[OpenSIPStack] OpenSBC Media Proxy Problem
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[OpenSIPStack] OpenSBC Media Proxy Problem
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From: Whit T. <de...@wh...> - 2008-05-20 16:45:14
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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
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| Office #1 LAN Router |
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[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 |
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[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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