VCS Interworking

Haydn von Imhof · ‎10-31-2012

Hi all.

I have the following issue.

Whenever I make a sip call from my TelePresence Blade or TX system through my VCS control and Expressway to systems outside my compnay the call is always interworked from SIP to H323 by my VCS control.

If I make a sip call from another endpoints other than my TX systems and from any other MCU other than the 8710 Blade the calls are not interworked and sip is maintained to the destination, which I want.

My VCS control interworking is set to register only and my TP Blade is only registered Sip.

I member from the London TSEC staff that I have been testing with have the same issue.

They got around the issue by creating a new VM VCS and turning interworking off and registering the TP Blade to that VCS to make calls.

I don’t have the ability to purchase a new VCS at the moment.

I canT turn off interworking on my VCS as I have a few ISDN Gateways in use so I need interworking as the Codian/Cisco gateways are still currently H323 only.

Is there a way I can get the calls made from my TP 8710 Blade not to interwork.

Martin Koch · ‎10-31-2012

As the start could you post the search details for a failed call?

What exactly do you dial when you dial out?

Please remember to rate helpful responses and identify

Haydn von Imhof · ‎10-31-2012

Hi Martin

Below is the seach history for the call made from my 8710 TP Blade to a Cisco TSEC venue.

The call connected but using H232 after getting a "Session Timer too small" message"

Search (401)

State: Completed
Found: True
Type: SIP (INVITE)
CallRouted: True
CallSerial Number: e321bec2-23db-11e2-90fa-0010f31a006f
Tag: e321c070-23db-11e2-a388-0010f31a006f
Source (1)
- Authenticated: True
- Aliases (1)
- - Alias (1)
  - - Type: Url
  - - Origin: Unknown
    - Value: 0number@omegadigital.co.za
- Zone (1)
- - Name: Omega Data Centre
- - Type: Local
- Path (1)
- - Hop (1)
    - Address: 172.X.X.X:5060
Destination (1)
- Alias (1)
- - Type: Url
- - Origin: Unknown
  - Value: sip:8number@cisco.com
StartTime: 2012-11-01 06:23:29
Duration: 7.33
SubSearch (1)
- Type: Transforms
- Action: Not Transformed
- ResultAlias (1)
- - Type: Url
- - Origin: Unknown
  - Value: 8number@cisco.com
- SubSearch (1)
  - Type: Admin Policy
  - Action: Proxy
  - ResultAlias (1)
    - Type: Url
    - Origin: Unknown
    - Value: 8number@cisco.com
  - SubSearch (1)
    - Type: FindMe
    - Action: Proxy
    - ResultAlias (1)
    - - Type: Url
    - - Origin: Unknown
      - Value: 8number@cisco.com
  - SubSearch (1)
  - - Type: Search Rules
    - SearchRule (1)
    - Name: Traversal Rule - URI
      Zone (1)
      Name: Traversal Client Zone
      Type: TraversalClient
      Protocol: SIP
      Found: False
      Reason: Session Timer too small
      StartTime: 2012-11-01 06:23:29
      Duration: 3.07
      Gatekeeper (1)
      Address: 41.X.X.X:7001
      
      Alias (1)
      Type: Url
      Origin: Unknown
      Value: 8number@cisco.com
      Zone (2)
    Name: Traversal Client Zone
    Type: TraversalClient
    Protocol: H323
    Found: True
    StartTime: 2012-11-01 06:23:32
    Duration: 0.62
    Gatekeeper (1)
    Address: 41.X.X.X:6001
    Alias (1)
    Type: Url
    Origin: Unknown
    Value: 8number@cisco.com

Nick Halbert-Lillyman · ‎10-31-2012

One thing I've noticed is that in order to keep the call as SIP the whole way through, you need to make sure that both the URI you are dialling as well as the destination search rule maintain full SIP URI format - i.e. URI@doman.com

If you are just using an E164 e.g. 12345 instead of 12345@domain.com, try modifying your search rule to append the domain at the end. We had this issue when dialling from TP server to CTS-3000 via CUCM and this solved the issue.

Haydn von Imhof · ‎10-31-2012

Ni Nick

When making a call from my TP Blade i am dailing full URI's.

My search rule also keeps the full URI and does not modify it.

Tomonori Taniguchi · ‎10-31-2012

What is value configured for “Minimum session refresh interval (seconds)”, Under VCS configuration > Protocol > SIP > Configuraiton?

If you make this value larger (for example, 700 or 1000), will that makes any difference?

Haydn von Imhof · ‎10-31-2012

It’s currently set to 500.

I will try a few higher values and provide feedback.

Paul Woelfel · ‎11-01-2012

Like Tomonori said, I would check if the SIP timer is to small. You should check on your VCS Expressway, because the SIP call fails over the Traversal Client - Server relationship.

A VCS allways try to preserve the protocol used by the calling party, so it will first try to reach the other party via the original protocol and then via the other. In your call the sip call fails, so the call is interworked on the vcs and sent to the Expressway via H323.

Regards, Paul

Haydn von Imhof · ‎11-01-2012

Update...

Ok I have tried upping the “Minimum session refresh interval” to 1000 but this made no difference, the call is still interworked by my VCS control.

If I switch interworking off on my VCS control then the call goes through fine from my TP Blade to the UK TX system and I get 3 Screen TelePresence and SIP>SIP is maintained.

But with interworking set back to "registered only" my VCS control always then interworks the call from SIP>H323 when calling out from my TP Blade to the UK TX system.