Troubleshooting SUT Lite and SIP Trunking issues
Applying Client Side Debug
Directions:
Edit rcpinstall.properties
http://publib.boulder.ibm.com/infocenter/sametime/v8r5/topic/com.ibm.help.sametime.v851.doc/trouble/trbl_client_locate_wsp.html
add the following lines to the rcpinstall.properties file.
.level=FINE
com.ibm.collaboration.realtime.multimedia.level=FINE
com.ibm.collaboration.realtime.internal.telephony.level=FINE
com.ibm.collaboration.realtime.telephony.level=FINE
com.ibm.collaboration.realtime.telephony.tcspi.level=FINEST
com.ibm.collaboration.realtime.telephony.softphone.level=FINER
Note, this generates substantial amounts of debug, so after restarting the client, reproduce the problem and immediately quit the client and collect the logs.
Problem resolution
1) Did the Sametime client receive the SUT Lite capabilities via the Sametime policy?
Open logfiles and Search for the term,
“av.allowSIPTrunking=true”
If this option is true, you should see the client UI options for SUT Lite including the Sametime Phonebook
You will also see the “or number” text in the quickfind area
If you are receiving the SUT Lite policy, but don't see any phone or video icons, then there is likely a problem with the softphone registration.
2) Did the Sametime softphone register successfully?
for softphone clients to "find" each other, they register their contact information (IP, name) with a common registrar.
Add the debug flags referenced above and open the sip.log file.
Here is a successful register routine
Client attempts to register:
Apr 4, 2011 8:44:26 AM com.ibm.collaboration.realtime.telephony.softphone.sip.SIPDispatcher logMessage
FINE:
> SIP MESSAGE OUT:
REGISTER sip:st852primary.austin.ibm.com:5080;transport=TCP SIP/2.0
Call-ID: 0.0.70ED92A958945A2A@9.76.46.16
CSeq: 1 REGISTER
From: ;tag=1054.5380571533403
To:
Via: SIP/2.0/TCP 9.76.46.16:5062;branch=z9hG4bK4597379333695066260
Max-Forwards: 70
Contact: *
Expires: 0
User-Agent: Sametime-Softphone-8.5.2.20110322-2107
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, INFO, MESSAGE, UPDATE
Supported: path
Supported: outbound
Content-Length: 0
First registration can result in a 404 error this response is sent when the client tries to delete the old registrations
so the server responds that there are no registartions for this client.
Feb 17, 2011 9:32:18 AM com.ibm.collaboration.realtime.telephony.softphone.sip.SIPDispatcher logMessage
FINE: <<< SIP MESSAGE IN:
SIP/2.0 404 User Not Found
Call-ID: 0.2.18868D6306FEB0E0@9.77.138.44
Content-Length: 0
CSeq: 1 REGISTER
From:
;tag=465.13777184385543
To: ;tag=058008402189113184_local.1302886363466_111_212
Via: SIP/2.0/TCP 9.77.138.44:5060;branch=z9hG4bK2683042831963392189;received=9.77.138.44
after this you should see another REGISTER request
FINE:
> SIP MESSAGE OUT:
REGISTER sip:st852primary.austin.ibm.com:5080;transport=TCP SIP/2.0
Call-ID: 0.3.21F303D795C69BE4@9.77.138.44
CSeq: 1 REGISTER
From: ;tag=373.9213820990672
To:
Via: SIP/2.0/TCP 9.77.138.44:5060;branch=z9hG4bK2432648771467316541
Max-Forwards: 70
Contact: ;+sip.instance="";reg-id=990722829
Expires: 1800
User-Agent: Sametime-Softphone-8.5.2.20110213-0700
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, INFO, MESSAGE, UPDATE
Supported: path
Supported: outbound
Content-Length: 0
and a 200 OK response indicating a successful registration
Apr 4, 2011 8:44:27 AM com.ibm.collaboration.realtime.telephony.softphone.sip.SIPDispatcher logMessage
FINE: <<< SIP MESSAGE IN:
SIP/2.0 200 OK
Call-ID: 0.1.28FF90AB9F33A63E@9.76.46.16
Contact: ;expires=1800
Content-Length: 0
CSeq: 1 REGISTER
Date: Mon, 04 Apr 2011 13:44:20 GMT
From: ;tag=18.229686826254365
Path:
Require: outbound
Supported: path
Supported: outbound
To: ;tag=09451799906563574_local.1301676058593_26775_44991
Via: SIP/2.0/TCP 9.76.46.16:5062;branch=z9hG4bK400519828446889484;received=9.76.46.16
If the response from the SIP Registrar is not a 200 OK, you should receive an error. The error message should include the SIP error that prevented the softphone from registering.
3) Is TelephoneNumber in LDAP anonymously accessible?
If not Sametime Connect Client fails to get softphone number resulting in softphone function disabled. If you can not find phone icon on toolbar after logged in, check trace log under Help menu. If error regarding failure of telephone number found check LDAP whether telephoneNumber is anonymously accessible or not. This happens when LDAP binding is configured as anonymous access.
401 Unauthorized error
Causes
This will happen if the username is found in two directories
Open the STSC ISC and goto Users and groups
Manage users and make sure there is only 1 LDAP repository and that each name only has one unique entry
The SIP Proxy/Registrar is configured for TLS and LTPA tokens are not setup properly.
I checked the logs and see one of these errors
T] FFDC Exception:java.io.FileNotFoundException SourceId:com.ibm.ws.classloader.ClassLoaderUtils.addDependents ProbeId:246 Reporter:java.lang.Class@18901890
java.io.FileNotFoundException: C:\IBM\WebSphere\STMedia1\AppServer\profiles\st852primSTMSPNProfile1\optionalLibraries\proxy-registrar\com.ibm.ws.sip.interface.jar (The system cannot find the file specified.)
and a few of these errors
[2/21/11 8:09:03:625 CST] FFDC Exception:com.ibm.websphere.security.auth.WSLoginFailedException SourceId:com.ibm.ws.security.web.WebAuthenticator.validate ProbeId:3928 Reporter:com.ibm.ws.security.web.WebAuthenticator@37c037c
com.ibm.websphere.security.auth.WSLoginFailedException: Validation of LTPA token failed due to invalid keys or token type.
Workarounds:
1) Check the option to save password when logging in
2) Disable TLS
3) Setup LTPA Tokens between the ST Community Server and the SIP Proxy/Registrar
Debugging server side issues
The server can also provide useful trace information. First of all, what sort of installation do you have? Is it using TLS (default) or TCP? Are the Media Manager and Proxy/Registrar co-located on the same WAS instance or on separate servers? This page explains what trace options are required:
http://publib.boulder.ibm.com/infocenter/sametime/v8r5/topic/com.ibm.help.sametime.v85.doc/trouble/trbl_av_diagtrace.html
:com.lotus.sametime.telephony.*=all:
com.ibm.mediaserver.*=all:
com.ibm.telephony.conferencing.spi.*=all:
com.lotus.sametime.telephony.sipfocus.*=all:
com.lotus.sametime.telephonymanager.*=all
com.ibm.ws.sip.*=all:
com.ibm.sip.*=all:
For registration issues, the last two are most telling since they'll indicate from the server's viewpoint why the client cannot register (assuming of course the client managed to contact the server; there could be a firewall blocking the registration ports)
404 Error
Case sensitivity
The sametime client softphone registration is case sensititive. So if the 3rd party trunk is setup to route calls to ST852Primary.austin.ibm.com, but the client registers with lowercase st852primary.austin.ibm.com, the lookup will fail and result in a 404 error
503 Error
For what it's worth, 503 Unavailable usually means the port is wrong, i.e., it's trying to connect via TLS to a non-secure port. WAS returns "not available" since what's being requested isn't possible on the selected port. It can also happen if a secure port is missing certificates (e.g., when I setup MyAV with TLS / SSL and forget to exchange certs between the P/R and MyMCU, this is the error I see).
Error details
[2/21/11 13:45:29:421 CST] 0000003a SIPConnection 3 SIPConnectionAdapter connectionError error
java.net.ConnectException: Connection refused: no further information
Outgoing calls not getting to destination
This is usually caused by an outbound rule not being found. You can search the log files and search for the strings below to see the Proxy rule evaluation. Each entry returns a true or false value and two sequential true values must occur for a positive match. If a positive match is not found, check the rules.
genericproxy 3 GenericProxyHandler proxyRequest ENTRY
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: SourceAddress, value: 9.3.186.215} {9.3.186.187}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {false}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: RequestURI, value: 140.*} {sip:14851;transport=TCP}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {false}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: RequestURI, value: sip:*@vcs.lotus.com} {sip:14851;transport=TCP}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {false}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: SourceAddress, value: 9.32.134.120} {9.3.186.187}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {false}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: SourceAddress, value: 9.51.253.170} {9.3.186.187}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {false}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches ENTRY {Condition type: Method, value: INVITE} {INVITE}
[5/18/11 15:11:12:031 CDT] 0000003b routing 3 RoutingCondition matches RETURN {true}
603 error on incoming call
This error can occur if a call bypasses the Conference Focus. An inbound SIP Trunk route must be configured to terminate at the Conference Focus. If the route is terminating to the SIP Proxy/Registrar, then it will forward the request directly to the client and this is not allowed.
Check the incoming route and make sure it's terminates at the Conference Focus.
The conferencefocus port maps to Application servers > STMediaServer > Ports
SIP_DEFAULTHOST
SIP_DEFAULTHOST_SECURE
For example, in this case the Conference Focus is listening on Port 5063 and the Registrar is listening on Port 5080. So you would setup the 3rd party SIP trunk to send traffic to 5080 and then the incoming Sametime route would direct to the conference Focus on Port 5063
Example incoming route
INVITE
9.3.186.215
−
−
Check registered bindings
It's possible that some clients or the conference focus or packet switcher may not be registering properly.
Proxy Registrar fails to Start
If the Parser is unable to properly read the proxy.xml file containing the Routing rules, the server will not start and will throw an error in the logs, “Unable to load the configuration file proxy.xml”