DSP resource usage

[redyps90210]

Hi guys,

I just want to know if there's any documentation that explains how DSP resource in G450's are being utilized.

We're currently adding traffic to our system but our current vendor cannot give an accurate answer how many DSP's are being utilized in every call. Some says it will use 1 for the SIP trunk, 1 for the music on hold and 1 for the agent. Some says it only uses 1 DSP = 1 call. Some says it doesn't if we're using the same codec althroughout (no codec conversion from G711 to G729 and vice versa).

By the way, we're currently set up as a pure IP solution. SBC --> Session Manager --> Communications Manager. All inbound, outbound and internall calls are via SIP. We only used G450's for the DSP's and Voice announcements.

Thanks in advance!

 

[kyle555]

Presuming IP stations everywhere, presuming no call recording, presuming you're allowing direct IP and are using IP early media in SIP to set up your calls RTP directly from the SBC to the stations, then yes, you're just using DSPs for announcements.

Broadly, 1 DSP per call that requires it. In the background they have "points" that correlate roughly to # of calls. G.729, T.38, V.150.1, all make it vary a little bit.

--------------------------------------------------------------
Board type     : MP80
Hw Vintage     : 7 A
Fw Vintage     : 132

DSP#1 CURRENT STATE
--------------------------------------------------------------
In Use         : 2 of 80 channels, 60 of 2400 points (2.5% used)
State          : In Use
Admin State    : Release

Core# Channels Admin     State
      In Use   State
----- -------- --------- -------
    1  1 of 20   Release In Use
    2  1 of 20   Release In Use
    3  0 of 20   Release Idle
    4  0 of 20   Release Idle

But basically, if announcements for hold music have people bridge in wherever the announcement is at, that's 1 DSP. You'd also use 1 DSP per call for every simultaneous announcement that plays independently of others.

 

[redyps90210]

Thanks kyle for the prompt response!

Yes we have IP stations everywhere. We do have call recordings but not for all phones. Yes we allow "direct IP-IP audio connection" on all stations. If you're referring to the early media between the SBC and Media-gateway (where the DSP's are located), then Yes we are set up like that.

So my understanding is, if we have let say 100 calls in queue and playing the same music-on-hold (set as integ-mus), that would use around 100 DSP's? Let's exclude the call recording for simplicity.

Sorry but I used to work with the legacy CM where we have the MedPro cards. But now getting used to the DSP concept and just the "processor ethernet".

 

[kyle555]

It would be 1 DSP to stream the same music on the same loop to 100 people connecting to the gateway over IP.
It would be 100 DSPs to stream the same message, but starting at the beginning of the message to 100 people independently of one another.

Call recording is 2 or 3 DSPs per call usually.

 

[redyps90210]

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 100 DSP's for calls in queue and playing the same music on hold (set to integ-mus).

Any idea where else can we check?

 

[kyle555]

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.

"When Type is integ-mus, this field defaults to b. Callers are
connected to the announcement at any time while it is playing.

 

[redyps90210]

It has been set to "integ-music". We will test and let you know.

 

[jimbojimbo]

DSP utilization is dependent on many factors. For example, a single inbound SIP trunk call could use half a dozen DSP resources throughout the call (with call recording, service observer, and IGC resulting from a permanent service link connection) or could use none (direct from SBC to endpoint and no call recording).

For a caller to hear MoH on an inbound SIP trunk they would have to use at least 1 DSP to reach the TDM bus on the gateway where the MoH is playing. On the other hand, an inbound ISDN call on the same gateway would use 0 DSP resources while the same call coming in on a gateway without the MoH announcement would utilize 2 DSP resources (good reason for audio groups).

Also keep in mind the number of DSP resources per gateway can vary depending on the type and quantity of daughter boards and the CM software release (older CM versions only support up to 240 DSP resources).