALL <------------------------> SIP UA 2 Is this possible? Will using a STUN server help this at all? I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want. I want * out of the audio path) I have even tried putting the private IP of SIP UA 1 in the DMZ of the NAT Firewall and still no luck. Any Suggestions??? Bill -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 4/6/2005 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050407/2ae9d7d0/attachment.htm
Martijn van Oosterhout
2005-Apr-08 03:18 UTC
[Asterisk-Users] SIP UA behind NAT and REINVITE ???
On Thu, Apr 07, 2005 at 03:57:11PM -0400, William M. Sandiford wrote:> Hello: > > I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. > > Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? > > Original Call Made > SIP UA 1<--> NAT FIREWALL <---> Asterisk <--> SIP UA 2 > > Then REINVITE occurs and > SIP UA 1<--> NAT FIREWALL <------------------------> SIP UA 2Possible, yes. Whether it works depends on the firewall. Your problem is that UA2 is sending directly to the firewall and the firewall will block it because it knows nothing about UA2. Or not, if it supports partial matching on UDP ports. In theory a packet of UA1 to UA2 should open the back channel, except you run the risk of the firewall assigning a new port number, thus breaking everything. This is a problem uPNP was supposed to solve, the client can request an externally visible port on the router. Never seen any client that does this though. If you only have one UA you can get around it with port forwarding on the firewall... But you need to know in advance what ports SIP is going to use...> I have tried and tried and tried to get this working but with no luck > (well, I can get it to work with canreinvite=no, but thats not what I > want. I want * out of the audio path)Good luck! -- Martijn van Oosterhout Ecomtel Pty Ltd
MessageMy understanding (by no means definitive): You need a solution to the NAT problem for the audio stream. STUN will help with non symmetric NAT but not with symmetric NAT so it's not a complete solution. If you have UAs behind symmetric NAT you will need Asterisk or an RTP proxy in the middle of the call. Regards Cameron ----- Original Message ----- From: William M. Sandiford To: asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 7:57 AM Subject: [Asterisk-Users] SIP UA behind NAT and REINVITE ??? Hello: I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? Original Call Made SIP UA 1<--> NAT FIREWALL <---> Asterisk <--> SIP UA 2 Then REINVITE occurs and SIP UA 1<--> NAT FIREWALL <------------------------> SIP UA 2 Is this possible? Will using a STUN server help this at all? I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want. I want * out of the audio path) I have even tried putting the private IP of SIP UA 1 in the DMZ of the NAT Firewall and still no luck. Any Suggestions??? Bill -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 4/6/2005 ------------------------------------------------------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050411/29bc24bb/attachment.htm