24 Aug
2006
24 Aug
'06
9:50 a.m.
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@sip.example.com;user=phone>;tag=743335557afcade4
To: <sip:frick@sip.example.com;user=phone>
Contact: <sip:frick@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(a)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(a)sip.example.com'
~~~Contact(0x2864d4e0)~~~
domain : 'location'
aor : 'frick(a)sip.example.com'
Contact : 'sip:frick@NAT-IP:3280;user=phone'
Expires : 34
q :
Call-ID : '11f6defb0857db17(a)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@sip.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8
From: "Grandstream"
<sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951
To: <sip:16509415674@sip.example.com;user=phone>
Contact: <sip:frick@NAT-IP:3280;user=phone>
Supported: replaces
Call-ID: 1f04f9c6ec474f70(a)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
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