I know you cannot share a DSP between a conference profile and voice ports, I have a 2801 router with PVDM2-32 which is 2 DSPs and I am issues sharing these PVDMs between 1 conference session and 2 FXO trunks, when I turn on the conferensing profile and attempt to make a call I get an error indicating not enough PVDMs, when I shut the conference profile I can make call fine, any ideas? Here is the partial config:
dsp services dspfarm
sccp local FastEthernet0/0
sccp ccm 10.10.11.100 identifier 2 version 4.1
sccp ccm 10.10.11.101 identifier 1 version 4.1
sccp ccm group 1
associate ccm 2 priority 2
associate ccm 1 priority 1
associate profile 1 register BAR-CONF
dspfarm profile 1 conference
maximum sessions 1
associate application SCCP
Have you tried changing codec complexity under voice-card to see if that helps. 32 channel dsp is plenty to support that many sessions of conferencing and 2 FXO ports.
If you are not using g729 for conferenceing, you may remove those. That should free up some dsps.
You really need to check your resource requirement with the dsp calculator tool- does that give you any errors?:
I think there are some limitations for the 2801 and conferencing, I will look for that. I am pretty sure analog ports also preallocate resources, and those are not shared, but since you have 2 dsps, and only 2 ports, I think the two voice ports should stay on one of them, and leave the other available. Here is something from the SRND:
Hardware Audio Conference Bridge (Cisco NM-HDV2, NM-HD-1V/2V/2VE, 2800 Series, and 3800 Series Routers)
DSPs that are configured through Cisco IOS as conference resources will load firmware into the DSPs that is specific to conferencing functionality only, and these DSPs cannot be used for any other media feature.
The following guidelines and considerations apply to these DSP resources:
?Based on the C5510 DSP chipset, the NM-HDV2 and the router chassis use the PVDM2 modules for providing DSPs.
?DSPs on PVDM2 hardware are configured individually as either voice termination, conferencing, media termination, or transcoding, so that DSPs on a single PVDM may be used as different resource types. Allocate DSPs to voice termination first, then to other functionality as needed.
?The NM-HDV2 has 4 slots that will accept PVDM2 modules in any combination. The other network modules have fixed numbers of DSPs.
?A conference based on these DSPs allows a maximum of 8 participants. When a conference begins, all 8 positions are reserved at that time.
?The PVDM2-8 is listed as having ? a DSP because it has a DSP that has half the processing capacity of the PVDM2-16. For example, if the DSP on a PVDM2-8 is configured for G.711, it can provide (0.5 * 8) bridges/DSP = 4 conference bridges.
?Use Table 6-1 and Table 6-2 to determine how many DSPs may be provisioned with specific hardware.
?A DSP farm configuration in Cisco IOS specifies which codecs may be accepted for the farm. A DSP farm that is configured for conferencing and G.711 provides 8 conferences. When configured to accept both G.711 and G.729 calls, a single DSP provides 2 conferences because it is also reserving its resources for performing transcoding of streams.
?The I/O of an NM-HDV2 is limited to 400 streams, so ensure that the number of conference resources allocated does not cause this limit to be exceeded. If G.711 conferences are configured, then no more than 48 DSPs should be allocated per NM because (48 * 8) participants = 384 streams. If you configure all conferencing for both G.711 and G.729 codecs, then each DSP provides only 2 conferences of 8 participants each. In this case, it is possible to populate the NM fully and configure it with 16 DSPs so there would be 256 streams.
?Conferences cannot natively accept calls utilizing the GSM codec. A transcoder must be provided separately for these calls to participate in a conference.
?Any PVDM2-based hardware, such as the NM-HDV2, may be used simultaneously in a single chassis for voice termination but may not be used simultaneously for other media resource functionality. The DSPs based on PVDM-256K and PVDM2 have different DSP farm configurations, and only one may be configured in a router at a time.
Calculator looks fine with just the one conference session and 2 fxo ports.. if you do a 'sh voice dsp group all' does that show you your dsp status, and look like you expect? Maybe try it with both configs, see what happens...
I wonder if it is registering one dsp with the Sub and one with the Pub, if it registers to both? You could remove one for test to see if that is what is going on...
I know it only registered once, unfortunattley the router is down now and wont go live for couple of weeks, so I cannot test anything. I will take a look at it once again once I am ready to deploy it, it was just puzzeling to me why this was occuring.
You mentioned to assign the DSPs to voice ports first, aren't the PVDMs assigedn to voice ports automatically when they're detected in the chassis? I added the dspfarm and conference profile after the voice-ports were installed. What is the best way to check if DSPs are associated with voice port?
sh voice dsp will show you what dsps are being used, in the old days, if you had a T1, it would statically allocate all of the channels for that, and for any analog ports. Now, with dynamic allocation, if you have a T1 you only see dsps used for ports with active calls, but your analog channels still show the static allocation, it shows the dsp number and the channel,and then the voice port. For those analog ports, it should happen at bootup, you don't even have a chance to do anything else, so I am really surprised it lets you take them away later on. The only reason I can think of that you get that error is if the special conference dsp code gets loaded to both the dsps for some reason, because that takes it away from voice ports - we have run into trouble with the order of building things with a PRI, because those are dynamic, but not with these..I would get it turned back on and try some of the troubleshooting commands from this doc:
and definitely tell us what you find!
In this generation of DSPs the allocation of resources is dynamical (as Mary stated) Voice channel is dynamically allocated on per-call-base for flex mode, the number of voice channels is not fixed. It keeps on changing depending on the number of calls terminated on the dsp and the codec used by those calls.
Analog ports on any other product like the NM-HDV2 or NM-HD-* series of products cannot share ISR onboard DSP resources and must be serviced from a DSP local to the NM. If you throw in T1/E1 interfaces then DSP sharing with the ISR onboard DSPs is possible, and then calculations for DSP counts become a bit more complex.
Check this link for more info:
On ISR platforms the onboard C5510 DSPs operate in flex mode by default and can hence be used by other telephony interfaces.
I had had a tac case open on dsp sharing, having trouble with sharing across MB ports and some on NM, and got that working, but in the course of that I asked about the dynamic allocation and analog ports, since I have 28xx/38xx out there with 5510 dsps on the MB and analog ports along with T1 ports, and if I do a sh voice dsp I show the analog ports always assigned to dsps, where the T1 ports only show up as they have calls. I asked the TAC engineer about this, and he said that the analog ports are still allocated at bootup, and therefore always grab a dsp, and you need to account for that when deciding how many dsps you need. Maybe he was wrong, but it matches what I have been observing.