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AVAYA Call Routing Mechanisims

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Olisa

Technical User
Dec 10, 2009
30
DE
I have an IP412(v4.2) and an IP500(v5.0) on different sites. The two sites are linked by a VSAT IP connection. On each site, LAN1 of the PABX and the IP phones are on a private subnet (192.168.40.1-254) and the WAN port of the PABX has an office wide routable IP address.
I want to set up an IP trunk btw the two PABX through their WAN ports and enable small community networking but need some clarification first.

1. When an IP extension on site A calls an IP extension on site B, (after signaling and call establishment) does the call routing go through the two PABXes or is it direct IP phone to IP phone?
2. The phones on the two sites have the same private IP address range (192.168.40.1-254). Can this cause IP conflicts?
3.What ports do I need to open on my firewall btw the sites?
4.Do the IP phones on site A need to 'see' the IP phones in site B or only the two PABXes need to 'see' each other through their WAN ports?
5.Is H450 the 'best' protocol to use for signalling?
6.For 5 channels? How much bandwidth do I need to set aside for this?

Thanks
 
The answers to all these questions are in the documentation (See below), you are going to need to read them anyway so you may as well start now. But what you want to do is possible as we do it over Sat links to link together Oil rigs and offices literally all around the world. :)


ACS - IP Office Implement

"What the Crocodile Hat....was that?
 
Thanks but have been thru and thru that documentation and it does not address my specific querries.
 
I'm not 100% sure on my answers but I will try to help you:
1. It will go through IP Office but without using VCMs (verify you have direct media path ticked);
2. No, it does not!
3. Go to System | LANX and you can see there the RTP range ports;
4. It is enough for the IP Offices to see each other!;
5. I prefer SIP (more secure!);
6. For 5 channels you need maximum 64Kbps per channel if you use G711 or 8Kbps if you use G729. (The other codecs try to ignore them; they are not so good);

Best Regards,
P.

Help me to help you! Helping each other makes us stronger, smarter!
 
The only way you will be able to do IP phone to IP phone calls without the VCM involved is if the phones can see each other directly. so coming in on LAN1 and out on LAN2 may not allow this. Also on number 2 i think it may in the end cause issues. I would make sure everything is on different IP's. I have never tried to do it the way you have it but to me i dont think it will work.

Kevin Wing
ACS- Implement IP Office
ACA- Implement IP Office
Carousel Industries
 
I agree with Kevin, never tested 2 sites on same subnet because it just doesn't make any sense to do it this way!

bandwidth to allow is 64kb +overhead =90kb approx for good quality, both directions,
8kb + overhead = 35kb each way for G729 compressed quality

you need to open more ports than the RTP, have a look here

vpn is always recommended rather than NAT.
you need a h323 networking license which gives you 4 calls between systems.

I would upgrade the 412 to v5 also so that the network license will allow you to create an SCN and giving you more advanced features.
 
Indeed, that's wht I said read the docs, it is all in there, you also learn it on the Implemetation course, Kevin and TT are correct, DMP is direct between sets not via the system, you are thinking of RTP relay which isn't the same thing, and same subnets is a big no no :)

ACS - IP Office Implement

"What the Crocodile Hat....was that?
 
Is it me or does this one have disaster written all over it?

I'd suggest either paying someone to design and configure this or investing in some IPO training which will take you through all of this.

I just think with the questions you are asking about on this, suggest you are in for a steep learning curve. Not impossible, but will be a bit of a grind. It's not the questions that you have asked, but the questions you may not have thought to ask that will be your downfall.

Good luck with whatever you decide to do...

:)
 
Do you use VLAN's, Layer 3 switches?
Change the ip range one of the 2 sites, and let a good BP and/or network engineer have a look at you network!

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________
 
I am 100% sure of most of my answers but sometimes i am surprised with people who can do the theoretical impossible in a live situation.....

Answer 1 : If Direct media path is ticked on both sites then the voice traffic will NOT go through IP Office if the codecs are the same all the way.

Answer 2 : If both sites have the same ip range and subnet then i would be surprised you can address a Ethernet device on the remote site.
The local nic will never go to a router if packets are on it's own subnet, impossible!
So the answer to your question 2 is : No, it will not give any IP Address conflict but it will never work.

Answer 3 : No firewall between the sites, use a IPSec VPN Tunnel.

Answer 4 : With Direct media path enabled ( question 1 ) then YES, the phones must be able to `see` each other. Without Direct media path enabled then only the IP Offices must `see` each other.

Answer 5 : H.450 is manatory for SCN to work and also for the Avaya phones as in this protocol all the extra features are embedded.

Answer 6 : If using G.711 over a VPN Ipsec tunnel you need 124kbs bandwith per conversation, using G.729 you need 66 kbs bandwith. Without a VPN IpSec tunnel then you need 88kbs for G.711 and 29.8 for G.729 per conversation.

Advise 1 : Ask a professional for help, training or whatever. Without any help and experience you will hardly get a good working environment.
 
where did you get the bandwidth using ipsec intrigrant, sounds very high on both codecs
 
TT, that including overhead +/- 20 kbps.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________
 
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