[Serusers] NAT - Lots of flavours...

[Jan Janak]

Hello, comments inline.

On 04-07-2005 19:44, Ricardo Poppi wrote:
> 
> Hi list,
> 
> I´m trying to put to work a NATed environment and want to share some 
> information and request some I don´t realized yet.
> 
> I use an asterisk gateway, with a public IP, working really fine for UAs 
> with public IPs. At the same machine I runs SER that receives all SIP 
> messages and handle when it should go to a SIP UA or to asterisk, 
> rewriting the port (to the one asterisk uses) and sending to it. I don´t 
> replicate register to asterisk, and use the user accounts as "peer", 
> instead of "friends".
> 
> My ser.cfg is using the "force_rport()" and "fix_nated_contact()" for 
> every REGISTER it receives from nat UAs - I know when it comes from a 
> NATed UA using nat_uac_test("2").
> 
> Every INVITE that comes from NATed UA passes through a 
> "fix_nated_sdp("2"), that rewrites the IP address of SDP headers. Using 
> a onreply route I fix the 200 OK INVITE message, just in case that the 
> NATed UA is on the called side.
> 
> The UAs I´m using are X-Lite, Clipcomm CP-100 IP Phone, and Grandstream 
> HT-488.
> 
> 
> Below I wrote the different kinds of configuration into the UA and in 
> ser.cfg, and the results I got:
> 
> 
> 1) Using without touching the UA - It don´t know it is a NATed UA.
> -----------------------------------------------------------------------------------------------------------------------------
> 
> All REGISTER are treated ok because the force_rport make SER respond to 
> the register on the same external IP:Port it received. On the same hand, 
> it stores the right URI into the location database making the UA receive 
> the subsequent INVITES or other SIP messages through the external IP:Port.
> 
> The INVITES that comes from NATed UA have their SDP IP address rewriten 
> by SER and the external IP takes place. But the port is kept the 
> internal value, so when the called UA tries to reach the 
> External_IP:Internal_port the NAT/Firewall probably block/drops the 
> packets, and the result is a one-way audio - The one-way audio is 
> probably due to the right value that comes from the SDP headers of the 
> called UA - asterisk -, that has a public IP.

  I just quickly looked into nathelper sources and it looks like it can
  only rewrite the port number in SDP if you run force_rtp_proxy,
  fix_nated_sdp seems to change the IP address only. I CCed to Maxim,
  maybe he could clarify this better than myself.

> 2) a=direction:active
> ----------------------------------
> 
> If I add into ser.cfg a "fix_nated_sdp("1")"  command, it will add the 
> "a=direction:active" parameter to SDP header of INVITE that comes from 
> NATed UAs. I saw that it´s happening but the asterisk seems to not 
> understand that and don´t expect for the first RTP packet to get the 
> IP:Port information of the media. A one-way audio is the result of that. 
> The asterisk is probably sending RTP packets to the 
> Ext_IP:Internal_port, and the firewall is blocking the packets.

  If asterisk does not support symmetric RTP then you will have to put
  the rtpproxy between the user agent and asterisk and call
  force_rtp_proxy instead of fix_nated_sdp in the script. I am not sure
  if I remember it correctly, but I think that asterisk should support
  symmetrict rtp, so maybe the problem is in fix_nated_sdp function
  which does not alter media ports.

---------
  Does anyone on the list know if asterisk supports symmetric RTP ? In
  other words, can asterisk interpret a=direction:active parameter from
  SDP and send media to the source IP and port of the incoming media
  stream, instead of the IP and port from SDP ?
---------
  
  You can also try to put rtpproxy between user agents and asterisk and
  call force_rtp_proxy instead fix_nated_sdp. It's not the best
  solution, but this way you could verify if the problem is in unaltered
  port number in SDP.

    Jan.