1 Replies Last post: Nov 13, 2007 1:02 AM by Guest  
Guest

Nov 13, 2007 1:02 AM

[OpenSBC] OpenSBC SIP Trunking - Null Session in CallSessionManager.cxx

Guest
1. Nov 20, 2007 12:23 AM in response to: Guest
Re: [OpenSBC] OpenSBC SIP Trunking - Null Session in CallSessionManager.cxx
Hi Jogen,

I am attaching a wireshark traces along with my configuration details.
Hope this will help you in analyze the problem with Outbound calls using Sip
Trunk.

Hello Joegen,


I grabbed the latest source code from CVS and configured OpenSBC for SIP
trunking. While it may not be prime time, it works quiet well except for a
couple of issues:-


1) Upon initialization, OpenSBC is able to register successfully with
various service providers. I configured a couple of Xlite softphones to
register with OpenSBC and used them for testing inbound/ outbound calls

with
various service providers.


  • Placing an outbound call from one of the Xlite softphones to an
external service provider (Gafachi) works fine (attached log -
outgoing.log)
* Placing an inbound call from an external service provider to opensbc
results in the following error in the log (attached log - incoming.log)

Many things could go wrong in a SIP trunk configuration. Routing rely
solely on the correct formating of the To URI. You need to let me know
about the specifics of your configuration like the domain of the SIP
provider and the kind of INVITE the provider is sending to reach your
trunk. BTW, your attachment did not make it. You can send it to me
directly and i'll see what I can figure out. An ethereal capture would
also be nice just in case we are investigating low level interop issues
aside from configuration issues.

4:43:39.304 DTL: CID=0x0af8 Event: ---> Inbound - INVITE
sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0

4:43:39.306 DBG: CID=0x0af8 Session CREATED

4:43:39.306 INF: CID=0x0af8 *** CREATED (UAS) CALL ***
d82a729522b3f145f0798b363d25b6a1@64.192.112.13

4:43:39.306 INF: CID=0x0af8 *** DESTROYED CALL ***
d82a729522b3f145f0798b363d25b6a1@64.192.112.13

4:43:39.307 DBG: CID=0x0af8 CALL: (inbound) : Session DESTROYED

4:43:39.308 ERR: CID=0x0000 GC: .\src\CallSessionManager.cxx:528
CallSessionManager::OnCreateServerSession::CallSession Attempt to
CreateReference() a NULL Pointer or none descendant of PObject!!!

4:43:39.308 DBG: CID=0x06cb *** MESSAGE ARRIVAL *** No Session
available
to handle INVITE sip:16462781042@192.168.96.115:5066;transport=udp SIP/2.0

4:43:39.312 PWL: CID=0x0000 Using Iface: 192.168.96.115 to send to
Dest:
64.192.112.13


Can you confirm if it's a bug/ configuration issue? The log file is

attached
for reference


2) While exploring various service providers, I found an issue with
authentication in SIP trunking mode. While placing an outbound call to an
external service provider from one of the UAs registered to SBC, if the
external service provider requests authentication and returns a 407, the

407
is relayed back by OpenSBC to the UA. This should not happen as all the
credentials for service provider (trunk-account information) is present
with
the SBC itself. Any comments?


I guessing OpenSBC was not able to identify the call as a trunk call
properly. You are correct that the trunk should have handled the
authentication instead of relaying the 407. If you are using the Main
trunk to route your calls to the SIP Trunk, you may try to use
"sip-trunk" parameter in our b2bua route

Example: sip:1212* sip:mytrunkprovider.com;sip-trunk=true

This would automatically tell the b2bua to route all calls bound to New
York to be routed to the SIP Trunk.

  • When I added sip-trunk=true without space as given above, OpenSBC
crashes
in SBCSIPTrunkReg::GetEgressAuthInfo function at the start of For loop.

Here are the details of few configurations, also attached the ethereal
capture here with.

IP's of various Entities:
OpenSBC 192.168.96.123
Xlite 192.168.96.36
Global IP of LAN 59.95.152.178
IP of voipvoip.com 69.90.209.57

Various Config Entries on Web Pages:

Sip Trunk Config:

B2BU Page

  • Rewrite TO URI check box is checked.

  • sip:*@192.168.96.123 sip:sip3.voipvoip.com:5060


I look forward to hearing from you regarding these issues. Please let me
know if you need any other information regarding the same.


Regards,

Gaurav



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