I've got a group of IP phones separated from their Call Mgr (3.3.2) and PRI PSTN gateways (used by both these users and users local to the CM LAN. 6608 ports so there's no H.323 or IOS involved at all) over a T1 dedicated for IP telephony (CBR PVC between an IMA port on 3620 @ remote & T3 @ CM site). It seems to work fine with light usage (< the T1 PVC), but I'm concerned what will happen if the link runs out of capacity (codec = G.711). There are about 50 users so saturation is unlikely, but certainly possible.
Is there anything I can do to protect the link from saturation? There is only voice over the link; data has another path.
Use locations in callmanager for CAC and you might want to use g.729 to the other IPphones at central site this will cut the banwidth and voice quality will be good so have a look at regions in CM for this .
I wont recomend G.729 remote-site--> PSTN because of tandem encoding issues and this will result in severe degradation of voice qaulity .This is my opinion from my experience.
Though you can test it and see for yourself call some mobile and international numbers see how it goes.
The ratio you have T1 to users 50 is good 33% of remote office could make g. 711 calls unless this is a call centre / telesales or something simillar where phone abuse is regular.
That's pretty much what I thought. One other wrinkle is that there are really 2 PVCs between this remote office and the centeral: a 1.5Mbps CBR used for voice and a 2Mbps VBR-nRT used for data. At the remote end, these all originate from a 3xT1 IMA port and terminate at the central on a T3 ATM port (central also terminates 2 10Mbps VBR-nRT PVCs to other sites on this same T3 port). I use policy routing on both ends to direct voice traffic based on the remote end's IP address (since IP phones are in their own remote VLAN/subnet). We do not run any IP phone services so I'm assuming that the non-voice traffic to/from the IP phone subnet is negligible.
I guess what I was looking for was some kind of CAC where if the link was full, the call would be rejected, rather than just piling on & degrading everyone with increased latency, discards, and jitter. G.729 would allow me to handle more calls and I'd hit the wall later, but doesn't really solve the problem.
One more thing... I had a bad experience with 12.2.13T with my IMA boards when chasing another problem. TAC eventually told me that 13T & 15T didn't play nicely with IMA (or maybe ATM in general, I disremember). I've never been able to find a bug or caveat in the RNs to substantiate this, but the IMA interfaces would not come up on anything after 11T. Made for an interesting evening a month or so ago.
Depending on how your dial-peers are set, you can configure "max-calls " to limit the maximum number of simultaneous calls on a dial-peer. This doesn't really help if you have multiple VoIP dial-peers for outbound calls, but can work in a true hub/spoke environment.
You may have to use a Gatekeeper solution to handle CAC and bandwidth management.
Another solution would be to use Cisco's SAA to monitor link quality and disallow calls if some basic service requirements are not met:
of course. but I'm a little confused. I have hub and spoke centralized call processing. So I define locations for the hub and each spoke. Bandwidth for the spokes seem obvious as the BW of the link connecting the hub with the spoke. What is the BW of the hub location? 0?
What if there is a mesh circuit between two spokes that will actually carry the traffic (that CM wouldn't know about)?
For hub and spoke using locations, I always specify the hub location as having 10000000, no chance of running out of bandwidth there.
Also with CM 3.3 you can have AAR working so that if you limit the spoke to say 1.2M, then run G.729, in the Very unlikely event you run out of bandwidth (and don't forget it's not a real physical limit, just an imposed one) the calls can re-direct over the PSTN.
One word of warning about AAr though, it seriously screws up if you have unity (or any other voice mail) at the hub location. because the VM has no idea who the incoming call is for.
Oh and don't worry about G.729 call quality because in your scenario double encoding shouldn't happen, so most users will have absolutely no idea.
The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
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Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...
This document describe how DST changes and how time changes are
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