[Asterisk-Users] SIP UA behind NAT and REINVITE ???

[Cameron Beattie]

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 at 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


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