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SBCE Initial setup 2

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Mar 23, 2012
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I have installed EMS/SBCE in a virtual environment with a 50 license on a stand alone weblm all v6.3.

First issue I am having is having the SBCE connect to the WebLM it keeps error out I did the Address):52233/WebLM/LicenseServer and all sorts of break downs.

Second I install a device but I am completely confused on setting it up waht IPs go where etc.. and reading the Docs make me more confuse Avaya has no straight forward documents that I came across on the support site.

Also this is a test environment where the SBCE ove has been setup for two dedicated VM NICs. Me assuming 1 for internet and 1 for intranet. So my plan was to put an IP Phone ont the internet side and see if I could connect to the Intranet through the SBC.


Today Makes Tomorrow
 
Are you able to reach or ping WebLM server from the SBC management interface? Test this as well on the second issue when you register a phone to SBC internet i interface by pluggingin a laptop. Also make sure that your interfaces are set to Connect on power in the VM NIC config.
 
Yeap I can ping both ways and power option is enable as well.

Today Makes Tomorrow
 
It should be https not http. If you can try that.
 
You're supposed to have 3 NIC, internal/management/external

"Trying is the first step to failure..." - Homer
 
For the H/A SBC is it still 3 NICs each and for the EMS just management/internal ? Agree here that the doc for this isn't very clear.
 
Think in 7.0 you can have management same as internal but in 6.3 it's 3 interfaces, at least on mine =)

"Trying is the first step to failure..." - Homer
 
So I Have two physical NICs on the VM server dedicated to the SBC, but in the SBC its 3 NICS Management A1, A2, B1.

Today Makes Tomorrow
 
SO you can put SBC on VM? I thought you need external proprietary hardware for SBC's? I know EMS can be virtualized, but wasn't sure about SBC.
 
Both EMS and SBCE starting in 6.3 can run on VMware no special hardware is needed for SBCE.
 
Question about the required ports to be opened between the sbc private interface and SM / CM. It shows RTP ports 35000-40000 between SBC and MG/Phones. Does that indicate the IP of CM or the MG? Our security team gets very granular and won't just open on a subnet.
 
MGs

Ideally you'd want to match up your network regions UDP port assignments with those. It doesn't matter if they're in the same VLAN and the SBC asks for media on 35000 and the G450 asks for media on 2048, but if you ever get into WAN QOS stuff, and just for the sake of consistency, make darn sure everything - h323 sets, sip sets, gateways, app servers like voicemail/IVR and SBCs are using the same ports.

The SBC should only ever be sending SIP signaling inside to the Session Managers.
 
Hence why the tcp/udp/tls only gets opened btwn sbc and sm and the media from sbc to mg?
 
Yessir to both questions

Also, checking how you shuffle in your environment could have phones go straight to the SBC and get the MG out of the equation if not required.

Also, there's how you shuffle - initial early media, early h323 station media, do you set up straight from the IP station to the SBC? Do you setup SBC--MG--IP station and stay that way, or go direct after the call's set up?

All very important questions that might expose gaps in your SIP PSTN provider's service or your other platforms. No shuffling, 1 gateway DSP per call will never fail you though - so long as you've got the DSPs to burn.
 
Just a heads up , if you have your SBC`s in a HA config , something that Avaya slipped under the radar is the your private interfaces need to be on a different subnet to your core equipment (session manager , CM) as if they are not the way the SBC`s handle GARP during an inter change will result in loss of audio , trust me this is something that will happen and cause head scratching as its totally random.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
Just another one on this , we discovered the issue after initial testing when we had no call recording in place the Audio issue on interchange of the SBC`s was not an issue , we then implemented DMCC call recording via AES it was at that point, although the call was direct media from endpoint to SBC , that the DMCC nailed the call to a gateway and voila fault introduced , just something else to consider when thinking about environment and DSP resource allocation.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.
 
Since I'm POC'ng the SBC's is there any docs on how to use the default/demo cert on the iPhone?
 
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