Re: [OpenSIPStack] OpenSBC Media Proxy Problem

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Whit,

Have you tried B2BUpperReg mode instead of full mode?   The logs 
indicate that your invite is relayed (proxy) instead of using the 
B2BUA.  Relay only call does not spawn the RTP Proxy.

Joegen

Whit Thiele wrote:
>  
>
>  
>
> 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>
>
>             |
>
> [192.168.0.0] Internal Office #1 network  
>
>        ----------------------
>
>       | Office #1 LAN Router |
>
>        ----------------------
>
> [25.x.x.x] External IP address of office #1
>
>             |
>
>             |
>
>            INTERNET
>
>             |
>
>             |
>
>  
>
> [44.x.x.x] External IP of Office #2
>
>        ----------------------
>
>       | Office #2 LAN Router |
>
>        ----------------------
>
> [192.168.1.0] Internal Office#2 network
>
>             |
>
>             |
>
>  
>
> OpenSBC <192.168.1.100>
>
>             |
>
>             |
>
> 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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