Benefits from multiple Callmanagers in a multi-site network?

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Oct 12th, 2007
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I am administering a multi-site environment connected via VPN. Currently, the remote site depends on a single Callmanager at the main site. The PRI is also located at the main site. This means that all of the call processing and paths for the remote site come across the VPN to the single Callmanager. The remote site suffers, at time, from terrible reliability and quality issues.


Would placing a secondary Callmanager at the remote site improve their reliability and quality issues? I'm not sure if it will because the calls are still at the mercy of the VPN and PRI being hosted at the main site. I've implemented voice VLAN's and QoS on both sides of the VPN, but that hasn't seemed to help. Our main site has multiple T1 connections with all of the traffic load balanced across them. This being the case, our provider says there is no way to reserve bandwidth for the VPN because of our T1 implemenatation. The provider adds that with this configuration there's no way to dictate which T1 the call packets will travel in on. Our option would be to reconfigure our data setup.


I would appreciate any input on this, especially on adding the secondary call processor at the remote site. My boss believes that just by adding this that there qualtity and reliability will improve.


Many Thanks.

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It really depends on what your issues are.


If you are experiencing a lot of call control problems(i.e. picking up calls, transferring call, putting calls on hold, etc) then a local CM might help.


If you are experiencing a lot of voice quality problems then the extra CM won't do any good since your voice circuits go over the VPN.


The load balanced T1s, however, are probably your main culprit. Voice HATES load balanced connections.


The reason it hates load balanced connections is because you are separate your RTP, voice bearing traffic, over multiple interfaces that have different transmit queues.


Multimedia is intolerant of multipath communications. Packets are received out of sequence all the time and dropped.


I would recommend you talk to your carrier about either a Fractional DS3 or MPPP bundles T1s. That would likely alleviate a considerable portion of your voice quality problems.


In the interim you should speak to your carrier about setting up policy based routing for RTP streams so that they traverse only one circuit. You would need to do so on your side as well.



tom_parker Fri, 10/12/2007 - 12:45
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instead on having to re-engineer our lines here, could we add a P2P T1 or a frame-relay circuit just for voice traffic? Is voice a comfortable with frame?

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