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Mitigating VoIP issues when QoS cannot be guaranteed across SCN?

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slambram

MIS
Jun 12, 2010
15
US
Site A: IP Office 406v2 4.2(23), VCM 20, 2x PRI, VM Pro
Site B: IP Office 406v2 4.1(15), VCM 20, 2x PRI

We have multiple VPN tunnels between the sites using multiple ISPs, however none are of the guaranteed bandwidth type, so we can’t guarantee QoS. Thus, we have some issues with VoIP latency on SCN calls.

One thing you may notice is the excess PRI capacity on each site. Additionally, LD calls between the sites are “on-net” (free). We’re hoping to leverage this in a couple ways,

1. What’s the best way to program the IPO to ‘intercept’ SCN calls and route them out the PRI instead so that user notices nothing except a slight delay?
2. Is it possible to establish a dial-up/PPP ISDN data connection between the sites using PRI channels, so that VM Pro and other SCN-only VoIP traffic can have guaranteed bandwidth?

Any other suggestions for improving SCN VoIP, besides an MPLS or other dedicated link?

Thanks, -Joe
 
the only way to do that without dialing more than the 3 or 4 digits for your extension is to break the SCN, then setup shortcodes that will dial the LD number to the remote auto attendant and then the extension number.
Basically you dial Extn. 1234
the SC would be
1xxx
dial
18001231234D1N (or whatever number you need to reach the remote AA)
line group ID of your local PRI

that will dial out and then wait for the connection to be established and then dial the numbers you dialed as extensions.

it will take significantly longer to dial an extension but you have full line quality.

Joe W.

FHandw, ACSS (SME), ACIS (SME)


“This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter
 
sorry to your number 2, you can probably get some devices that will do that for you but that is something I have never dealt with as I usually tell customers to either cough up the money for a good connection or deal with the fallout of being cheap.
I have a lot of customers that just use their regular Internet connection for SCN via VPN and it usually works quite well. You can make sure that QoS is at least in your router/firewall configured properly and therefore not getting any self induced problems on the connections like people watching youtube all day or streaming music.

Joe W.

FHandw, ACSS (SME), ACIS (SME)


“This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter
 
I'm terrible on the ancient 406, but here's an undocumented feature that works on 500's:

1)
On the SCN line set the Outgoing and Voice Channels to 0
g8coxpyskbw28olfg.jpg


Make an ARS named **NET** (incl the stars) and add a short code for dialling the other site via the PRI (like Westi suggested).
This way you keep the signalling between the sites, but not the voice channels.
Even if your VPN connection is horrible, a slight delay on the signalling should not matter much.

Keep in mind that there will be no mechanism to prevent tromboning.
If a call is sent from System A to System B, and then transferred back to A, 2 PRI channels will be consumed.

As for the Voicemail, you might get some surprises....

2)
Never done and never will do this kind of connection, but press F1 in Manager and search for PPP.
You'll find quite a bit of info on how to configure this. Look at the topics 'WAN PPP' and 'Using a Dedicated T1/PRI'

What kind of connection do you have today? (ADSL, SDSL, etc)
And what's the brand/model of the VPN router?

Kind regards

Gunnar
______________________________________
Mille viae ducunt homines per saecula Romam

2cnvimggcac8ua2fg.jpg
 
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