E1 PRI lines 4331

[adamgibs7]

Dears Experts,

I was sizing the DSP resources fro the router 4331 with 2 port E1 card by a  DSP calculator tool, hence with PVDM 64 with G711 codec i can have 60 channels that is good enough, my router will be only used for PSTN calling 

My question is below.

• In DSP calculator it was asking for codec G711 or G729 hence if i select G729 it ask for PVDM 128 but for G711 PVDM 64 are enough  hence my plan is to configure the Gateway as H.323 or MGCP,  how i shld confirm the codec for the PSTN calls.
• If i choose the no's of conference the DSP resources increases, hence what i would like to know when DSP resources of router for conferences will be used becz my router will be acting only as PSTN gateway ( H.323 or MGCP)
• Do i require a conference resources in my scenario ???  or a conference resources are used only when a router is acting as a CME ?? or in which other situation if any body can help me to understand more scenarios.

[Chris Deren]

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• In DSP calculator it was asking for codec G711 or G729 hence if i select G729 it ask for PVDM 128 but for G711 PVDM 64 are enough  hence my plan is to configure the Gateway as H.323 or MGCP,  how i shld confirm the codec for the PSTN calls.

PSTN calls would not use codec as they will use the PRI, it's about the codec between the Gateway and the endpoint making/receiving the call which is controlled by the Region setting between the gateway/trunk configuration in CUCM and the endpoint i.e. phone.  

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• If i choose the no's of conference the DSP resources increases, hence what i would like to know when DSP resources of router for conferences will be used becz my router will be acting only as PSTN gateway ( H.323 or MGCP)

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Not sure I follow the question, you can dedicated some DSPs for hardware conferencing and register them with CUCM using SCCP, but with PVDM4-64 you would not have any DSPs left for that

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Do i require a conference resources in my scenario ???  or a conference resources are used only when a router is acting as a CME ?? or in which other situation if any body can help me to understand more scenarios.

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Above you talked about MGCP which implies you have CUCM, here you talk about CME, which phone system are you using?  

In either case CUCM can you software conferencing from one or more of the servers, but it supports only G711 codec for ad-hoc, meet-me, etc conferencing.  CME can also use IOS based conferencing but is limited to only 2 participants per conference. 

[adamgibs7]

Dear Chris

Thanks for the reply, let me rephrase my question,

PSTN calls would not use codec as they will use the PRI, it's about the codec between the Gateway and the endpoint making/receiving the call which is controlled by the Region setting between the gateway/trunk configuration in CUCM and the endpoint i.e. phone.

here is the link that that says that PSTN uses G711, https://www.globalknowledge.com/us-en/resources/resource-library/articles/cisco-ip-phone-audio-codecs/

Above you talked about MGCP which implies you have CUCM, here you talk about CME, which phone system are you using?

Yes i will configure my router as an MGCP or H.323 Gateway no doubts for my knowledge i was asking you this question,,

what i mean to say here is that  while calculating DSP in the DSP tool  there is an option it prompts that router will be used for transcoding, conferencing   so if i have CUCM 12.X with MGCP gateway the conferencing will be on the server so i shld not consider the DSP resources on router for conferencing, Please correct me if i m wrong.

thanks 

[Chris Deren]

It is misleading, PRI uses digital signaling such as PRI NI2, etc.  not voip codec such as G711. The link you posted does not say anytime about PRI signaling but merely talks about VOIP audio codecs, which is what is used between the Gateway and IP leg of the call such as an IP phone.

DSPs on the router can provide media resources such as conferencing, transcoding, MTP in which case you would defined the DSP farms on the routers and configure SCCP protocol to have them register with CUCM and then assign to proper MRG.  If you need/desire these resources then you would provision them as such, but they are not required for the PRI functionality as that is separate supplementary feature.  Conferencing and MTP can be provided by software resources on CUCM assuming G711 is all you need, transcoders on the other hand require hardware PVDMs and if you expect the need for them you would need to account for it. 

[adamgibs7]

Dear Chris

thank you very much for your explanation, i went to the below link the same as you are explaining me,

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/91863-trans-conf-iosgw-ccm.html

I have some question here for the above link, becz in my scenario i have a WAN connection but i don't have a router becz i am getting a MPLS link from ISP that link i m connecting on my firewall or i will terminate on core switch.

1. why we need routers on both the sides when a inter cluster trunk or sip trunk can be configured when i have a IP reachability ( if the call manager are of different cluster)
2. I can configure regions and accordingly the codec will be used by the call manager if the call manager are in same cluster acting as an centralized server.
3. so if don't keep the router ,,  transcoding and Conference resources will be used on call manager please correct me if i wrong

Thanks

[Ayodeji Okanlawon]

The VoIP codec you use is important because your dsp module on the router has to modulate your G711 or G729 PCM to TDM for transmission over PSTN. You will require more dsp resources when using G729 compared to G711...

Secondly, this is 2020...the year of Covid19...why on earth are you using MGCP? Or even H323....We are not in 2003( when China gave us SARS)

[adamgibs7]

Dear ayodeji,

Secondly, this is 2020...the year of Covid19...why on earth are you using MGCP? Or even H323....We are not in 2003( when China gave us SARS)

so which protocol i shld use for registering the gateway to the CUCM ???

[Chris Deren]

Use SIP to control the gateway, even if it's a PRI gateway.  On CUCM build SIP trunk pointing to the gateway, and configure the gateway with SIP settings including proper SIP dial-peers, there is plenty of documentation on how do do that.  SIP has been de facto Voip protocol in the industry and preferred by Cisco for several years with new enhancement being introduced in every release of CUCM and IOS. H323 and MGCP have not been updated in years.

[adamgibs7]

Dear Chris

you and Ayodeji are expert  on Cisco community, please bare with me,

Use SIP to control the gateway, even if it's a PRI gateway.  On CUCM build SIP trunk pointing to the gateway, and configure the gateway with SIP settings including proper SIP dial-peers, there is plenty of documentation on how do do that.

can u convince me to have additional huge configuration of a sip as a hop on a router  ,when i able to route the calls by CUCM by partitions, CSS, regions, Device pool, and i also can use conferencing resources of a server which will give me 256 participants in once conference and when i have an IP reach ability to other location then why i should invest in the router. 

As ayodji told that why i have to use H.323 and MGCP for PSTN gateway ??? if the itsp doesn't advise to  SIP trunk solution then i have to go with this solution do any other solution exist except , H.323,MGCP,SIP.

[TONY SMITH]

You need a router because you're talking about connecting ISDN lines, and these can't plug straight into CUCM.  CUCM only talks IP, so you plug your ISDN into the router.  Then configure router and CUCM to talk IP to each other, and SIP is the best way to do this nowadays.

In other posts you talk about multiple CUCM clusters.  What is the context? 

[adamgibs7]

Dears 

I hope things are getting mixed and consfused.

i have 4 sites site A,B,C,D, all sites  will use Site A ISDN router to call PSTN and on site B i will keep additional CUCM to register IP phones of site B,C and site D,

Question 1

Now i need to have inter site calling so why do i need to have a router for connecting these branches i have IP reachability to these site servers so i will configure dial pattern,DP,Region,Location, in CUCM and i can achieve the goal i don't need a router and DSP resources on in, ,please correct me if i m wrong.

Question 2

@TONY SMITH 

as what you said for PSTN we need router no doubt, its mandatory to have that is clear i was asking apart from sip,mgcp,h.323 are there any other protocols to configure router and CUCM to talk IP to each other, , i don't think so please correct me if i m wrong

Question 3

And if a router is only used for PSTN calling then the aspect for DSP resources will be only E1 channels , no other aspect should be considered ( MTP, conference etc etc) while sizing the DSP resources, Please correct me if i m wrong. 

there was an another question y i should invest in voice router for branch connectivity,

[TONY SMITH · Accepted solution]

For clarity let's refer to the router which will have the E1 PRI as a "Gateway"" to reflect its specific function in your design.

Question 1.  If you have IP reachability between the four sites then you don't need addition routers in the path specifically to support voice calls between the sites, or from the remote sites to the PSTN gateway.

Question 2.  The PSTN Gateway will need to have specific configuration for the PRI lines,  then SIP configuration to talk to CUCM.  CUCM in turn will need SIP configuration to talk to the Gateway, in addition to the Route Patterns etc that make up the CUCM dial plan.

Question 3.   Whether or not you need Transcoder or Conference resources will depend on your overall design.  However you are correct, if you only want the Gateway to act as a PSTN gateway then you only need DSPs to match the number of channels and the type of codec you're using on the CUCM side.  

In your wider network are you using G.711 between your sites?  And does conferencing already work satisfactorily?  The relevance is that the software conference and MTP resources on the CUCM servers only support G.711.  So if you use different codec between the sites then conferencing may not work.

[adamgibs7]

Dear Tony

In your wider network are you using G.711 between your sites?  And does conferencing already work satisfactorily?  The relevance is that the software conference and MTP resources on the CUCM servers only support G.711.  So if you use different codec between the sites then conferencing may not work.

ooh i see so if i configure a region calling with G729 which will use low bandwidth but  conferencing for the users sitting in different regions will not work that's the reason i need to have a router with DSP resources as an hop to reach to the other sites, Please correct me if i m wrong.

Thanks

[TONY SMITH]

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ooh i see so if i configure a region calling with G729 which will use low bandwidth but  conferencing for the users sitting in different regions will not work that's the reason i need to have a router with DSP resources as an hop to reach to the other sites, Please correct me if i m wrong.

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That's correct. A conference bridge using DSPs can conference participants using different codecs. Or a transcoding profile could be used to convert participants to G.711, this would make sense if the PSTN gateway is at the same site as CUCM.

One possible compromise might be to put the CUCM conferencing into a different region, allowing G.711 from everywhere.  Then your normal site to site calls, and site to PSTN would all use low bandwidth, and only conferenced calls would use G.711.  This isn't a scenario I've configured so there may be a gotcha somewhere in there.

[Chris Deren]

Ayodeji, thanks for pointing that out, I tried to stay on focus with answering the question, but I concur with guiding the poster to use correct approach.