UC Voice Network

[brian.vanoy]

Question regarding the possible restructure of our UC voice network.

Current UC Implementation: Cisco Unified Communications 8.6.x

DIDs/DNs are assigned to a respective SIP trunk that terminates at the gateway (ISR router) located at each branch office.  Each branch connects to either the primary/subscriber server that is located at one of two data centers.  Each branch office also connects directly (static routing) to the primary data center over MPLS circuits; where outgoing internet (web filtering, IPS) is handled for all sites.

Possible Restructure

Company CFO desires IT to look into individual internet connections for each branch office using local ISPs in order to eliminate the cost associated with our MPLS/SIP circuits.  The idea is that we would implement Cisco ASA/FWs that would provide Site-to-Site VPN connectivity to our data center for each branch, however, how we would handle voice has become a concern. 

1. We have looked into transferring it out to a hosted service provider, however, we already have bought into the Cisco UC platform.

2. If we keep our UC platform, can we simply implement a single SIP that carries all of our DIDs to the data center and have a ISR router implemented that would handle voice for all branch offices, or would this demand that we implement PRIs at each location and keep the routers in tact.

[Jaime Valencia]

This is the Video Over IP forum, you probably want to try the IP Telephony or Other Collaboration communities for this.

[brian.vanoy]

Thanks Jaime, I'll relocate it.

[Robert Shaw]

Hi Brian,

This totally depends on the underlying infrastructure.  A centralised SIP solution would mean that your RTP streams will be traversing the internet links to each office.  We have started to use a centralised SIP solution but we have dedicated MPLS links at each location so we can implement QoS a lot easier.  Cheaper internet links usually don't have this ability, it depends on your provider.

If this was being built on standard internet links I would opt for a distributed SIP model.  Have an ISR sitting behind your ASA configured as a CUBE.  It does mean more equipment but the quality of the voice traffic will be better as it's all local traffic, this is especially important if you have any Fax over IP requirements.  Most ISP's now offer SIP as part of their internet connections.

I used the below white paper from Cisco to help me decide the best way forward,

http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/prod_white_paper0900aecd806780df.html

Ultimately your decision will come down to cost versus acceptable downtime.  How much can the business afford to be without a phone system for any period of time?  The distributed model is the safest way to go minimising downtime but is also the most costly.  Usually problems in distributed installations affect a single office.  The centralised model on the flip side is by far the cheaper option but carries the greater risks of affecting all offices if there is a problem.  You would obviously need to build in resiliency to allow for hardware or provider issues.

One last word of advice.  Whichever provider you choose, try to stick to one that has a presence in each country you operate in rather than going for local ISP's.  Of course this is assuming you are in the UK or Europe.  We have offices all over Europe and it is a lot easier to maintain a relationship with one provider than several in other countries.

Hope this helps.  Please rate useful posts.

Thanks
Rob

[brian.vanoy]

Rob, Thank you for your response and the helpful link.

Our current voice config actually operates using MPLS circuits that connect each branch to the central data center via static routing.  These MPLS links actually have the SIPs attached and are connected to the local Cisco ISRs that are installed at each office. These are setup as the Cisco UC gateway at each location with dialing-peers, etc. configured. 

We also have two Cisco UC servers (publisher/subscriber) that are setup to operate and function in an active/standby configuration for redundancy purposes. One located at our primary office and the secondary at our central data center.

Currently, the company is looking into local ISPs to provide internet connections at each location rather than reside on the local area network using the existing MPLS circuits. They are looking to save cost, but in the long run they would now have to reside on Site-to-Site VPN to communicate back to the central data center, meaning we would have to purchase a Cisco ASA for each location, as well as, since they want internet only - no SIP, have to find a manner to keep the voice system active.  It's all a bit frustrating, but a learning process for sure.

Thanks, Brian

[Robert Shaw]

Hi Brian,

No problem.  In that case the only real option is a Central SIP trunk.  You'll need to size a CUBE correctly for your environment, we have two 2921's one at each datacentre.  We only have around 500 users apparently the 2921's can carry thousands of calls though.

We have a dedicated line in the datacentre connected to the primary CUBE just for SIP which again will need to be sized correctly depending on call volume along with the codec you plan to use.  I would recommend using G729 personally, that way your overhead per call is around 8kbps instead of 64kbps.  You will need to think about DSP Resources too.  This will depend on what conference resources your are planning to use, if transcoding will be required for say Voicemail applications.  You will also need to think about Faxes, if of course you still have them, I reckon every company still has one lying around in a cupboard that users take out from time to time! ;-)

The secondary CUBE is connected to our Internet bearer.  Our ITSP uses this as a backup in case of failure with the primary.

As long as each office has a good Internet Connection you can configure ASA's for QoS to at least prioritise the voice traffic over the Site-to-Site VPN.

http://www.cisco.com/c/en/us/support/docs/security/asa-5500-x-series-next-generation-firewalls/82310-qos-voip-vpn.html

This should work fine.  You just need to do your homework and make sure everything is designed to handle the volume of calls each office makes and their is a suffucient amount of bandwidth to handle Voice and Data.

The fact that you have local SIP Trunks at the moment you should be able to get a good idea of the amount of simultaneous calls each office makes.  This will then give you a good idea of bandwidth requirement, CUBE build and QoS planning.

As for ITSP's, as I mentioned previously try to get a SIP ITSP that has a presence in each country you have an office in.  The internet breakout can be supplied by anyone, but for the voice it would be worth going with one supplier with local PSTN breakouts.  As an example we use COLT as they seemed to cover the majority of countries we have a presence in.

Let me know if you need any more help.

Thanks

Rob