Re: [OpenSIPStack] OpenSBC Media Proxy Problem
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Re: [OpenSIPStack] OpenSBC Media Proxy Problem
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From: Craig G. <cra...@gm...> - 2008-05-20 23:45:18
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Hi Whit,
Are the offices VPN or just double natted? If natted then your OpenSBC is
in the wrong place - it needs to have two interfaces - one on the internal
LAN and the other with a public facing address.
Alternately you could omit OpenSBC and port map an external IP address to
your asterisk box and set the 'externip' in sip.conf to your external port
mapped address and add the appropriate value to 'localnet'
Probably easier to create an inter office VPN IMHO.
Craig
-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf Of Whit
Thiele
Sent: Wednesday, 21 May 2008 1:45 AM
To: ope...@li...
Subject: [OpenSIPStack] OpenSBC Media Proxy Problem
Hey list members,
I have a couple of questions on a simple configuration with OpenSBC that I'm
having trouble proxying the media stream. I've read some other posts on this
topic, but nothing has help so far. Here is the setup:
Office SIP Phone <192.168.0.100>
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[192.168.0.0] Internal Office #1 network
----------------------
| Office #1 LAN Router |
----------------------
[25.x.x.x] External IP address of office #1
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INTERNET
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[44.x.x.x] External IP of Office #2
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| Office #2 LAN Router |
----------------------
[192.168.1.0] Internal Office#2 network
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OpenSBC <192.168.1.100>
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Asterisk<192.168.1.101>
The phone in Office #1 is registering to the Asterisk box in Office #2 using
the UpperRegistration of OpenSBC which is in 'Full Mode'.
I am able to register through OpenSBC with the Asterisk box from Office #1,
however I am not getting any media to get proxied via OpenSBC. I've selected
the 'Proxy-All-Media' but nothing seems to work.
I must be missing something simple in the configuration. Where should I
focus my attention? In the B2BUA routes? Proxy-Relay-Routes?
While debugging the SIP messages on the Asterisk output I get the following
SIP message while launching a call:
//---------------------------
// from Asterisk CLI
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.100:5060;iid=7375;branch=z9hG4bK74bda256ea24dd1184caf507b23fcaa4-9
4d4d1526a6e1a341bd360c9d23def44;uas-addr=192.168.1.101;received=192.168.1.10
0;rport=5060
Via: SIP/2.0/UDP
192.168.0.100:61798;branch=z9hG4bK-d87543-77271a19797af374-1--d87543-;rport=
61798;received=25.X.X.X
Record-Route: <sip:192.168.1.100:5060;route-shift=192.168.1.100:5060;lr>
From: "7101" <sip:71...@in...>;tag=114af83a
To: "5555" <sip:55...@in...>;tag=as262aed0b
Call-ID: ZTMxZTgzMzFiNDQ0Njk1ODFjNWYxNGJlZDhlMmFhMzY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5555@192.168.1.101>
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 22965 22965 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 14662 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
//-------------------------------
internalsip.com is an internal mapped DNS to the asterisk box:
[sip:internalsip.com] sip:192.168.1.101:5060
Debugging Asterisk RTP it shows that its trying to send RTP to 192.168.0.100
which is the NAT'd address of the office phone in Office #1 which doesn't
make sense to me since obviously it can't reach that network.
//-----------------
// from Asterisk CLI
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016083, ts 056160,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016084, ts 056320,
len 000160)
Sent RTP packet to 192.168.0.100:45976 (type 00, seq 016085, ts 056480,
len 000160)
//------------------
I feel that I'm 98% there... Does anyone have an idea on where to focus my
efforts?
I've attached a b2bualog snip as well in case this helps.
Best Regards,
Whit
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