It would be DSPs to stream the same message, but starting at the beginning of the message to people independently of one another. Call recording is 2 or 3 DSPs per call usually. Apparently it looks like the "1 DSP to stream the same music on the same loop" doesn't work on us. We already checked the announcement file and it's properly configured.
But we're still using around DSP's for calls in queue and playing the same music on hold set to integ-mus. Any idea where else can we check? Is your announcement of type "integrated" or "integ-music". Check the CM Screen Reference, and test 2 calls in to that announcement. You'll get away with 1 DSP for all if they bridge into the looping announcement, but you'll use 1 DSP per call if the announcement always plays from the beginning on each call that hits it.
Callers are connected to the announcement at any time while it is playing. It has been set to "integ-music". We will test and let you know. DSP utilization is dependent on many factors. When you have determined how many resources of each type that you need, you must determine the number of modules you need to supply the above resources, by doing the following:.
Determine the total number of streams you need by dividing the number of resources you need by the maximum number of channels per stream, then rounding up. Determine the number of DSP chips required by dividing the total number of streams by two. For example, if you want the maximum MFR1 receivers for a stream, you would need to license x 5 Resource Points 1, Voice cards that do not have C DSPs can be configured only for medium or high complexity.
Voice cards that are equipped with C DSPs have an additional complexity option called flex complexity. Setting the codec complexity to medium or high sets the number of voice terminations per DSP to a static number. Flex mode has an advantage when calls of multiple codecs must be supported on the same hardware because flex mode can support more calls than when the DSPs are configured as medium or high complexity. However, flex mode does allow oversubscription of the resources, which introduces the risk of call failure if all resources are used.
This can reduce the possibility of oversubscription when using flex complexity. It can also make it easier to add DSPs to an existing gateway. You should configure all voice cards that share DSPs for the same complexity.
Because this situation is rare, the practical maximum number of transcoding sessions per DSP is 8. Note: Software-based MTPs can support two voice streams with the same packetization rates.
If the voice streams use different packetization rates, a DSP is required. You need to consider only one factor when calculating conference bridge DSP requirements: the number of conferences required. For example, the C supports two mixed-mode conferences with up to eight participants each.
Therefore, it is technically accurate to say that the C supports 16 conference participants. But this does not mean we can have 4 conferences of 4 participants each. Number of DSPs required depends on number of Conferences. Particpants per conference are irrelevant. In this case, a packet-to-packet gateway transcodes the G. See Figure In Figure , a remote user joins a conference call at the central location.
The following list gives the channel usage:. On the Cisco Catalyst , all voice streams are sent to single logical conferencing port where all transcoding and summing takes place. If transcoding services are needed between clusters, configure the intercluster trunks with MTP. The Cisco Catalyst module uses the MTP service regardless whether transcoding is needed for that particular intercluster call. Figure shows an intercluster call flow. To scale IP telephony systems in large enterprise environments, you must use hardware-based conferencing.
The hardware for the Cisco Catalyst and switch families keeps this requirement in mind. The Cisco Catalyst voice modules can handle conferencing in hardware, eliminating the requirement of running a software conferencing service on a Windows NT server in the IP telephony network. The following points summarize the design capabilities and requirements of the Cisco Catalyst voice modules:.
Transcoding can convert G. The Cisco Catalyst conferencing module supports up to six callers per conference call with a maximum of 32 simultaneous G. This configuration results in a maximum of conference participants per module with G. See Table for a summary of conference call densities for each module.
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