[Users] NAT (again)

[Bogdan-Andrei Iancu]

Hi Mark,

you might consider using force_rport() function to overcome the port 
problem - openser will use the received port instead of the one 
advertised in Via.

also, if you have a script with nat traversal support, you should 
include in your nat_uac_test() the flag 16 . see:
    http://openser.org/docs/modules/1.1.x/nathelper.html#AEN369

regards,
bogdan

Mark Kent wrote:

>I'm testing my rtpproxy config with a linksys phone and a grandstream phone,
>either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric).
>
>Things generally work, but I have a problem with the grandstream
>behind the symmetric nat.
>
>This is the register:
>
>  U  NAT-IP:3284 -> PROXY-IP:5060
>  REGISTER sip:sip.example.com SIP/2.0
>  Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02
>  From: "Grandstream" <sip:frick at sip.example.com;user=phone>;tag=743335557afcade4
>  To: <sip:frick at sip.example.com;user=phone>
>  Contact: <sip:frick at NAT-IP:3280;user=phone>
>  Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar"
>  Call-ID: 11f6defb0857db17 at 192.168.0.100
>  CSeq: 116 REGISTER
>  Expires: 60
>  User-Agent: Grandstream BT100 1.0.6.7
>  Max-Forwards: 70
>  Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>  Content-Length: 0
>
>This is the resulting contact:
>
>  ...Record(0x2864b680)..
>  domain: 'location'
>  aor   : 'frick at sip.example.com'
>  ~~~Contact(0x2864d4e0)~~~
>  domain    : 'location'
>  aor       : 'frick at sip.example.com'
>  Contact   : 'sip:frick at NAT-IP:3280;user=phone'
>  Expires   : 34
>  q         : 
>  Call-ID   : '11f6defb0857db17 at 192.168.0.100'
>  CSeq      : 119
>  User-Agent: 'Grandstream BT100 1.0.6.7'
>  received  : 'sip:NAT-IP:3284'
>  Path      : ''
>  State     : CS_SYNC
>  Flags     : 1
>  Sock      : PROXY-IP:5060 (0x8122f50)
>  Methods   : 5695
>  next      : 0x0
>  prev      : 0x0
>  ~~~/Contact~~~~
>  .../Record..
>
>
>Now, what I don't understand is the two ports used above: 3280 and
>3284.  It looks like the phone thinks it is being natted to port 3280,
>but it's really getting port 3284
>
>When I make a call I get
>
>  U  NAT-IP:3284 -> PROXY-IP:5060
>  INVITE sip:16509415674 at sip.example.com;user=phone SIP/2.0
>  Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8
>  From: "Grandstream" <sip:frick at sip.example.com;user=phone>;tag=2dc2fc3c94e93951
>  To: <sip:16509415674 at sip.example.com;user=phone>
>  Contact: <sip:frick at NAT-IP:3280;user=phone>
>  Supported: replaces
>  Call-ID: 1f04f9c6ec474f70 at 192.168.0.100
>  CSeq: 26381 INVITE
>  User-Agent: Grandstream BT100 1.0.6.7
>  Max-Forwards: 70
>  Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>  Content-Type: application/sdp
>  Content-Length: 419
>
>  v=0
>  o=frick 8000 8002 IN IP4 NAT-IP
>  s=SIP Call
>  c=IN IP4 NAT-IP
>  t=0 0
>  m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101
>  a=sendrecv
>  a=rtpmap:0 PCMU/8000
>  a=rtpmap:8 PCMA/8000
>  a=rtpmap:4 G723/8000
>  a=rtpmap:18 G729/8000
>  a=rtpmap:2 G726-32/8000
>  a=rtpmap:15 G728/8000
>  a=rtpmap:97 iLBC/8000
>  a=fmtp:97 mode=20
>  a=rtpmap:9 G722/16000
>  a=ptime:20
>  a=rtpmap:101 telephone-event/8000
>  a=fmtp:101 0-11
>
>and my side sends back the Proxy Authentication Required, but it sends
>it to port 3280.  Now, is that based on the Via: header or the Contact:
>header, or (unlikely) the stored contact info from the REGISTER?
>
>The phone acts like it never got it (and I'm guessing it didn't) and
>sends the INVITE again without any auth data.
>
>The thing that bugs me is that this is an outbound call, from the
>phone out through the NAT box, I figured that would be doable.
>
>The Linksys942 works, so I'm hoping this is a Grandstream bug,
>or something I can fix in my openser config.
>
>Thanks,
>-mark
>
>_______________________________________________
>Users mailing list
>Users at openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users
>
>  
>