CME OVER A WAN

[saxon747]

Hi All,

Just curious if anyone has CME working over a WAN ... was there any gotchas or issues u faced ? ... looking on Cisco web site it is possible with the HDV card to convert the g.711 to g.729 ... I want to come up with some solution to that has a CME on HQ site and 5 remote sites on 128K , cost is an issue ... other than the HDV card is there any other way of getting g.729 over the WAN ? anyone know if Cisco are going to be supporting it on a future release of CME ?

Tnxs in advance for ur help

Rgrds

 

[computerhighguy]

128K WAN connections is not a good thing. A G711 call with take about 90K. A G729 call will take about 30K. Here is the link to the bandwidth calculator by codec.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

Will CME work over a WAN? Yes. But you must make sure your WAN can handle the traffic. Voice is time sensative and therefor will need priority on your network (QoS). You also need latency less than 150 ms for best results. Assuming you have a stable WAN and use the G729 codec, you can get about 3 calls per WAN link at the same time.

What does G729? The simple answer is the DSP does the analog (or digital) conversion to IP using the codec you select. I am not aware of any Cisco DSPs that don't convert to G729, but I also am not totally familair with the older analog gateways like the AT-8. If you are looking for a cheap gateway, look at the MC3810. If you have a T-1 coming into your facility, then it is really cheap (around 250-300 dollars). It won't run CME, but CME can use it as a gateway. Why are your WAN links so slow?


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[saxon747]

Ta for that ... I'm new to this voice stuff! , I'm just trying to get me head around it all ... I wasn't sure if perhaps in the future a new version of CME or router IOS would support the g729 codec.

Trying to get some ideas of what I could do for this customer with the hardware they have ... they have 2600xm at the core and remotes sits have 1760's ... core has a E1 card to their PBX (not in use), each site has 2 E&M cards installed, but not connected to their PBX or on the routers

it was just last week when I was playing with CME for the first time, wanted to see if I could use it on their network

excuse my lack of voice knowledge but would I need a 3810 on each site ? does the 3810 have the DSP's built in already ? would each site need a gateway ? I guess this gateway converts the g711 to g729 ... I don't think the HDV card will with the 1760's ... I'm tossing some idea around in my own head ;-) , so migth be ranting ;-) ... sure it's all good learning and this sort of forum is great for ideas.

WAN circuits where I am are expensive and the customer won't spend the extra money ... I will be testing Peribit on the links next month ... ping responses are all below 100ms and I can enable QOS.

Rgrds

 

[computerhighguy]

The MC3810 comes in serveral configurations. The most pupular, and the cheapest is the MFT (Multi Flex Trunk) configuration. This would allos you to hook up to PRI card for sure, and possible an E-1. Some research would have to be done on that.

The MC3810 takes the raw voice and converts it to whatever format you want (i.e. G729). The 2600xm would work for CME call processing and it could use the MC3810 as the gateway between the VoIP landscape and the existing PBX. The MC3810 is old, but cheap. You can easily purchase a second one as a backup for little $$$. Cheaper than sticking some HDV modules into your 2600. You would only need 1 MC3810 at the mainsite. After that, the Cisco phones know how to decode the signal. That is what is so great about the VoIP. It is easily extensible. All you need is a reliable connection back to the call processing device (in this case the 2600xm) and you are golden (well there is a little more to it than that).


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[saxon747]

Hi, maybe I'm missing something because I'm confused!. ... Coffee has not kicked in yet.

Do I not need a 3810 on each site or HDV ... I want to conserve bandwidth both directions ... won't the remote sites still try to connect to HQ using G711 as the remote sites will not have some method of converting G711 codec to G729 before sending it over the WAN ... or is it possible to have the phones use a different codec.

Rgrds

 

[ccmuser]

How many analog voice ports are you looking for?

All cisco router DSP with enough PVDMS will support g.729. It is actually the default codec for IPV4 peers! You can get down to about 15K per call running RTP header compression. on PTP links with a decent router this actually works quite well. My quick advice.. 37/3800 central site with NM-hdv. depending on size of site.. 2600 with ethersiwtch mod may be a cost effective choice for remote. or cheap 1700 with seperate switch will be fine


But all said and done.. Just buy full blown call manager!!! The major cost is in phones and switches.. Call manager and unity server bundle can be had for under 9000 together!!!

 

[computerhighguy]

You only need one MC3810 (or some kind of HDV) to initially convert the calls from your PBX into IP. After that, you don't need another one.

Cisco 7900 series phones:
Each 7900 series phone has it's own DSP in it. You can, to great extent, control how these phones communicate. CME is a bit different from CCM. When you setup a dial-peer in CME, you can specify which codec to use. The phones will automatically use that codec for both transmit and receive. You can also specify how each phone should communicate to each other. You would want to use G729. I am not totally sure about CME, but with CMM you can put phones in Device Pools and make global changes about how each Device Pools communicates (i.e. which codec) to use on the downstream and the upstream. So it is possible for a phone ro receive a call using G729 and transmit back using G711.




It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[ccmuser]

computerhigh guy.. Have you verified seperate upstream and downstream codec? I don't that actually works..

What I am sure you can do though is set up a call so that call dialed from one side of the call will settup with a codec differnt then calls originating from the opposite end. But once the call codec is negotiated both ends will use the same codec for upstream and downstream.

Press the I button twice on your phone when you are in a call to verify this. You will see that both sides are( and must be) the same..

 

[computerhighguy]

For CCM it works. I doubt that the same functionality works in CME exactly the same way, but you can get it to work if you fanagle it a bit (I am pretty sure, haven't had much call for it).

Phone to phone communications:
Go into the Regions and you can setup how the different regions (I know I said device pool, I was wrong) communicate with each other. You can specify which codec to use in which direction.

Phone to gateway or gateway to phone communications:
When you communicate with a gateway, however, you pretty much are stuck with the same codec for both ways.


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[ccmuser]

computerhighguy,

Have you ever verified this? Press the I button twice while you are in a call and see if you ever see different codecs on the phones without the use of a transcoder over a trunk.

The reason I don't think this works is all the regions really due is supply the H.323 Proxy(cisco callmanager) h.245 call setup information about SCCP endpoints(ip phones)

I do not believe that ITU-t defines independant callsetup for each direction of the call.

Also DSP limitation of devices such as the ATA186 would fall apart if this paradigm was used. I am not saying I am sure for 100 percent.. But I would bet a pretty good piece of the farm on this one!!!

Again setup a lab and press I twice on the phones It displays codec activly being used!!!

 

[ccmuser]

Now that I think about it.. region configs are bound to each other.. Once you change one direction it automatically changes the other! The phones must use the same codec... I don't think you have a choice

 

[computerhighguy]

Hmmm. I'll look into it. i am pretty sure i have verified this. I'll get back with you tomorrow with the results.


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[computerhighguy]

Alrighty. I did some testing this morning and once again, I am incorrect. The Tx and Rx Codec are the same. The region setting does determine which codec is used for a phone to phone call.


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)