Can someone explain to me what kind of resouces will be taken on CCM server to make a conference call? and how many conference can be supported on a single MCS7825 server?
My scenerio is as follow. I have one side which is a H323 third party gateway and needs to use MTP (Media Termination Point) so the line can be place on hold. An IP Phone answers the incoming call from the gateway and conference another call in (which was made out to public PSTN using the same H323 gateway). Currently, on all sides, I had configured to use G711 but I would like to move to LBR codec such as G723a if supported.
Thanks very much.
The software MTP and software Conference Bridge only support the G.711 codec.
If you want to use supplementary services such as hold and transfer with a LBR codec then you need to either use: a transcoder such as one of the cards available for the Cat6k or Cat4k switches, or use a 3rd party H.323 gateway that is H.323v2 compliant and uses the "Empty Capabilities Set" H.323 feature, also known as Open/Close Logical Channel. Then, you will not have to use the software MTP.
If you want to include LBR audio streams in a conference call, you need to use a hardware conference bridge such as one of the cards available for the Ca6k or Cat4k switches.
Please correct me if I am wrong.
I read the doc and it appears that the conference bridge must be on the same network where CCM is?
Also, there doesn't seems to be a way to specifically used a conference bridge pool. Basically, you can't allocate certain calls to use specific conference bridge hardware?
This makes conference bridge very limited and can't really be used on a very distributed network.
Is there any conference solutions available out there?
No, the conference bridge does not have to be on the same network where the CallManager is located.
However, with CallManager 3.0 there is no way to force specific devices to use one hardware conference bridge before any other one... at least not with much granularity. What does happen is that the CallManager points the device wanting to initiate a conference at the first available conference bridge device in the same device pool as the device.
In CallManager 3.1, there will be a "Media Resource Group List" where you will be able to tell specific devices which conference bridge, transcoder, music on hold server, etc. to try to use and in what order if you have multiple devices of each type.
Thanks for the quick reply. I believe as long as I create a well-planned device pool structure, I should be able to get certain gateways to use certain conference bridge.
Besides from Catalyst cards, what other product from Cisco is available for transcoding and conference? I like products like DT24+ as it fits our scheme perfectly.
FYI. We're finalizing to use DT24+ and AS5300 as our gateway of choice.
Right now, the Cat4k and Cat6k cards are the only options I know of to do hardware based conference and transcode.
In the next couple months there will be IOS software available for the VG200 gateway (and hopefully for additional gateways later) that will allow you to use it as a hardware conference / transcoder / both when using the NM-HDV card with C549 DSP chips installed. Note that this feature will also require CallManager 3.1
Again, I must thank you for the detail information.
Let me further explain our scenerio and would you recommended for us what Cat4k equipment we would need.
We will setup several POPs; in each of these POPs, we have an IVR system (programmed using Cisco TAPI driver). These IVR systems will verify the users and use conference bridge to conference two calls. The minimum configuration is required to handle 24 calls (or one PRI). Currently, only G711 is supported.
The calls comes in to each POPs using PRI; therefore, we will need VoIP gateways to answer the call. Currently, I planned to use DT24+ and AS5300. Like many others, we prefer to use G723a throughout. However, since IVR only supports G711, transcoding is needed.
The call manager is centrally located in one of our data center.
To this extent, we will need to invest extra hardware for conference bridge, transcoding, MTP features.
It will be great if we can locate a Cisco equipment that handles 1) the basic VoIP gateway function, and 2) have extra resouce (DSP?) for conference bridge, transcoding, and MTP (if necessary).
Thanks for taking the time to help me.
I had tried to dig up as much info as possible on Cat4k but still not exactly sure which part I will need to get to do it right.
It will be great if you can spend some time and help me on this one.
You would need to the WS-X4604-GWY card with the 4 DSPs and a VWIC with the T1/E1 interface you need. See the following URL:
and look at the IP Telephony Gateway Mode section.
I had also dig up this information based on keyword "WS-X4604-GWY".
From Table 9-1, Cat4k doesn't really provide what I need; since it only supports max 8 conferences all based on G711. This would require extra transcoding service needed.
From my application, assuming assuming 1) gateway running G723, 2) IVR running G711, and 3) minimum 24 channels configuration. I will require 1) 24 conferences with three participant each; 2) 24 G723 to G711 transcoding service.
Below is the configuration I think that works, please correct me if I am wrong.
1 x Cat6k with 1 x WS-X6608-T1 (which has 8 ports)
Ports are configured as follow,
2 x PSTN PRI gateway (46 channels)
6 x Conference bridge (192 sessions available)
192 sessions are allocated as follow for each call,
3 x participants (1 x G711, 2 x G723)
1 x transcoding (G 711 to G723)
This means total of 48 calls can be used with this configuration.
I really appreciate your help on this. Thank you so much.
With above configuration in mind, I should be able to purchase one 6006 Chassis with WS-X6608-T1.
Do I need to purchase any extra modules? ie, Supervisor Engine, ethernet port module.
I know this isn't much VoIP related but it will be great if you can help me complete the configuration. The picture is getting clear and clear, all thanks to you.
A 6006 with a 6608 should fit your need. Take a look at the following URL which describes DSP provisioning on the 6608:
Basically, with one 6608 you will be able to support at least 2 PRI's worth of calls including the necessary conferencing resources. No additional transcoding resources will be required since the conference bridge ports on the 6608 can do transcoding natively. Let me explain how the resource utilization will look:
For each call, we will use 1 port on the gateway and 3 participants on one port of the conference bridge. Since you are using G.711 and G.723 only (not G.729) you can have up to 10 3-party conferences on a single port. This means that every 10 calls will use one port. Since you have a maximum of 23 calls port port and 2 ports, this is 46 calls which would require 4.6 conference bridge ports (5 ports). This leaves you with one additional port.
The above is a worst-case scenario. Assuming that a particular call goes out the same gateway, then you're using 2 gateway ports in each conference, so you'd only need half the conferencing resources. Either way, one blade is enough for your needs.
As for your question regarding the other modules, the 6608 only connects to the backplane of the switch and to the PSTN via the front panel PRI ports. You will need at least one Supervisor module which has 2 Gigabit Ethernet ports on it. If you don't have Gig-E at your Pop, then you'll need a 6348-RJ45 card for 10/100 Ethernet connectivity as well.
Let me know if this clears things up for you.
Thanks for clearing this up for me. It seems like I got the right idea about the DSP usage.
However, I am still a bit confused as to the "Supervisor" module. Since I don't need the Gig-E, why would I still need it? Can't I just use the 6348-RJ45?
Also, I would simply hook up an ethernet uplink cable to the 6348-RJ45 and all voice traffic would use that as outgoing port?
You can't run the Cat6k switch without a supervisor module. It contains the CPU for the switch as well as certain other core functions which you need to have to operate the box.
What Paul was saying was that the supervisor module comes with Gig-E ports. So if you are using a Gig-E backbone at the POP then you can use those. If you are using 10/100 then you would need a regular 10/100 ethernet blade such as the 6348 module. And yes, if you have a supervisor module (required) AND a 6348 module (required if you need 10/100 connectivity) then you can plug one of those ports on the 6348 module into your WAN router and the VoIP traffic will use that port.